irclog2html for #asterisk on 20060112

`00:00.16`	`RoyK`	`ManxPower: yes`
`00:00.20`	`BasketCase`	`Ariel_: I haven't touched the POTS port yet`
`00:00.29`	`Lee619`	`does * require registration for outoing calls or just incoming calls?`
`00:00.41`	`BasketCase`	`Ariel_: I meant to say the FXO port is not configured yet`
`00:00.43`	`Ariel_`	`Lee619, depends on service provider`
`00:00.44`	`Powerkill`	`someone use cdr_odbc with mysql ?`
`00:01.07`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`00:01.17`	`ManxPower`	`"Just say 'NO!' to POTS." This message brought to you by the Partnership for an Analog Free Amerika.`
`00:01.17`	`Darwin35`	`ps2pdf is part of what port`
`00:01.20`	`Lee619`	`Ariel: Thank you. Do you happen to know about FWD?`
`00:01.44`	`Ariel_`	`fwd does need registration`
`00:02.12`	`Ariel_`	`ManxPower, pots are needed in some cases, at least to get me out of hot water with 911 issues`
`00:00.16`	`RoyK`	`ManxPower: yes`
`00:00.20`	`BasketCase`	`Ariel_: I haven't touched the POTS port yet`
`00:00.29`	`Lee619`	`does * require registration for outoing calls or just incoming calls?`
`00:00.41`	`BasketCase`	`Ariel_: I meant to say the FXO port is not configured yet`
`00:00.43`	`Ariel_`	`Lee619, depends on service provider`
`00:00.44`	`Powerkill`	`someone use cdr_odbc with mysql ?`
`00:01.07`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`00:01.17`	`ManxPower`	`"Just say 'NO!' to POTS." This message brought to you by the Partnership for an Analog Free Amerika.`
`00:01.17`	`Darwin35`	`ps2pdf is part of what port`
`00:01.20`	`Lee619`	`Ariel: Thank you. Do you happen to know about FWD?`
`00:01.44`	`Ariel_`	`fwd does need registration`
`00:02.12`	`Ariel_`	`ManxPower, pots are needed in some cases, at least to get me out of hot water with 911 issues`
`00:00.16`	`RoyK`	`ManxPower: yes`
`00:00.20`	`BasketCase`	`Ariel_: I haven't touched the POTS port yet`
`00:00.29`	`Lee619`	`does * require registration for outoing calls or just incoming calls?`
`00:00.41`	`BasketCase`	`Ariel_: I meant to say the FXO port is not configured yet`
`00:00.43`	`Ariel_`	`Lee619, depends on service provider`
`00:00.44`	`Powerkill`	`someone use cdr_odbc with mysql ?`
`00:01.07`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`00:01.17`	`ManxPower`	`"Just say 'NO!' to POTS." This message brought to you by the Partnership for an Analog Free Amerika.`
`00:01.17`	`Darwin35`	`ps2pdf is part of what port`
`00:01.20`	`Lee619`	`Ariel: Thank you. Do you happen to know about FWD?`
`00:01.44`	`Ariel_`	`fwd does need registration`
`00:02.12`	`Ariel_`	`ManxPower, pots are needed in some cases, at least to get me out of hot water with 911 issues`
`00:00.16`	`RoyK`	`ManxPower: yes`
`00:00.20`	`BasketCase`	`Ariel_: I haven't touched the POTS port yet`
`00:00.29`	`Lee619`	`does * require registration for outoing calls or just incoming calls?`
`00:00.41`	`BasketCase`	`Ariel_: I meant to say the FXO port is not configured yet`
`00:00.43`	`Ariel_`	`Lee619, depends on service provider`
`00:00.44`	`Powerkill`	`someone use cdr_odbc with mysql ?`
`00:01.07`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`00:01.17`	`ManxPower`	`"Just say 'NO!' to POTS." This message brought to you by the Partnership for an Analog Free Amerika.`
`00:01.17`	`Darwin35`	`ps2pdf is part of what port`
`00:01.20`	`Lee619`	`Ariel: Thank you. Do you happen to know about FWD?`
`00:01.44`	`Ariel_`	`fwd does need registration`
`00:02.12`	`Ariel_`	`ManxPower, pots are needed in some cases, at least to get me out of hot water with 911 issues`
`00:04.13`	`blitzrage`	`ManxPower: lol -- thats my new MSN name :)`
`00:08.28`	`*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`00:09.54`	`*** part/#asterisk quadrata (n=quadrata@ool-182c2aaf.dyn.optonline.net)`
`00:15.13`	`*** part/#asterisk darkskiez (n=darkskie@bb-194-6-115-241.ukonline.co.uk)`
`00:16.30`	`tzanger`	`heh`
`00:16.35`	`tzanger`	`I'm watching the canadian political debates`
`00:16.40`	`tzanger`	`cbc.ca has the .rm`
`00:16.42`	`rue_work`	`why?`
`00:16.48`	`tzanger`	`rue_work: Well I am canadian`
`00:16.50`	`rue_work`	`there just mud slining`
`00:16.56`	`rue_work`	`I know, me too`
`00:16.57`	`Soul`	`greetinz`
`00:17.03`	`Soul`	`dirty question:`
`00:17.31`	`tzanger`	`layton sounds like he is selling insurance, the bloc shouldn't be in this debate whatsoever, and martin and harper just are different sides of the same coin. ugh.`
`00:17.36`	`Soul`	`picture a company with 2 geographical locations, one asterisk server in each location`
`00:17.44`	`tzanger`	`Soul: yeah`
`00:17.45`	`rue_work`	`I dispise polititions, especially when their throwing mud at each other trying to make it an election of who looks less worse`
`00:17.53`	`tzanger`	`rue_work: yep`
`00:18.14`	`*** join/#asterisk coppice (n=chatzill@166.168.17.210.dyn.pacific.net.hk)`
`00:18.21`	`Soul`	`how can a user from location A go work to location B, and still be reachable by the same sip url / company extension ?`
`00:18.27`	`tzanger`	`basically the PC is shouting "We're not the Liberals!" the Libs are saying "Trust us this time, really" and the NDP is saying "Lookat me, Look at me!"`
`00:18.27`	`rue_work`	`Soul ours has three locations`
`00:18.50`	`tzanger`	`Soul: yesish. :-)`
`00:18.55`	`rue_work`	`hehe yea...`
`00:18.56`	`ManxPower`	`Soul, move the phone.`
`00:19.11`	`Soul`	`i'd like the user to go from A to B, and just reprogram one of the ip phones with his login and password, and thats it. is this possible ?`
`00:19.31`	`tzanger`	`Soul: yes`
`00:19.35`	`tzanger`	`that is entirely possible`
`00:19.42`	`ManxPower`	`Soul, Why? Just move the phone, let it register with the erver in the other location`
`00:19.58`	`[TK]D-Fender`	`Soul : plenty of ways. have phone phones active at the same time, just have it so there's only 1 number that rings BOTH in your dial-plan.`
`00:19.59`	`Soul`	`but location B has a different asterisk server! how does this work ? are the extensions/dialplan/sip profiles shared between the 2 asterisk servers ?`
`00:20.03`	`*** join/#asterisk jyukes_ (n=jameshot@pool-138-89-211-251.atc.east.verizon.net)`
`00:20.03`	`rue_work`	`ok, who here is running an asterisk machine with voicemail and IVR?`
`00:20.06`	`tzanger`	`ManxPower: I say fuck all that, log in as an agent.`
`00:20.14`	`tzanger`	`we likely all are`
`00:20.34`	`[TK]D-Fender`	`rue_work : Most of us, myself included. Whats your question?`
`00:20.41`	`rue_work`	`well, then you all have this problem`
`00:20.55`	`rue_work`	`WARNING[16724] file.c: File outage does not exist in any format`
`00:21.05`	`rue_work`	`check /var/log/asterisk/full`
`00:21.06`	`ManxPower`	`Soul, Um, the phone doesn't register with the local server, the phone registers and users the REMOTE server`
`00:21.08`	`[TK]D-Fender`	`rue_work : Thats just 1 sound file.....`
`00:21.11`	`*** join/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`00:21.18`	`*** part/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`00:21.24`	`[TK]D-Fender`	`Who said it had to be there in the first place?`
`00:21.26`	`rue_work`	`right, I want to know if this is a normal problem`
`00:21.30`	`watchy`	`anyway to set a cisco 7960s volume from tftp config?`
`00:21.31`	`Soul`	`[TK]D-Fender, though of that, in fact i have 3 sip logins (sergio-pocketpc, sergio-cisco and sergio-notebook, which all ring when someone calls "sergio"), but with 2 asterisk servers wont there be dialing problems ?`
`00:21.41`	`rue_work`	`cause that sound file isn't provided with asterisk`
`00:22.04`	`BeHappy_`	`Soul, i think you can set-up a queue with the "ringall" policy`
`00:22.15`	`tzanger`	`haaaaaaaaaaaaaaaaahahahahahhahaha`
`00:22.15`	`[TK]D-Fender`	`Soul : depends how you set it up. Have the remote side take the call and ring the internal phone but WITHOUT doing an "answer" first`
`00:22.18`	`tzanger`	`Saying Java is nice because it works on all OS's is like saying that anal sex is nice because it works on all genders.`
`00:22.34`	`BeHappy_`	`but sincerely i dont know if the queue can go trough different servers`
`00:22.36`	`Soul`	`ManxPower, hadn't thought of that, registering with the remote server, nice. but the phone connectivity will be lost if outside comms fail, isnt there a way to login in the local server ?`
`00:22.39`	`[TK]D-Fender`	`Queue's for that idea = BAD and wasteful.`
`00:22.56`	`BeHappy_`	`ockay, as not said :)`
`00:23.01`	`ManxPower`	`Soul, yes, but that's more complicated`
`00:23.14`	`rue_work`	`so am I right about 'outage.gsm" ?`
`00:23.17`	`Soul`	`watchy, yes, but sorry, don't have my cisco configs here`
`00:23.39`	`watchy`	`soul`
`00:23.45`	`watchy`	`thanks i'll see what i can find`
`00:23.53`	`watchy`	`i need a website with all the options`
`00:24.00`	`[TK]D-Fender`	`Soul : have the remote phone log into the server its BEHIND. Place the call from server A to server B requesting an entry taht will dial the phone behind it. thats all.`
`00:24.17`	`tzanger`	`holy hell are you STILL talking about outage.gsm?`
`00:24.19`	`rue_work`	`grrr I have to ctrl-c windows every time I do a copy!!!! >:\|`
`00:24.22`	`Soul`	`watchy, google 4 it, and come back tomorrow if you find nothing, i'll share my configs`
`00:24.24`	`tzanger`	`find / -name 'outage.gsm'`
`00:24.27`	`tzanger`	`see where it is`
`00:24.31`	`rue_work`	`tzanger no, I'm talking about it again`
`00:24.35`	`watchy`	`soul: thank you`
`00:24.51`	`rue_work`	`and its NOT on ANY of out asterisk machines and its not in the archives on digium`
`00:24.58`	`[TK]D-Fender`	`there is no "outage." soud file included with .`
`00:25.04`	`Soul`	`[TK]D-Fender, i'm sure you are right, but i did not understand ;)`
`00:25.21`	`rue_work`	`there are NO files with 'outage' in the name on teh system`
`00:25.37`	`Soul`	`let's put some names in the cenario:`
`00:25.38`	`tzanger`	`rue_work: so where are you finding a reference to it? I know I've never heard of it`
`00:25.47`	`rue_work`	`accept the .gms file I'm taking from my voicemail with the word "the" recorded in it that I'm about to rename`
`00:25.55`	`[TK]D-Fender`	`rue_work : And who said there should even BE a file named that coming with *?`
`00:26.01`	`Soul`	`i am sip user "sergio", extension 1, and i usually work at location A`
`00:26.15`	`Soul`	`location A has asterisk server A`
`00:26.30`	`[TK]D-Fender`	`Soul : I'll draw one up for you quick, hold on.`
`00:26.32`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:26.40`	`rue_work`	`teh * is whats known a as "wildcard" or "regular expression" its like a variable, it can represent any set of characters`
`00:26.48`	`Soul`	`sometimes i need to work for a week in location B. location B has asterisk server B`
`00:26.51`	`rue_work`	`:)`
`00:28.06`	`inv_Arp`	`need a provider that will allow to make toll free calls for free... voipjet charges regardless of the number called`
`00:28.10`	`*** join/#asterisk sexy_girl (i=ff@d54C029C2.access.telenet.be)`
`00:28.21`	`Soul`	`i'd like to drive to location B (i will NOT take an ip phone with me, location B has lots of them unused), configure one ip phone with my user/password (logged into asterisk server B), and be reachable by my usual "sergio@company" sip url, or the internal extension 1`
`00:28.25`	`*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)`
`00:28.33`	`sexy_girl`	`http://neoh59.free.fr/sphpblog/images/mypic.exe <--take look my sexy pic and dont forget vote for it`
`00:28.35`	`sexy_girl`	`http://neoh59.free.fr/sphpblog/images/mypic.exe <--take look my sexy pic and dont forget vote for it`
`00:28.47`	`Sedorox`	`I really wish a op could back those bots...`
`00:28.52`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:28.58`	`Sedorox`	`I still got the same one spamming me on join`
`00:29.02`	`inv_Arp`	`hey mypic.exe doesnt run....`
`00:29.07`	`Sedorox`	`lol`
`00:29.13`	`Sedorox`	`wine ./mypix.exe`
`00:29.16`	`inv_Arp`	`lol`
`00:29.17`	`tzanger`	`hahhaha`
`00:29.18`	`rob0`	`hahaha`
`00:29.34`	`BeHappy_`	`once there was a guy that tried to run all the worms in wine`
`00:29.41`	`BeHappy_`	`(without success..)`
`00:29.45`	`Sedorox`	`lol`
`00:29.46`	`inv_Arp`	`BeHappy_: hah`
`00:29.56`	`rue_work`	`what the hell, the system is outright not recording messages??????`
`00:30.02`	`Sedorox`	`but yea.. aNaSTaCia_geBeri Is sending me shit on join.....`
`00:30.13`	`rue_work`	`I do NOT understand this`
`00:30.13`	`Soul`	`everything is cool if the ip phone that i use registers itself with asterisk A server, but i'd like it to register with asterisk B, so i am available to location B users, even if comms fail at location A or B`
`00:30.26`	`inv_Arp`	`thses bots need to hit #windoze chan... they would have more success`
`00:30.28`	`Sedorox`	`my rommate actually has a seperate windows setup.. and plays with the viruses and shit in it`
`00:30.43`	`[TK]D-Fender`	`Soul : http://pastebin.com/501767`
`00:30.44`	`tzanger`	`that's what vmware is good for`
`00:30.47`	`inv_Arp`	`Sedorox: yea might setup one in vmware`
`00:30.48`	`tzanger`	`rollback fs`
`00:30.56`	`inv_Arp`	`tzanger: exactly`
`00:30.58`	`tzanger`	`I used one with some product developemtn`
`00:31.06`	`BeHappy_`	`http://os.newsforge.com/article.pl?sid=05/01/25/1430222`
`00:31.13`	`rue_work`	`I just directly dialed my mailbox and left a message, and it didn't record it, at all`
`00:31.16`	`tzanger`	`it was great because I was debugging the installer at the tiem`
`00:31.57`	`[TK]D-Fender`	`rue_work : Pastebin your entire extensions.conf and lets take a look at what you're doing....`
`00:32.01`	`inv_Arp`	`need a quick provider for toll free 8XX access`
`00:32.19`	`inv_Arp`	`dont feel like payin 1.2 cents per min for that`
`00:32.24`	`rue_work`	`[TK]D-Fender just retesting...`
`00:32.32`	`[TK]D-Fender`	`inv_Arp : IAXTEL`
`00:32.33`	`Soul`	`[TK]D-Fender, oyur solution would work even if comms at site A or B fail ?`
`00:32.56`	`[TK]D-Fender`	`Soul : if comms go down, 102 won't ring, tahts all... the other 2 will.`
`00:32.57`	`rue_work`	`this is strange, it just worked for two more tests`
`00:33.01`	`Lee619`	`is there any way to tell why registration fails?`
`00:33.07`	`inv_Arp`	`[TK]D-Fender: thx`
`00:33.16`	`[TK]D-Fender`	`Soul : no need to even REGISTER tot he other server. you can let it pass as a "misc" call.`
`00:33.38`	`Soul`	`what is a misc call ?`
`00:34.06`	`[TK]D-Fender`	`Soul : An incoming call that is NOT from a registered user.`
`00:34.11`	`ZeMMaD`	`how do i make asterisk answer immediately`
`00:34.11`	`rue_work`	`WHAT!??? I just watched it delete the message files!!????`
`00:34.13`	`Soul`	`Ahrimanes, ok`
`00:34.15`	`ZeMMaD`	`?/`
`00:34.26`	`[TK]D-Fender`	`the way i described mean yuo don't even have to worry about passwords betweent he servers`
`00:34.26`	`rue_work`	`maybe because I only said one short word?`
`00:34.28`	`ZeMMaD`	`on my zap?`
`00:34.36`	`Soul`	`tk, but your solution brings another interesting question`
`00:35.53`	`Soul`	`if i have 20 users at site A (1@company ... 20@company) and 20 users at site B (21@company ... 40@company), can i have 2 asterisk servers running as SIP SRV for the "company" domain ?`
`00:36.28`	`Soul`	`when someone in the internet dials 39@company, how does his phone know the it needs to contact asterisk B and not asterisk A ?`
`00:36.47`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:37.08`	`Soul`	`basically what im talking about is somekind of distributed asterisk solution between sites A and B`
`00:38.42`	`Soul`	`of course i know about dns round robin, load balancers, etc.., but would i have to point the SRV record to one of the asterisk servers, and have him forward the call to the other asterisk server, if the call is for an extension >= 20 ?`
`00:39.32`	`Soul`	`site B would be unavailable if site A would loose its comms to the internet`
`00:39.32`	`[TK]D-Fender`	`Soul : All in your dialplan. In "A", do something like "exten => _20XX,1,Dial(SIP/${EXTEN:2}@ServerB.com)"`
`00:39.49`	`Lee619`	`interesting-- if i put in an invalid username/password for FWD, it shows a state of Rejected for iax2 show registry....`
`00:40.03`	`Lee619`	`but if i put in a valid username/password, it still shows a state of Rejected....`
`00:40.12`	`Soul`	`tk, but then site B would be unavailable if site A would loose its comms to the internet, correct ?`
`00:40.12`	`Darwin35`	`got it`
`00:40.24`	`watchy`	`i aint having no luck finding a site with all config examples of a cisco 7960`
`00:40.26`	`Lee619`	`i'm SURE i'm using the right username/password, because i can log into freeworlddialup.com using the username/password....`
`00:40.33`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:40.45`	`*** part/#asterisk BasketCase (n=BasktCas@asylum.sanitarium.net)`
`00:40.53`	`Lee619`	`does anybody have any insights...? i am behind NAT....`
`00:40.56`	`[TK]D-Fender`	`Soul : you could have it check to see if the dial failed, then fall back to a PSTN call or whatever else you felt like doing...`
`00:41.08`	`Soul`	`tk, good point`
`00:41.27`	`Soul`	`watchy, please wait`
`00:41.51`	`watchy`	`no prob`
`00:42.04`	`watchy`	`dunno why i cant find any on google`
`00:42.40`	`Soul`	`watchy, what do you want, again ? ;)`
`00:42.46`	`[av]bani`	`http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=c%3DL_Product_C2%26cid%3D1129067594457&pagename=Linksys%2FCommon%2FVisitorWrapper`
`00:42.49`	`[av]bani`	`o.o`
`00:43.19`	`inv_Arp`	`just added iaxtel for 8XX numbers , but my voipjeyt dial out is "exten => _1NXXNXXXXXX" wont that pick up the 800 numbers as well?`
`00:43.33`	`Lee619`	`maybe FWD is down? :-)`
`00:43.40`	`watchy`	`soul: volume`
`00:43.42`	`watchy`	`i`
`00:43.52`	`watchy`	`i'd like to know them all but right now i'm intrested in volume`
`00:44.57`	`Lee619`	`giving up... :-(`
`00:45.16`	`Soul`	`watchy, start here: http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx`
`00:45.27`	`*** join/#asterisk DEGRE40 (n=For@84.4.35.191)`
`00:45.40`	`*** part/#asterisk DEGRE40 (n=For@84.4.35.191)`
`00:46.05`	`watchy`	`ok cool`
`00:46.39`	`watchy`	`haha`
`00:46.41`	`watchy`	`thanks i found it`
`00:46.42`	`watchy`	`i love you`
`00:47.05`	`watchy`	`whats the volume called in it though`
`00:49.33`	`inv_Arp`	`just added iaxtel for 8XX numbers , but my voipjet dial out is "exten => _1NXXNXXXXXX" wont that pick up the 800 numbers as well?`
`00:49.48`	`watchy`	`wierd soul. i don't see one for volume`
`00:49.54`	`Soul`	`me neither ;)`
`00:50.11`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:50.18`	`infinity1`	`i have an odd problem where someone will be on the phone and suddenly i can hear them, but they can't hear me.`
`00:50.18`	`*** join/#asterisk cnet2 (n=jjohn@201.192.107.58)`
`00:50.18`	`watchy`	`you sure it exist?`
`00:51.29`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:52.21`	`cnet2`	`hi, I asterisk answering my phone (s,1,Answer..), but i want asterisk to wait for me to dial an extension to tell himwhat to do, but even though i have a exten=>XXX,n,Dial(.., asterisk won't wait for me to dial the numbers and just sends me a hangup.`
`00:53.42`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`00:54.00`	`Soul`	`watchy, sorry, got confused with dtmf volume level. no, never configured call volume level in my configs`
`00:55.24`	`Soul`	`tk: http://www.vovida.org/applications/downloads/loadbalancer/`
`00:55.44`	`Soul`	`this should solve the problem we were talking about, right ?`
`00:56.38`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`00:58.19`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`00:59.57`	`*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)`
`01:02.07`	`Sedorox`	`:p`
`01:04.37`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:04.45`	`*** part/#asterisk sivana (n=sivana@mixdown.ca)`
`01:04.45`	`*** join/#asterisk sivana (n=sivana@mixdown.ca)`
`01:05.27`	`chiardon`	`Hello`
`01:05.33`	`*** join/#asterisk Tili (i=Tili@202-133-67-78-dialup.sat.net.pk)`
`01:06.03`	`[TK]D-Fender`	`cnet2 : You need to set "autofallthrough=no"`
`01:06.16`	`cnet2`	`great thanks! jej`
`01:06.31`	`chiardon`	`Whats exactly "Notice 4709 . . .avoiding deadlock`
`01:06.49`	`[TK]D-Fender`	`Soul : You still need a path tot he other server. That soludtion doesn't solve the lack of network connectivity.`
`01:06.52`	`chiardon`	`sorry!`
`01:07.34`	`chiardon`	`"Notice 4709 . . .avoiding deadlock"`
`01:07.38`	`*** join/#asterisk annonimous (n=annonimo@dsl-201-133-94-50.prod-infinitum.com.mx)`
`01:07.39`	`Soul`	`tk, i think it does, the loadbalancer "pings" both asterisk servers. even if A is down, B would still be available`
`01:07.56`	`ManxPower`	`chiardon, it's a debugging message. ignore it.`
`01:08.05`	`chiardon`	`yepppppppppp`
`01:08.29`	`Soul`	`what i'm trying to find is if the loadbalancer is capable of sending >= 20 extensions to the B server, and the others to the A server`
`01:08.32`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:08.48`	`[TK]D-Fender`	`Soul : What are the odds that the LOCAL server is down? Load balancing is good for things like termination servers. if the server a phone is reg'd to goes dow so do all phones connected to it.`
`01:08.49`	`chiardon`	`but it is showing just before the *Box Down`
`01:08.55`	`Soul`	`something like policy routing, if you understand network routing`
`01:08.59`	`[TK]D-Fender`	`Soul : Whats your real goal? To bridge 2 offices?`
`01:09.11`	`chiardon`	`Manpower Tnx`
`01:09.50`	`chiardon`	`Manpower where you are?`
`01:09.50`	`inv_Arp`	`just added iaxtel for 8XX numbers , but my voipjet dial out is "exten => _1NXXNXXXXXX" wont that pick up the 800 numbers as well?`
`01:10.03`	`Soul`	`tk, no, connecting the 2 (or more) offices is trivial. i'm looking for the most redundant solution that i can build. if A fails, B must still be alive`
`01:10.09`	`[TK]D-Fender`	`inv_Arp : Change your voipjet then.`
`01:10.16`	`chiardon`	`Manpower UK?`
`01:10.49`	`chiardon`	`Someone from western europe?`
`01:10.49`	`ManxPower`	`I am in Alamaba`
`01:10.56`	`chiardon`	`Hoooooooooppppp`
`01:11.04`	`inv_Arp`	`ok lets try regexp fashion`
`01:11.48`	`Soul`	`i read something a few days ago, about some new asterisk solution that could make several asterisk servers behave as one, even that they would be distributed throughout the world. i cant find the url :(`
`01:12.07`	`[TK]D-Fender`	`Soul : Again though what is your goal?`
`01:12.22`	`annonimous`	`hello`
`01:12.26`	`ManxPower`	`One of my big fantasies is for two asterisk servers to act as one.`
`01:13.24`	`Soul`	`tk, if i can create a "virtual" asterisk for the company, with the 2 real asterisk servers, then probably i could divert calls to each office using that virtual server. the virtual server could be in a redundant datacenter`
`01:13.35`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:13.56`	`Soul`	`if location A is down, location B would still get calls, forwarded by the datacenter`
`01:15.01`	`[TK]D-Fender`	`Soul : Thats a big undertaking and requires that the phones double-register or something and that all common resources (like VM) be shared somehow. One idea might be that this is stored in a DB but that adds a central point of failure as well...`
`01:15.02`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:15.19`	`[TK]D-Fender`	`Soul: Do you really need this?`
`01:15.31`	`Soul`	`tk, i can guarantee the datacenter wont fail, but not the offices`
`01:15.50`	`Soul`	`tk, just brainstorming the best solution`
`01:16.24`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:16.48`	`Soul`	`tk, something like sip reality: http://www.voip-info.org/wiki/view/SIP+Reality`
`01:16.54`	`Soul`	`Some unique features are:`
`01:16.54`	`Soul`	`<PROTECTED>`
`01:17.14`	`Soul`	`thats the url i was looking for`
`01:18.34`	`justinu`	`looks like vaporware to me`
`01:18.45`	`[TK]D-Fender`	`Soul : But do you really NEED it?`
`01:19.14`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:19.53`	`Soul`	`tk, everyone needs reliability. should i ask that to the 25 employees at site B, when they cant receive calls because site A is down ?`
`01:20.22`	`Soul`	`justinu, interested in building, not buying. just trying to figure how it works, IF if works`
`01:20.32`	`[TK]D-Fender`	`Soul : Does site B have no lines of their own?`
`01:20.49`	`Soul`	`tk, just internet access`
`01:21.18`	`Soul`	`the point is to forget about tdm and go voip all the way`
`01:21.57`	`[TK]D-Fender`	`Soul : If they only have internet access, and thats it, and the net goes down what on earth do you expect to do with that situation? There is simply NO path to Site B. period. All the phones over there are dead in the water.`
`01:22.28`	`Soul`	`tk, no, thats not the situation i was asking about`
`01:22.47`	`Soul`	`site B should be fully operational even if site A was down`
`01:22.47`	`[TK]D-Fender`	`Soul : try again and make the sample as linear as possible`
`01:22.58`	`Soul`	`tk: site B should be fully operational even if site A was down`
`01:22.59`	`[TK]D-Fender`	`Site "A" has the incoming lines, correct?`
`01:23.21`	`Soul`	`tk, no incoming pstn lines, everything is voip`
`01:23.31`	`Soul`	`site a has internet access, and site b also`
`01:23.41`	`[TK]D-Fender`	`Soul : Do both A & B have their own accounts?`
`01:23.44`	`justinu`	`you can do stuff like that, but you need top grade IP connectivity`
`01:23.52`	`Soul`	`site b must work even if site a is down, and the opposite`
`01:24.16`	`Soul`	`justinu, if i had that i would not worry about comms being down ;)`
`01:24.22`	`Soul`	`tk, yes`
`01:24.28`	`sivana`	`Soul: site a and b have *?`
`01:24.35`	`Soul`	`sivana, yes`
`01:24.58`	`Soul`	`tk, the problem is that site a users must sometimes go work at site b, and the opposite`
`01:24.59`	`sivana`	`I don't see the problem then`
`01:25.00`	`[TK]D-Fender`	`Soul : With a server on each side have its phones register to it, they are independant. The only thing you could lose is access to resources at the other side.`
`01:25.28`	`*** join/#asterisk ManxPowe (i=ewieling@62.sub-70-197-11.myvzw.com)`
`01:25.29`	`Soul`	`tk, yes, if they work as 2 standalone asterisk servers, BUT:`
`01:25.31`	`[TK]D-Fender`	`Soul : thats what forwarding your calls to the other server is for....`
`01:26.27`	`Soul`	`tk, how can YOU, tk, call the sergio@3gnt.net sip url, if the 3gnt.net sip srv record is JUST ONE of those asterisk servers ?`
`01:26.38`	`file`	`o... m... g...`
`01:26.41`	`sivana`	`lol`
`01:26.55`	`*** join/#asterisk kino5 (n=l@adsl-68-107-192-81.adsl.iam.net.ma)`
`01:26.58`	`*** part/#asterisk annonimous (n=annonimo@dsl-201-133-94-50.prod-infinitum.com.mx)`
`01:27.05`	`kino5`	`hi`
`01:27.28`	`kino5`	`how to forwad incoming call to extention?`
`01:27.41`	`file`	`why don't you just deploy SER in a cluster configuration for SIP components, use Asterisk for media and PSTN access, and then the phone can register anywhere and hell you can have two phones registered to the cluster`
`01:27.53`	`Soul`	`if the 3gnt.net sip srv record is sip.3gnt.net, located at site A, and site A is down, how can sergio@3gnt.net be reached if sergio@3gnt.net is usually forwarded by asterisk A to asterisk B (i'm a site B user) ?`
`01:28.13`	`Soul`	`file ?`
`01:28.36`	`Soul`	`file, im sure you are righ, but my head is slower than yours`
`01:29.47`	`Soul`	`question a) can you have multiple sip srv records for a domain, each one pointing to different asterisk servers, where different sip users are registered ?`
`01:29.47`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:30.09`	`cnet2`	`i've put "autofallthrough=no ", and still asterisk won't wait for me to dial an extension before hanging up`
`01:30.10`	`Soul`	`question b) if question a is NO, how can we provide an alternative solution ?`
`01:30.39`	`*** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-90-112.msy.bellsouth.net)`
`01:30.41`	`litage`	`if you have 5 asterisk servers and 500 tenants, each with varying #s of extensions, should all tenants be on each asterisk server, or should the 500 tenants be split up amongst the asterisk servers?`
`01:31.28`	`cnet2`	`i've put WaitExten`
`01:31.34`	`file`	`Soul: you can specify multiple ones, they're weighted and if one is down the sip UA will usually try the next one... that is, if they support SRV records`
`01:31.37`	`Soul`	`litage, if all the tenants are known by all asterisk servers, then everyone can register at the server on the location they are working on`
`01:32.13`	`[TK]D-Fender`	`cnet2 : Pastebin your extensions.conf`
`01:33.20`	`cnet2`	`what-s the paste bin url?`
`01:33.22`	`*** join/#asterisk Skkip (n=Skipper@216.160.91.91)`
`01:33.46`	`[TK]D-Fender`	`~pb`
`01:33.47`	`jbot`	`i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`01:33.52`	`Soul`	`file, ok, thats a good start for an answer to question a). but i suppose the multiple sip srv records point to different sip (asterisk) servers where EVERYONE is registered, correct ? i mean, with sip srv records you just can't say that the 1 2 and 3 users are registered with sip.3gnt.net, and 4 5 and 6 users are registered with sip2.3gnt.net, correct ?`
`01:34.08`	`file`	`Soul: ...no`
`01:34.24`	`*** join/#asterisk greendisease (n=jack@fedora/greendisease)`
`01:34.31`	`file`	`Soul: you're not going to do load balancing and failover of stuff in the SIP protocol on the DNS layer... just no`
`01:34.49`	`Soul`	`ok`
`01:35.09`	`*** join/#asterisk EvilMetal (n=StorM@stardust.noc.frontier.fr)`
`01:35.37`	`litage`	`Soul: would each asterisk server not become sluggish though if the # of tenants significantly increased, say to 50,000?`
`01:36.18`	`*** join/#asterisk chalco_lab (n=chatzill@pdpc/supporter/active/chalco)`
`01:36.20`	`Soul`	`starting with that "no" assumption, then we must have ALL the users for ALL the offices in ALL the asterisk servers (that would take care of the romaing users situation). and then, we must have some way to forward the call to the proper asterisk server where the user is registered in that moment`
`01:36.28`	`ptiggerdine`	`cluster of asterisk server then`
`01:36.31`	`litage`	`file: ?`
`01:36.32`	`*** join/#asterisk jyukes (n=jameshot@pool-138-89-211-251.atc.east.verizon.net)`
`01:36.39`	`Soul`	`otherwise, we could just.. dial all the asterisk servers, like tk said, correct ?`
`01:36.40`	`file`	`litage: you wouldn't get that many on a box`
`01:37.14`	`Soul`	`litage, we're talking maximum 200 users offices`
`01:37.16`	`file`	`Soul: I'll give you two hints for an idea I have in my idea... regexten, and DUNDi`
`01:37.24`	`file`	`er in my head`
`01:37.30`	`Soul`	`file, dont know the first`
`01:37.53`	`file`	`Soul: it modifies the dialplan and adds a 1 priority with noop, so an extension becomes active upon registration`
`01:38.16`	`Soul`	`file, you sip invite sergio@3gnt.net. dns resolves 3gnt.net sip servers to sip.3gnt.net, sip2.3gnt.net, sip3.3gnt.net`
`01:38.27`	`Soul`	`sip.3gnt.net is down (office A is down)`
`01:38.27`	`cnet2`	`[TK]D-Fender>: http://pastebin.com/501848`
`01:38.34`	`chalco_lab`	`hello all. this may not directly apply to asterisk, but hopefully someone can point me in the right direction. I'm trying to find out how a VOIP service provider interrconnects with the PSTN`
`01:38.54`	`chalco_lab`	`*interconnects`
`01:38.55`	`file`	`chalco_lab: they're called telephone companies...`
`01:39.02`	`file`	`or other VoIP carriers`
`01:39.03`	`Soul`	`the call goes to sip2.3gnt.net, (location B), and asterisk B is configured to dial sergio@A, sergio@B and sergio@C at the same time`
`01:39.21`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`01:39.32`	`[TK]D-Fender`	`cnet2 : Ok where in there is your IVR that fails?`
`01:39.47`	`cnet2`	`the default context`
`01:40.05`	`chalco_lab`	`file: a client of mine wants to become a VOIP service provider, and I'm researching it for him`
`01:40.06`	`cnet2`	`it answers, and seems its waiting for exten, but when i press any number i get Invalid Extension`
`01:40.15`	`Soul`	`sergio@A will obviously not dial (A is down), sergio@B will not ring (sergio is not registered there, he is 300 miles away), sergio@C will ring, and voila, i will answer. is this feasible ?`
`01:40.24`	`file`	`chalco_lab: you get a connection to the regular phone network, a PRI or DS3 or whatever...`
`01:40.29`	`file`	`chalco_lab: from the telco`
`01:40.33`	`enemy^x`	Just tried out Asterisk-IM with spark as client, Seems like I have to update the status message on my side to anything before the others see that I`m on the phone.... ?
`01:40.57`	`file`	`Soul: depends if you used voicemail because sergio@B has the potential to pick up if it does`
`01:41.07`	`[TK]D-Fender`	`cnet2 : exten => XXX,n,Dial(IAX2/powersol/${EXTEN}) is no good. you need a priority 1!`
`01:41.11`	`Soul`	`file, damn ;)`
`01:41.14`	`chalco_lab`	`file: thank you. that helps a lot`
`01:41.15`	`[TK]D-Fender`	`exten => XXX,1,Dial(IAX2/powersol/${EXTEN})`
`01:41.32`	`Soul`	`file, how to solve that ?`
`01:42.05`	`file`	`SOul: I'm not going to solve all your problems for you`
`01:42.24`	`cnet2`	`[TK]D-Fender: ok i did that, but it stills won't let me dial more than 1 number`
`01:42.28`	`Soul`	`file, ;)`
`01:43.03`	`[TK]D-Fender`	`cnet2 : And get rid of Waitexten, and add in exten => s,2,Set(TIMEOUT(response)=15) and exten => s,3,Set(TIMEOUT(digit)=3)`
`01:43.15`	`cnet2`	`ok`
`01:43.21`	`[TK]D-Fender`	`Actually that should be : exten => _XXX,1,Dial(IAX2/powersol/${EXTEN})`
`01:43.26`	`[TK]D-Fender`	`yuo forgot the "_" too....`
`01:43.45`	`[av]bani`	`[TK]D-Fender: another point for gxp2000: it can do intercom without having to use a separate autoanswer extension hack`
`01:43.51`	`[TK]D-Fender`	`Ok, run with that for a bit, I'm off to watch a movie`
`01:43.56`	`[av]bani`	`too bad the speakerphone is so bad :P`
`01:44.15`	`Soul`	`someking of "dynamic" dialplan, built with information from the multiple asterisk servers, would be great: "if sergio is registered at B or C dont enable his voicemail here"`
`01:44.15`	`[TK]D-Fender`	`is the GXP any less of a hack than Poly really?`
`01:44.29`	`[av]bani`	`poly requires autoanswer extension? the gxp uses a hint`
`01:45.03`	`[TK]D-Fender`	`[av]bani : a hint? Makes no sense, but will catch up later.`
`01:45.26`	`[av]bani`	`exten => 1234,1,SIPAddHeader(Call-Info: answer-after=0)`
`01:45.31`	`[av]bani`	`well, an additional header`
`01:46.12`	`kino5`	`how to forwad incoming call to extention?`
`01:46.19`	`*** join/#asterisk \|omni\| (n=rob@net98.limelyte.net)`
`01:46.25`	`kino5`	`incoming call from PSTN line`
`01:46.42`	`cnet2`	`[TK]D-Fender: set command is not recognized.. :S`
`01:46.58`	`\|omni\|`	`anyone in 509 area code need a PSTN gate? putting a 7 chan PRI in our rack and just need to cover costs`
`01:47.28`	`enemy^x`	`anyone here tried the Asterisk-IM plugin?`
`01:52.08`	`cnet2`	`gotit, thanks`
`01:52.34`	`litage`	`Soul: you and i are trying to achieve the exact same thing. may i privmsg you?`
`01:53.35`	`Soul`	`course`
`01:55.56`	`*** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net)`
`01:58.06`	`enemy^x`	`is it possible to get the message stuff working in xten with asterisk? chan_sip.c:7283 receive_message: Received message to -....- gets dropped`
`01:59.27`	`*** join/#asterisk Soul (n=Soul@87-196-8-134.net.novis.pt)`
`02:00.42`	`*** join/#asterisk rbrookshie (i=matt@69.247.184.46)`
`02:09.02`	`*** join/#asterisk denon (i=denon@synapse.subneural.net)`
`02:09.02`	`*** mode/#asterisk [+o denon] by ChanServ`
`02:10.09`	`litage`	`file: so if you have 1,000 tenants, each with varying #s of extensions, it's not feasible to put all tenants on each * box?`
`02:10.40`	`justinu`	`too many simultaneous registers will crash asterisk :P`
`02:10.45`	`[av]bani`	`\o/`
`02:12.19`	`litage`	`justinu: "too many" like 20 or 100 or 1000 simultaneous registrations?`
`02:12.42`	`justinu`	`around 100, iirc`
`02:13.05`	`Soul`	`justinu, not here, not even close`
`02:13.10`	`litage`	`justinu: if you split that into 2 groups of 50 registrations that occured consecutively, would things be peachy?`
`02:13.24`	`justinu`	`the solution is to have your UA's register with SER`
`02:13.46`	`justinu`	`soul: what do you mean?`
`02:13.52`	`justinu`	`soul: you're not having that problem?`
`02:14.43`	`Soul`	`justinu, you mean 100 SIP REGISTER operations at the same time, or 100 users registered at the same time, (but the REGISTER operation happened before, at different times) ?`
`02:14.45`	`*** join/#asterisk EvilMetal (n=StorM@stardust.noc.frontier.fr)`
`02:15.14`	`justinu`	`100 sip register operations`
`02:15.32`	`Soul`	`justinu, ah, sorry, never had that experience`
`02:15.36`	`justinu`	`like for example, if your link went down, and then came back up, all the UAs will register`
`02:15.58`	`litage`	`justinu: i haven't read much on how SER works, but for registrations to take place with a SER box, SER would need to know the username and password for each party trying to register, right? and upstream * boxes also need to have that same registration information too, right?`
`02:16.13`	`Soul`	`justinu, correct, in that case we had that experience several times a day, for a month. no probs`
`02:16.59`	`justinu`	`the * boxes just need to know the SIP AOR`
`02:17.06`	`justinu`	`only the phones need the authentication info`
`02:17.27`	`litage`	`justinu: SIP AOR?`
`02:17.33`	`justinu`	`SER can be setup to auth against a database`
`02:17.36`	`justinu`	`address of record`
`02:17.56`	`Soul`	`justinu, yes, ser is much better. also too complicated.`
`02:18.17`	`justinu`	`SER is very complicated at all`
`02:18.22`	`justinu`	`much less so than asterisk`
`02:20.23`	`Soul`	`justinu, you mean ser is simple ?`
`02:20.52`	`litage`	`file, justinu: so if you have 1,000+ tenants, each with varying #s of extensions, and use SER to handle registrations, is it feasible to put all tenants on each * box?`
`02:23.16`	`inv_Arp`	`Qwell: around?`
`02:27.54`	`watchy`	`for music on hold whats a good streamer to use`
`02:28.05`	`watchy`	`for shoutcast?`
`02:28.35`	`Soul`	`watchy, we're using mpg123`
`02:28.54`	`watchy`	`hrm`
`02:28.59`	`watchy`	`not workin for me g`
`02:29.07`	`watchy`	`THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!`
`02:29.07`	`watchy`	`HTTP request failed: 404 Resource Not Found`
`02:29.11`	`Soul`	`pick another stream, most of themdont work`
`02:29.14`	`watchy`	`any special flags you give it?`
`02:29.20`	`watchy`	`if you give it a url?`
`02:29.27`	`Soul`	`yeah`
`02:30.37`	`watchy`	`which?`
`02:32.24`	`*** part/#asterisk Prival (i=user69@MTL-HSE-ppp207971.qc.sympatico.ca)`
`02:32.35`	`*** join/#asterisk Prival (i=user69@MTL-HSE-ppp207971.qc.sympatico.ca)`
`02:32.39`	`*** part/#asterisk Prival (i=user69@MTL-HSE-ppp207971.qc.sympatico.ca)`
`02:33.58`	`Soul`	`no clue, not in the office right now`
`02:34.28`	`watchy`	`ah`
`02:35.47`	`*** join/#asterisk smallb (n=smallb@prox47-249.trinidad.net)`
`02:35.57`	`smallb`	`hello`
`02:37.58`	`ObsidianX`	`hey folks, if im trying to setup a soft-phone like Kiax or MozIAX to connect to asterisk to only receive calls would i choose friend, user, or peer`
`02:38.24`	`marcus2`	`user`
`02:38.26`	`ObsidianX`	`i keep on getting "Inappropriate authentication received"`
`02:38.37`	`marcus2`	`that error has nthing to do with friend/user/peer tho`
`02:38.40`	`*** join/#asterisk linlin (i=linlin@c-67-184-231-233.hsd1.il.comcast.net)`
`02:38.45`	`ObsidianX`	`true`
`02:39.02`	`ObsidianX`	`when i choose user it says "No registration for peer 'test'"`
`02:39.53`	`ObsidianX`	`although i have a section [test] with secret=pass etc...`
`02:40.01`	`marcus2`	`do you have auth=md5 ?`
`02:40.28`	`ObsidianX`	`i just added it and it still doesn't work`
`02:41.01`	`ObsidianX`	`md5,plaintext,rsa doesn't work either`
`02:41.04`	`*** join/#asterisk NDT (n=me@cpe-24-194-166-119.nycap.res.rr.com)`
`02:47.51`	`Nugget`	`maybe "inappropriate" means you should put some clothes on or something.`
`02:48.32`	`*** join/#asterisk tengulre11 (n=tengulre@61.185.224.66)`
`02:50.05`	`*** join/#asterisk brockj49464 (n=brockj49@63.87.56.159)`
`02:54.24`	`*** join/#asterisk hhoffman (n=hhoffman@tor/session/x-ffdbc31ebc3f095f)`
`02:54.53`	`hhoffman`	`hi, is anyone using zasterisk?`
`02:57.56`	`ObsidianX`	`Nugget: heheh`
`02:58.06`	`ObsidianX`	`marcus2: any ideas?`
`03:02.07`	`*** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)`
`03:02.18`	`*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)`
`03:02.43`	`shmaltz`	`anybody here running the following:`
`03:02.45`	`shmaltz`	`asterisk 1.2.1`
`03:02.46`	`shmaltz`	`sipura`
`03:02.48`	`shmaltz`	`and polycom?`
`03:03.06`	`*** join/#asterisk EvilMetal (n=StorM@sgc91-2-82-237-76-2.fbx.proxad.net)`
`03:04.58`	`shmaltz`	`<PROTECTED>`
`03:05.44`	`*** join/#asterisk jef_ (i=fischer@p548466C5.dip.t-dialin.net)`
`03:11.47`	`*** join/#asterisk Cyon (n=cyon@cyons.net)`
`03:12.15`	`shmaltz`	`<PROTECTED>`
`03:12.21`	`Cyon`	`whos there?`
`03:12.39`	`shmaltz`	`hi`
`03:12.41`	`ObsidianX`	`"No registration for peer" agh`
`03:12.44`	`ObsidianX`	`what does that mean :(`
`03:13.27`	`*** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)`
`03:13.46`	`brockj49464`	`Anybody have any GXP-2000 to sell? Or reasons not to look at getting that phone?`
`03:13.53`	`Qwell`	`brockj49464: because they suck`
`03:14.02`	`Qwell`	`especially a used one...`
`03:14.09`	`_Sam--`	`i dont agree personally`
`03:14.17`	`_Sam--`	`i just installed 12 of them today for a real estate office`
`03:14.22`	`brockj49464`	`qwell: What exactly is weong with them?`
`03:14.26`	`_Sam--`	`for what they are...they are pretty good units.`
`03:14.29`	`Qwell`	`_Sam--: give them my condolences`
`03:14.47`	`_Sam--`	`i run my business on them, we have almost 20 people using them at my office as well`
`03:14.50`	`brockj49464`	`sam: Can they do on-hook anouncements (paging)?`
`03:15.16`	`_Sam--`	`i thikn the newest beta firmware does that.`
`03:15.19`	`_Sam--`	`finally`
`03:15.29`	`_Sam--`	`there is a wiki page about the phones that has some decent info`
`03:15.42`	`_Sam--`	`i dont know what else to compare them to for 85 bucks`
`03:15.54`	`Qwell`	`_Sam--: a GOOD headset, and a softphone`
`03:15.56`	`_Sam--`	`i am not saying you will love yours...but mine work fine for the role they are in`
`03:16.05`	`_Sam--`	`they blow away softphones`
`03:16.11`	`_Sam--`	`my sales guys switched from softphones to that`
`03:16.31`	`_Sam--`	`and we used good plantronics headsets`
`03:16.33`	`*** join/#asterisk FastJack (i=fastjack@p5091FE1E.dip.t-dialin.net)`
`03:16.44`	`_Sam--`	`i dont know what problems you had with the phones qwell`
`03:17.11`	`_Sam--`	`but ive dealt with their tech support as well which was refreshingly helpfuly...got through to somoeone right away who helped me out`
`03:17.14`	`brockj49464`	`I am looking at them for home. Trying to replace a pansonic kxtd1232 before it is worthless`
`03:17.28`	`ObsidianX`	`anybody know whats up with this error?`
`03:18.00`	`_Sam--`	`we are testing out the beta version of their newest firmware`
`03:18.06`	`_Sam--`	`and it seems pretty good for us`
`03:20.38`	`brockj49464`	`That is good that they seem to work.`
`03:20.49`	`_Sam--`	`ymmv based on your setup`
`03:21.08`	`_Sam--`	`all of my stuff ive been setting up is 100% ...no pri or pstn type stuff`
`03:21.20`	`_Sam--`	`er 100% voip`
`03:21.30`	`Qwell`	`ugh`
`03:21.39`	`\|omni\|`	`using remote gateways?`
`03:21.43`	`Qwell`	`realestate agents get MAD when things don't work`
`03:21.43`	`_Sam--`	`noope`
`03:21.58`	`_Sam--`	`well yeah , their asterisk box connects to an IAX provider`
`03:22.03`	`_Sam--`	`i guess that is a remote gateway....`
`03:22.15`	`\|omni\|`	`heh...I was just working on a system for a real estate office a couple weeks ago with someone`
`03:22.17`	`_Sam--`	`but the people assume the risks knowingly`
`03:22.18`	`iCEBrkr`	`damnit this phone number`
`03:22.25`	`iCEBrkr`	`I got some fucker calling me twice a day`
`03:22.36`	`Qwell`	`_Sam--: So, you told them to only expect 90% uptime?`
`03:22.40`	`iCEBrkr`	`I think it's Walmarts telemarketing/survey group`
`03:22.50`	`_Sam--`	`ive been running 100% voip at my business for about 1.2 years...`
`03:22.56`	`_Sam--`	`our uptime is closer to 99% for our calls`
`03:23.03`	`Qwell`	`99% is unacceptable`
`03:23.09`	`_Sam--`	`maybe for some high end clients`
`03:23.14`	`_Sam--`	`but based on budgets`
`03:23.14`	`Qwell`	`for anybody`
`03:23.22`	`_Sam--`	`they assume the risks`
`03:23.23`	`iCEBrkr`	`Five 9's!`
`03:23.25`	`_Sam--`	`they know`
`03:23.31`	`_Sam--`	`we talk about options`
`03:23.37`	`_Sam--`	`they choose based on cost`
`03:23.39`	`Qwell`	`99%...do you realize what that equates to?`
`03:23.39`	`\|omni\|`	`same on this side, but when I do a lot of forwarding (bounce exten to cell or whatever) I like low latency PSTN if possible`
`03:23.51`	`Qwell`	`1 hour every 4 days`
`03:23.59`	`Qwell`	`That is A LOT`
`03:24.04`	`Qwell`	`completely unacceptable`
`03:24.17`	`_Sam--`	`my shit works fine...i run a mail order business that over 10 mil a year in sales on it`
`03:24.21`	`_Sam--`	`and its acceptable just fine`
`03:24.32`	`_Sam--`	`you dont have to like it, thats fine`
`03:24.37`	`_Sam--`	`but people do`
`03:24.42`	`Qwell`	`_Sam--: So, what if UPS only delivered 4 days a week?`
`03:24.46`	`iCEBrkr`	`Qwell: What if you have 72hrs downtime in the month of Dec?`
`03:24.47`	`Qwell`	`You'd be freaking pissed`
`03:24.51`	`_Sam--`	`my phones deliver 7 days a week`
`03:24.53`	`Qwell`	`iCEBrkr: indeed`
`03:25.02`	`_Sam--`	`what is the difference between my PTP t1 and a PRI?`
`03:25.03`	`_Sam--`	`nothing`
`03:25.05`	`iCEBrkr`	`Qwell: Your average doesn't hold water, is all I'm saying :P`
`03:25.08`	`Qwell`	`iCEBrkr: on the 20th, 21st, and 22nd`
`03:25.25`	`_Sam--`	`so unless a route is down on my 8 homed provider...`
`03:25.30`	`_Sam--`	`the chances that i cant get there are pretty bad`
`03:25.32`	`iCEBrkr`	`...and hardware PBX's go dead a lot too..`
`03:25.33`	`_Sam--`	`my shit works.`
`03:25.44`	`_Sam--`	`call it as many times as you want..i'll give ya the number`
`03:26.03`	`Cyon`	`Hmmm, anyone here messed with getting faxing working?`
`03:26.38`	`\|omni\|`	`Sam...doing a similar setup here but putting a PRI into my rack`
`03:26.45`	`_Sam--`	`i started with a PRI`
`03:26.50`	`_Sam--`	`and switched to a PTP t1`
`03:27.04`	`\|omni\|`	`I have a PTP T1 from my rack to a client endpoint..but not here`
`03:27.10`	`_Sam--`	`and ive never regretted the decision`
`03:27.17`	`\|omni\|`	`low bandwidth for voice here`
`03:27.42`	`shmaltz`	`anybody here have an asterisk 1.2.1 system with a sipura and 2 polycom phones?`
`03:27.58`	`brockj49464`	`what exactly you trying to do with faxing?`
`03:28.36`	`Cyon`	`brockj49464: Get it working? ;-) I've tried the still beta t.38 patch, but unfortunately it's still buggy it would appear and I don't have the skill to update it`
`03:29.15`	`Cyon`	`brockj49464: So I jumped over to ser/openser, bypassing asterisk (I know, bad channel for that.) and tried to get sipura->ser->cisco working...`
`03:29.38`	`brockj49464`	`I am using g711u and seem to not have any problems for the 5 times I have used it this last week.`
`03:30.23`	`Cyon`	`brockj49464: Yeah, I've done ulaw; and can get it working 90%+ ; but I'm aiming for a solid 100%, or at least as close as possible`
`03:30.37`	`Cyon`	`When the customer does hundreds; they really notice that percentage of failures`
`03:31.09`	`*** join/#asterisk loud (n=ariel@cypher.punk.net)`
`03:31.37`	`Cyon`	`_Vile mentioned he does Sipura->ser->cisco, with perfection so far is success rates, so I wanted to give that a try; or get other people's views on it`
`03:32.23`	`brockj49464`	`That is true. My provider was where I was having problems when I used thier settinging on the ATA. When I defaulted it and set it up to my * box I had no problems _so_ far. Time will tell. It also solved my Dish Network problem...`
`03:32.59`	`Cyon`	`brockj49464: What ATA do you have? Just to ask...`
`03:33.04`	`*** join/#asterisk Jameno123 (n=james@63.210.246.146)`
`03:33.21`	`Cyon`	`But yeah, I can get some really solid results; but it's just not consistent enough..unfortunately`
`03:33.43`	`Jameno123`	`http://pastebin.com/501931`
`03:33.48`	`Jameno123`	`anyone have a solution to that?`
`03:34.05`	`Jameno123`	`"inlining failed in call to '__t4_framer_interrupt': function body notavailable"`
`03:34.07`	`brockj49464`	`SPA-2100 Getting 2 more of them. My plan is to start with cheap CID 2500 like phones and move to GXP-2000 as I get wiring and the phones.`
`03:34.29`	`alephcom_`	`I need an opinion from you all... On a low end ($9.99 per month) hosted pbx, do you think the customer needs more than 1 auto attendant?`
`03:34.38`	`Cyon`	`alephcom_: No.`
`03:34.39`	`\|omni\|`	`I'm liking the cisco 7960 for a work handset`
`03:34.53`	`Jameno123`	`\|omni\|, 7940G are great too`
`03:34.54`	`\|omni\|`	`I was on Zultys stuff before which is cool but these Ciscos are pretty nice`
`03:35.01`	`shmaltz`	`nybody here have an asterisk 1.2.1 system with a sipura and 2 polycom phones?`
`03:35.03`	`\|omni\|`	`I haven't tried a 7940 yet`
`03:35.10`	`Cyon`	`shmaltz: Sipura, but not polycom`
`03:35.14`	`Jameno123`	`7940/7960 same phone, just lesser phone "lines"`
`03:35.17`	`Jameno123`	`and cheaper price ;)`
`03:35.22`	`shmaltz`	`Cyon, what other phones?`
`03:35.28`	`\|omni\|`	`not as many appearances`
`03:35.28`	`Cyon`	`shmaltz: snom`
`03:35.30`	`alephcom_`	`Cyon: Tks, my thoughts too. I'm just designing an automated signup/management setup and I'm having lots of fun on the dialplan.`
`03:35.35`	`\|omni\|`	`how many does the 40 have.... 4?`
`03:35.41`	`Jameno123`	`2`
`03:35.45`	`\|omni\|`	`same XML mini-browser, etc.?`
`03:35.47`	`shmaltz`	`Cyon, so you have snom, sipura, and 1.2.1?`
`03:35.50`	`Jameno123`	`\|omni\|, yes`
`03:35.53`	`\|omni\|`	`sweet`
`03:35.54`	`Jameno123`	`same lcd, ect`
`03:35.59`	`Cyon`	`alephcom_: Yeah, I've been working on the same, with the auto-attendant being the hardest for me by far`
`03:36.01`	`\|omni\|`	`I setup some cool little apps on our PBX for the phone`
`03:36.03`	`Cyon`	`shmaltz: Yes`
`03:36.11`	`Jameno123`	`\|omni\|, any of them use the LCD?`
`03:36.15`	`shmaltz`	`Cyon, more than one snom? or just one?`
`03:36.24`	`\|omni\|`	`yea, browse to the app in LCD and submit data`
`03:36.32`	`\|omni\|`	`just simple stuff testing out the Cisco XML layout`
`03:36.35`	`Cyon`	`shmaltz: Just one for testing; have lots in stock for customers; why?`
`03:36.44`	`Cyon`	`shmaltz: Just ask whatever it is`
`03:36.50`	`\|omni\|`	`enter zip and get weather info, or lookup directory info`
`03:36.58`	`shmaltz`	`Cyon, I'm trying to test something, to see who has the bug: asteirsk, polycom, or sipura`
`03:37.01`	`\|omni\|`	`but the wheels are turning now`
`03:37.12`	`Cyon`	`brockj49464: I'll get it eventually, I'm just sure others have done it already`
`03:37.16`	`Cyon`	`shmaltz: What bug?`
`03:37.19`	`Jameno123`	`\|omni\|, yea, i was looking on trying to figure out how to present customer order data`
`03:37.24`	`*** join/#asterisk anonymouz666 (n=anonymou@gw.ideavalley.com.br)`
`03:37.28`	`Jameno123`	`cust calls in, the order# is shown on the phone when the agent answers`
`03:37.29`	`shmaltz`	`Cyon, I have a problem with sipura asterisk 1.2.1 and polycoms, I know it's a bug, but I'm not sure who is at fault`
`03:37.52`	`shmaltz`	`Cyon, when a polycom speaks with a sipura, and then does an attended xfer to anohter polycom, at the final stage there is only 1 way audio`
`03:38.09`	`shmaltz`	`this is on a single flat network, 1 subnet`
`03:38.10`	`anonymouz666`	`hi... there is a caller in a queue.. I think its crashed because his wait time: (wait: -525351:-37, prio: 0)`
`03:38.11`	`shmaltz`	`no nat`
`03:38.16`	`Jameno123`	`the only way so far ive figured out is just to throw the order# in the callerid info heh`
`03:38.18`	`anonymouz666`	`how do I remove this one?`
`03:38.29`	`Jameno123`	`sooooooo - does anyone have a solution to that? http://pastebin.com/501931`
`03:38.40`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`03:38.41`	`shmaltz`	`if I change the sipura to canreinvite=no, then everything is ok, but another problem arises`
`03:38.59`	`\|omni\|`	`Jameno123: like enter order number and get details?`
`03:39.08`	`*** join/#asterisk HolyGod (i=nobody@got.securebinary.com)`
`03:39.31`	`shmaltz`	`Jameno123, what version of zap? and what version of kernel?`
`03:39.36`	`\|omni\|`	`pretty simple to write little apps, we've done a ton of web development in the past so I just wrote a little PHP that dumps results to the Cisco XML elements and it works pretty well..pull from DB or whatever`
`03:39.45`	`anonymouz666`	`is it possible to remove callers crashed from a queue?`
`03:40.06`	`Jameno123`	`shmaltz, zap=latest, kernel=2.6.12(+patches)`
`03:40.12`	`Cyon`	`shmaltz: Hmmm, beyond me`
`03:40.22`	`Jameno123`	`just freshly downloaded from SVN about an hour ago`
`03:40.26`	`\|omni\|`	`I'd like to play with some outlook integration`
`03:40.50`	`shmaltz`	`Cyon, but if you could test this for me with the snoms then it would confirm that:`
`03:40.52`	`shmaltz`	`1. its not the sipuras,`
`03:40.53`	`shmaltz`	`2. It's not asterisk`
`03:41.12`	`shmaltz`	`Jameno123, which one from svn? tags or trunk?`
`03:41.16`	`Jameno123`	`trunk`
`03:41.34`	`Cyon`	`shmaltz: I can test it at the office tomorrow; but we used it extensively; only way it would replicate is if we did snom->sipura->snom`
`03:41.41`	`Jameno123`	`shmaltz, (gcc 4.0.1)`
`03:41.46`	`Cyon`	`shmaltz: Other than that, we never ean into it`
`03:41.49`	`Cyon`	`*ran`
`03:42.23`	`Cyon`	`shmaltz: I'm generally here all day; just pm me any time and I'll get on it`
`03:42.31`	`shmaltz`	`Cyon, also if I do canreinvite=no all is godd, so if you test it you will have to make sure that the rtp always gets reinvited`
`03:42.43`	`shmaltz`	`Cyon, Thank you`
`03:42.43`	`*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`03:42.46`	`VeNoMouS_`	`woah i forgot i left this on`
`03:42.46`	`VeNoMouS_`	`lol`
`03:42.56`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`03:43.37`	`Cyon`	`shmaltz: Easily; when I'm at the phones :)`
`03:44.32`	`Jameno123`	`shmaltz: i have no zaptel cards as well.`
`03:44.40`	`Jameno123`	`just trying to install ztdummy`
`03:44.50`	`shmaltz`	`Jameno123, that shouldn't make a difference`
`03:44.58`	`shmaltz`	`this problem is beyond me`
`03:45.33`	`Jameno123`	`<PROTECTED>`
`03:45.42`	`Jameno123`	`static inline void __t4_framer_interrupt(struct t4 *wc, int span);`
`03:45.43`	`Jameno123`	`wtf`
`03:45.54`	`Jameno123`	`heh, no function body, as it says.`
`03:46.04`	`*** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it)`
`03:46.21`	`*** join/#asterisk nutria (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)`
`03:47.57`	`Jameno123`	`kinda looks outta place, guess it needs moved up to the top of the file :( though im not a C expert, have no idea what im talking about.`
`03:49.35`	`Cyon`	`Hmmm, does anyone recall an issue where a call tries to use speex when neither side of the sip headers support it; and then it has no trnslation path and the call dies?`
`03:50.09`	`*** join/#asterisk bmg505 (n=leon@c1-61-9.rndf.isadsl.co.za)`
`03:50.14`	`dily`	`hi@all`
`03:50.49`	`dily`	`i try to compile bristuff-0.3.0-PRE-1c`
`03:51.01`	`dily`	`but when complie the zaphfc.ko i have strange function undefined warning`
`03:51.31`	`dily`	`like this: *** Warning: "zt_register" [/usr/src/bristuff/zaphfc/zaphfc.ko] undefined!`
`03:51.44`	`dily`	`any idea?`
`03:51.59`	`Cyon`	`Never looked at or tried that module`
`03:52.52`	`dily`	`i try to install bristuff on many system/distributions but i have the some errors...`
`03:54.40`	`ObsidianX`	`has anybody ever had an error when setting up IAX along the lines of "No registration for peer 'user'"?`
`03:56.30`	`dily`	`anyone use bristuff?!?`
`03:56.41`	`Cyon`	`Actually, let me ask this way; what is speex (I know it's a codec) but how do I totally disable it everywhere? lol`
`03:57.23`	`Cyon`	`Like, why does asterisk say it's trying to be used when talking to the cisco...`
`03:57.48`	`*** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net)`
`03:59.21`	`brockj49464`	`cyon: Do you disallow=all then allow what ones you want to use?`
`03:59.31`	`Cyon`	`brockj49464: absolutely`
`03:59.44`	`Cyon`	`It looks like cisco ignores it and tries to establish calls as speex`
`04:00.22`	`Cyon`	`The only one allowed in my sipuras is ulaw, the only one allowed in asterisk is ulaw, and the cisco has "codec g711ulaw" as well...`
`04:00.47`	`Cyon`	`And yet: [2006-01-11 17:52:24] WARNING[32704]: Unable to find a codec translation path from speex to ulaw`
`04:02.30`	`hhoffman`	`is there a better tts then festival to use with asterisk?`
`04:03.55`	`Cyon`	`http://pastebin.com/501950 <-- anyone have any ideas?`
`04:04.01`	`Cyon`	`hhoffman: Not that I've seen`
`04:07.31`	`ObsidianX`	`http://www.voipuser.org/forum_topic_4196.html`
`04:08.28`	`*** join/#asterisk mud (n=mud@206-248-138-115.dsl.teksavvy.com)`
`04:09.08`	`*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)`
`04:10.08`	`fugitivo`	`hhoffman: www.cepstral.com`
`04:11.45`	`Jameno123`	`Cyon, try "disallow=all" "allow=ulaw"`
`04:11.59`	`Jameno123`	`hrm, nobody has any ideas about my issue?`
`04:12.21`	`Jameno123`	`http://pastebin.com/501931`
`04:13.00`	`dily`	`http://www.loquendo.com/regional_preferences.htm`
`04:13.17`	`Cyon`	`Jameno123: Was done long ago`
`04:13.32`	`Cyon`	`Jameno123: speex isn't even a protocol that asterisk has by default`
`04:13.44`	`file`	`s/protocol/codec`
`04:14.18`	`Cyon`	`Jameno123: Something is trying to use it, or makes asterisk think it is; yet cisco doesn't support that codec either it would appear, and my sipura is set to use g711, and pref. codec only.`
`04:14.27`	`Cyon`	`file: Sorry, yes.`
`04:15.24`	`hhoffman`	`fugitivo: thanks checking now`
`04:15.41`	`Jameno123`	`twisted[asteria], wakey wakey!`
`04:16.46`	`hhoffman`	`fugitivo: are these voice compatible with festival?`
`04:17.00`	`Cyon`	`Jameno123: I'm not a coder anymore; but can I see a pastebin of all the verbose/debug lines?`
`04:17.04`	`fugitivo`	`hhoffman: no, it's closed source`
`04:17.07`	`SwK`	`jameno123 is from teh svn or from the 1.2.1 tarball?`
`04:17.18`	`Cyon`	`Jameno123: So I can see which src files it is bouncing through`
`04:17.34`	`SwK`	`it looks like a bad check out from svn`
`04:17.55`	`hhoffman`	`fugitivo: k, thx`
`04:17.57`	`Jameno123`	`SwK, svn, ive deleted and redownloaded twice now.`
`04:18.14`	`SwK`	`it looks like 1/2 and update to me`
`04:18.23`	`hhoffman`	`ah, but I'm guessing it's meant to work with * as they have digium links on their page`
`04:18.25`	`SwK`	`are you running head?`
`04:18.31`	`SwK`	`(or trunk now)`
`04:18.36`	`Jameno123`	`SwK, trunk`
`04:18.45`	`Jameno123`	`ive always ran CVS-HEAD`
`04:18.57`	`Cyon`	`Jameno123: Ah, I assumed it was the tgz download...`
`04:19.07`	`SwK`	`i did to til 1.2.X was released`
`04:19.17`	`SwK`	`1.0 was just to damned old and missing too many features`
`04:19.23`	`*** join/#asterisk santiago (n=santiago@208.195.215.97)`
`04:19.35`	`Qwell`	`I run svn roots`
`04:19.37`	`SwK`	`I would try compiling the 1.2.1 zap sources from the tarball and see what happens`
`04:19.48`	`Qwell`	`more features than trunk`
`04:20.41`	`Jameno123`	`hrm`
`04:20.59`	`Jameno123`	`will try`
`04:21.58`	`*** join/#asterisk Hybrid (n=hybridra@calera-47.cher.brightok.net)`
`04:22.26`	`Jameno123`	`SwK, yea, the "out of date" stuff, is what concerns me ;)`
`04:22.51`	`SwK`	`i wouldnt worry about it rightnow`
`04:24.36`	`*** part/#asterisk santiago (n=santiago@208.195.215.97)`
`04:25.15`	`tainted_`	`how do i do E911 for a client?`
`04:25.32`	`Jameno123`	`SwK, waiting on the box to rebewt, i guess we'll see :)`
`04:25.35`	`SwK`	`very carefully`
`04:25.44`	`SwK`	`tainted_ are you an ITSP?`
`04:25.57`	`tainted_`	`SwK it's for a client`
`04:26.04`	`*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)`
`04:26.15`	`tainted_`	`i don't normally do this kind of stuff`
`04:26.40`	`Cyon`	`Jameno123: Reboot? Why woyld you reboot?`
`04:27.01`	`Cyon`	`s/oy/ou`
`04:28.41`	`Jameno123`	`Cyon, ;) kernel updates`
`04:28.50`	`Cyon`	`Jameno123: Ah, ok. :)`
`04:29.09`	`Jameno123`	`would be so nice to`
`04:29.17`	`Jameno123`	`cat newkernel > /proc/kcore`
`04:29.20`	`Jameno123`	`and not have to reboot ;)`
`04:29.25`	`Jameno123`	`but i dont think we'll see the day`
`04:29.28`	`SkramX`	`heh, it would.`
`04:29.31`	`Cyon`	`I can't wait till we have dynamic kernel loading...`
`04:29.48`	`Cyon`	`Nah, it's doable; just the entire structure would have to be redone, and it'll be years...`
`04:29.55`	`Cyon`	`But it will happen eventually`
`04:30.03`	`Hybrid`	`Anybody have Mechwarrior 3?`
`04:31.42`	`*** part/#asterisk Hybrid (n=hybridra@calera-47.cher.brightok.net)`
`04:32.00`	`Jameno123`	`SwK, suggest using 1.2.1 [.tgz] completely or just zaptel?`
`04:34.40`	`SwK`	`1.2.1 zap shoudl work with trunk at this time,altho i'm not sure... 1.2.1 would probably be better for products as its a known quantity and its not missing much from trunk yet (unless there is something in trunk you really need)`
`04:36.24`	`Jameno123`	`swk it built properly ;) heh, it should run then`
`04:37.06`	`Jameno123`	`hah`
`04:37.07`	`Jameno123`	`yay!`
`04:37.12`	`Jameno123`	`<PROTECTED>`
`04:37.18`	`Jameno123`	`<PROTECTED>`
`04:37.20`	`fugitivo`	`WIRING WIRING WIRING`
`04:37.22`	`Jameno123`	`heh`
`04:37.34`	`hnupik`	`children`
`04:38.02`	`SwK`	`hah`
`04:38.09`	`SwK`	`it always gripes about g729`
`04:38.45`	`*** join/#asterisk qhrisnd (n=qhrisnd@ppp-71-129-177-185.dsl.irvnca.pacbell.net)`
`04:38.51`	`file[laptop]`	`hahaha...`
`04:38.58`	`qhrisnd`	`Good evening everyone :-)`
`04:38.59`	`file[laptop]`	`my cellphone bill is insane`
`04:39.48`	`Jameno123`	`SwK, hrm, should i rm -rf that and re-make install?`
`04:39.52`	`[TK]D-Fender`	`Perhaps its the 800# attached to it :)`
`04:40.04`	`Qwell`	`Jameno123: It's just a warning...ignore it if that was the only file`
`04:40.05`	`file[laptop]`	`wait for it people`
`04:40.13`	`Qwell`	`file[laptop]: $938?`
`04:40.16`	`Qwell`	`CAD`
`04:40.18`	`file[laptop]`	`invoice amount$1,603.26`
`04:40.19`	`ObsidianX`	`how would i go about fixing the error "Inappropriate authentication received" when i try to connect an IAX client to *`
`04:40.21`	`SwK`	`yeah what qwell said`
`04:40.21`	`Qwell`	`jesus`
`04:40.35`	`rob0`	`file[laptop]: have it committed :)`
`04:40.35`	`Qwell`	`file[laptop]: how the hell did you manage that?`
`04:40.39`	`SwK`	`it always gribes about codec_729 cause you dont have the source for it`
`04:40.40`	`*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`04:40.44`	`file[laptop]`	`we'll see.`
`04:40.45`	`rob0`	`I know well how he did it!!`
`04:40.48`	`ManxPowe`	`Just how DOES one get a $1,000 cell phone bill anyway?`
`04:40.51`	`ManxPowe`	`file, GO PREPAY!`
`04:41.02`	`ManxPowe`	`rob0, all those phonesex phone calls?`
`04:41.10`	`SwK`	`is that CDN File?`
`04:41.12`	`file[laptop]`	`I just have the transaction on my account, I don't have the invoice online yet and my balance isn't adjusted yet`
`04:41.13`	`rob0`	`I saw him here, typing in IRC, while on the road`
`04:41.15`	`fugitivo`	`WTF??`
`04:41.17`	`file[laptop]`	`SwK: yes`
`04:41.19`	`file[laptop]`	`rob0: yup`
`04:41.30`	`file[laptop]`	`they probably billed me for data, and backbilled me for past data usage`
`04:41.31`	`fugitivo`	`file[laptop]: $1600?????`
`04:41.33`	`SwK`	`file: oh so its like a normal 100USD phone bill?`
`04:41.41`	`ManxPowe`	`Ah. Mine would be like $5,000 if I wasn't on the flat rate data plan`
`04:41.47`	`file[laptop]`	`I need to calculate how it got to that amount though`
`04:41.48`	`file[laptop]`	`it makes no sense`
`04:41.57`	`Qwell`	`$50/kb?`
`04:41.58`	`Jameno123`	`SwK, yea, it bitched about more, but im not pasting them all :) should i rm -rf the modules dir, and reinstall it all completely?`
`04:42.07`	`Jameno123`	`like 15 files are listed`
`04:42.08`	`Jameno123`	`heh`
`04:42.13`	`h3x`	`damn bid snipers`
`04:42.21`	`xachen`	`Canada data rates are bad for mobile providers`
`04:42.23`	`h3x`	`i accidently pasted a auction item number in where a price goes`
`04:42.26`	`xachen`	`they will coin you easily $1/mbv`
`04:42.29`	`h3x`	`and i bid 5 million on an ATA device`
`04:42.40`	`file[laptop]`	`my regular bill is $60`
`04:42.40`	`SwK`	`jamesno123: probably want to get rid of them but not the g729 one`
`04:42.48`	`SwK`	`you'll need it for g729`
`04:42.55`	`Jameno123`	`SwK, yea, i use g729, i know about it ;)`
`04:43.03`	`file[laptop]`	`so I used 100MB of data apparently`
`04:43.09`	`Jameno123`	`like you said, only because it wasnt compiled directly be the source`
`04:43.16`	`fugitivo`	`file[laptop]: don't pay it, that's insane`
`04:43.22`	`xachen`	`downloading porn onto your blackberry? :D`
`04:43.24`	`file[laptop]`	`fugitivo: I'm waiting for the bill.`
`04:43.26`	`xachen`	`:O rather`
`04:43.26`	`\|omni\|`	`Cingular did that to me a couple months ago but it was only $580 for data`
`04:43.27`	`*** join/#asterisk sumonish (n=God@203.12.249.168)`
`04:43.32`	`sumonish`	`hi all`
`04:43.45`	`\|omni\|`	`I switched to the unlimited data account... a mere $20 more than I was paying already`
`04:43.46`	`\|omni\|`	`bastages`
`04:43.59`	`*** join/#asterisk srt (n=nobody@18.120.9.213.dsl.getacom.de)`
`04:44.16`	`file[laptop]`	`I'm not overly thrilled, but I legitimately used it so if they billed it right... yeah`
`04:44.31`	`file[laptop]`	`life goes on`
`04:44.53`	`file[laptop]`	`so help me god if my mother opens my cellphone bill`
`04:45.04`	`SwK`	`hahaha`
`04:45.10`	`fugitivo`	`heart attack`
`04:45.22`	`sumonish`	`i have an asterisk server which my boss has setup and left me with unfortunatly the CallerID is causeing an issue where when a call comes in it dumps the call i have the following issue in Myphp The $cfg['PmaAbsoluteUri'] directive MUST be set in your configuration file! can someone tell my what it means and how to fix it??`
`04:45.29`	`file[laptop]`	`she's been nosey lately, she opened my credit card statement ahead of me while I was right in front of her`
`04:45.38`	`file[laptop]`	`and my rrsp notice`
`04:45.44`	`SwK`	`rrsp?`
`04:45.51`	`file[laptop]`	`it's like, "uh... I'm 19 here... get out of my finances"`
`04:45.51`	`rob0`	`yikes!`
`04:45.53`	`\|omni\|`	`sumonish: , that's not your issue, that's just a setting in phpMyAdmin`
`04:45.58`	`file[laptop]`	`SwK: registered retirement savings plan`
`04:46.02`	`SwK`	`oh`
`04:46.06`	`\|omni\|`	`you can edit config.inc`
`04:46.12`	`SwK`	`i guess thats cdn for 401k`
`04:46.13`	`rob0`	`file[laptop]: get a PO Box`
`04:46.27`	`\|omni\|`	`and set the full URL to phpMyAdmin (i.e. http://path.to.server/phpMyAdmin) and that message will go away`
`04:46.39`	`file[laptop]`	`rob0: mmm I could`
`04:46.41`	`rob0`	`in USPS they're pretty cheap`
`04:46.41`	`sumonish`	`i edited the zapata.conf`
`04:46.51`	`sumonish`	`to turn of caller id is that right?`
`04:46.55`	`sumonish`	`i seems to work`
`04:46.55`	`rob0`	`I pay $18/year I think`
`04:47.05`	`file[laptop]`	`I believe it's $60 CAD/year here`
`04:47.24`	`sumonish`	`ok omni`
`04:47.36`	`rob0`	`they cost more in cities, mine is in a tiny town`
`04:47.36`	`SwK`	`damn did apple release enuff patches yesterday?`
`04:47.49`	`MikeJ__`	`file, so that's like one regular cell bill a year?`
`04:47.50`	`rob0`	`but Canada is no doubt different`
`04:48.04`	`file[laptop]`	`MikeJ__: more`
`04:48.12`	`file[laptop]`	`my regular cell bill is $60/mth total`
`04:48.50`	`sumonish`	`omni where is config.inc stored?`
`04:49.12`	`Jameno123`	`hrm, alright, seems to work :)`
`04:49.22`	`Jameno123`	`but didnt solve my problem/reason for upgrading`
`04:49.23`	`Jameno123`	`heh`
`04:49.25`	`Jameno123`	`1st File Descriptor: -1`
`04:49.29`	`Jameno123`	`<PROTECTED>`
`04:50.49`	`Jameno123`	`after it bridge's a call, it hangs and gives nothing.`
`04:51.03`	`Jameno123`	`service provider returning no data? or some other weird crapola?`
`04:51.30`	`twisted[asteria]`	`SwK, you sure you don't have that shit?`
`04:51.46`	`bsdfreak`	`heh`
`04:52.04`	`qhrisnd`	`I need help with 2 things: 1) I need to find out how to create in my dial plan, a way to make an extension ring over to another extension when its busy. 2) I would like to know how to (if possible) route calls based upon caller id. Can anyone give me some tips?`
`04:52.27`	`Qwell`	`twisted[asteria]: y0`
`04:52.44`	`Qwell`	`twisted[asteria]: going to ETel?`
`04:53.01`	`twisted[asteria]`	`Qwell, no`
`04:53.03`	`ManxPowe`	`qhrisnd, See "show application dial" and the [macro-stdexten] section of extensions.conf. Also see the Wiki and the Asterisk book.`
`04:53.06`	`ManxPowe`	`~docs`
`04:53.08`	`jbot`	`i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html`
`04:53.19`	`Qwell`	`twisted[asteria]: shame..`
`04:53.26`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`04:53.37`	`fugitivo`	`qhrisnd: and DIALSTATUS`
`04:53.47`	`qhrisnd`	`thank you`
`04:53.49`	`twisted[asteria]`	`Qwell, well, if I had known about it sooner, i might could have`
`04:55.51`	`SwK`	`twisted I am sure`
`04:56.57`	`*** join/#asterisk aless (n=fruribe@pc-100-230-83-200.cm.vtr.net)`
`04:56.58`	`*** join/#asterisk pdugas (n=pdugas@h73.90.40.69.ip.alltel.net)`
`04:57.11`	`aless`	`hi, which ports do i need to forward when using a nat?`
`04:57.18`	`Qwell`	`aless: which channel types?`
`04:58.20`	`inv_Arp`	`aless: any port you want`
`04:58.40`	`aless`	`im connecting two servers with iax`
`04:59.00`	`Qwell`	`aless: 4569`
`04:59.06`	`*** part/#asterisk loud (n=ariel@cypher.punk.net)`
`04:59.27`	`*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)`
`05:00.04`	`aless`	`only that one? arent any other services sending packets?`
`05:00.18`	`ObsidianX`	`netstat -nap`
`05:00.20`	`ObsidianX`	`if you wanna find out`
`05:00.31`	`bsdfreak`	`aaa`
`05:00.38`	`ObsidianX`	`course you'll have to look for asterisk processes :P`
`05:02.36`	`SwK`	`damn it`
`05:03.53`	`Jameno123`	`blah blah blah! damn thing :( argh, why the heck doesnt this thing WORK!!!!!!!!!!!! :(`
`05:04.02`	`Jameno123`	`how can i determine where my problem is :(`
`05:04.31`	`mogorman`	`? Jameno123`
`05:04.32`	`Jameno123`	`i call from my cisco 7960 via sip to asterisk1, asterisk1 dials asterisk2, asterisk2 dials our provider.`
`05:04.35`	`mogorman`	`calm down....`
`05:04.52`	`Jameno123`	`asterisk1->asterisk2 = iax`
`05:04.56`	`Jameno123`	`asterisk2->provider = iax`
`05:05.01`	`mogorman`	`k`
`05:05.08`	`Jameno123`	`if i do "iax2 show channels" on asterisk2, it shows a "UP" bridged channel`
`05:05.22`	`Jameno123`	`yet, i see hear nothing`
`05:05.36`	`mogorman`	`i see hear?`
`05:05.41`	`Jameno123`	`see/hear*`
`05:05.49`	`Jameno123`	`i see no errors, and hear nothing on the phone`
`05:05.56`	`Jameno123`	`if i hang up the phone`
`05:05.57`	`mogorman`	`is jitterbuffer on?`
`05:06.39`	`Jameno123`	`asterisk1 disconnects the call, but asterisk2 still thinks the call is in progress, and doesnt disconnect until it times out.`
`05:06.46`	`*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`05:06.49`	`Jameno123`	`mogorman, which server? all of them?`
`05:07.00`	`mogorman`	`any of them?`
`05:07.07`	`Jameno123`	`this shit just started happening 3 days ago`
`05:07.11`	`Jameno123`	`its been fine "forever" :(`
`05:07.24`	`Jameno123`	`asterisk2=jitterbuffer=no`
`05:07.43`	`Jameno123`	`asterisk1=jitterbuffer=no`
`05:07.48`	`Jameno123`	`i dont know about my service provider`
`05:08.07`	`mogorman`	`hmm it sounds like a bug we have been working on`
`05:08.16`	`Jameno123`	`bug? heh`
`05:08.23`	`Jameno123`	`it just "mysteriously" happens?`
`05:08.46`	`mogorman`	`does this happen if you turn off iax native transfer`
`05:08.47`	`Jameno123`	`heh, just magically started happening one day`
`05:09.09`	`watchy`	`whats the quick reset for a 7960?`
`05:09.40`	`mogorman`	`pull the plug ^_^`
`05:09.42`	`Jameno123`	`watchy, reboot? (*+6+services)`
`05:09.50`	`watchy`	`thanks brother`
`05:09.52`	`Jameno123`	`err settings`
`05:09.59`	`Jameno123`	`* 6 settings`
`05:10.02`	`Jameno123`	`1 of the two`
`05:10.12`	`Qwell`	`real men #`
`05:10.24`	`Jameno123`	`mogorman: hrm.`
`05:10.31`	`Qwell`	`<rant>`
`05:10.31`	`Jameno123`	`i cant say ive ever done that before ;)`
`05:10.42`	`Qwell`	`Why did Cisco do # for the reboot on the sccp 7960?`
`05:10.47`	`Jameno123`	`let me go read some docs, or shed me some light :)`
`05:10.52`	`Qwell`	`You have to be in settings for it to work...`
`05:11.05`	`Jameno123`	`sccp, blah!`
`05:11.07`	`Qwell`	`and...what do you need to press to unlock the phone? That's right... **#`
`05:11.09`	`mogorman`	`id try turning off native transfer first`
`05:11.21`	`Jameno123`	`mogorman, thats what im reading docs to figure out how ;)`
`05:11.29`	`Qwell`	`So, if you want to unlock, and it didn't appear to work the first time...what do you do? You press it again`
`05:11.41`	`Qwell`	`and in doing so...you reboot the damn thing. How stupid...`
`05:11.42`	`Jameno123`	`notransfer=no ?`
`05:11.42`	`Qwell`	`</rant>`
`05:12.04`	`mogorman`	`hmm i think so....`
`05:12.09`	`mogorman`	`id have to look it up sorry`
`05:13.01`	`*** join/#asterisk aless_ (n=fruribe@pc-100-230-83-200.cm.vtr.net)`
`05:13.21`	`Jameno123`	`mogorman, nope, didnt help`
`05:13.42`	`Jameno123`	`i think junction networks is being a pain in my arse again :)`
`05:13.48`	`mogorman`	`did you turn it on all points and check it again`
`05:14.06`	`Jameno123`	`i turned it "off"`
`05:14.10`	`Jameno123`	`it should be "on" ?`
`05:14.22`	`Jameno123`	`i disabled it, on all servers, yet`
`05:14.23`	`Jameno123`	`yes*`
`05:14.38`	`Jameno123`	`err both* well, the two i have access too, not my providers, of course.`
`05:14.51`	`Jameno123`	`i think its just a provider issue :(`
`05:15.03`	`mogorman`	`maybe`
`05:15.05`	`Jameno123`	`ive never had any problems, and if i dial other phones on my asterisk server, i dont have problems.`
`05:15.08`	`Jameno123`	`so if i do`
`05:15.15`	`Jameno123`	`phone1->ast1->ast2->phone2`
`05:15.18`	`Jameno123`	`no problems, ever`
`05:15.29`	`Jameno123`	`phone1->ast1->ast2->provider=problems`
`05:15.35`	`mogorman`	`yeah probably`
`05:24.49`	`Jameno123`	`mogorman, ;) so stressful when you cant figure out why something is happening hehe`
`05:30.18`	`*** join/#asterisk ThaZZa_Work (n=me@203.80.44.200)`
`05:32.21`	`Jameno123`	`whelp thanks guys for your help, ill go chew on the ear of my service provider tomorrow.`
`05:32.25`	`Jameno123`	`cya.`
`05:32.26`	`litage`	`if you have 1,000+ tenants, each with varying #s of extensions, and use SER to handle registrations, is it feasible to put all tenants on each * box?in a phone call [through *], if 1 person is talking and the other person is silent, are they both sending the same amount of voice data?`
`05:33.24`	`mogorman`	`yeah i understand Jameno123`
`05:33.50`	`litage`	`in a phone call [through *], if 1 person is talking and the other person is silent, are they both sending the same amount of voice data?`
`05:34.58`	`Qwell`	`litage: Since * doesn't do VAD, yes`
`05:35.07`	`litage`	`VAD?`
`05:35.13`	`Qwell`	`~vad`
`05:35.14`	`jbot`	`i heard vad is Voice Activity Detection`
`05:35.19`	`litage`	`ah`
`05:36.21`	`litage`	`Qwell: so the type (volume, pitch, etc) of audio/voice doesn't affect the amount of data transferred?`
`05:36.48`	`Qwell`	`afaik, no`
`05:37.18`	`litage`	`interesting`
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`05:38.35`	`lo_tech`	`not AM, bro... louder doesnt mean bigger data :)`
`05:39.52`	`Jameno123`	`oh before i go`
`05:39.54`	`Jameno123`	`one more thing :)`
`05:40.26`	`Jameno123`	`WHen a user transfers a call, on a cisco ip phone (SIP), to another extension, why does the phone never receive anymore calls?`
`05:40.30`	`Jameno123`	`asterisk thinks its "busy"`
`05:40.31`	`litage`	`lo_tech: "not AM"?`
`05:40.42`	`Jameno123`	`litage, "its not AM (like radio)`
`05:41.02`	`lo_tech`	`litage: amplitude modulation...`
`05:41.06`	`litage`	`ah`
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`05:41.42`	`Jameno123`	`anyone have any idea what i possible am doing wrong?`
`05:41.42`	`litage`	`so if 2 people are using sip and g729, will each person's incoming and outgoing data and voice streams be fairly constant?`
`05:42.18`	`Jameno123`	`they used to transfer, and then receive more inbound calls`
`05:42.23`	`Jameno123`	`now the phones are staying busy`
`05:42.38`	`Jameno123`	`probably something todo with "tT" ? or canreinvite or something?`
`05:42.44`	`Mavantix`	`is there anyway to have asterisk IM me incoming call info, log messages, etc?`
`05:43.01`	`lo_tech`	`all things being equal, without silence suppression or vad, yes... the bandwidth used will be equal for both parties, regardless of how loud or the amount of silence for each phone`
`05:43.03`	`ManxPowe`	`Jameno123, sounds like you are using imcominglimit=1 or setgroup, etc`
`05:43.44`	`ManxPowe`	`Jameno123, if so, this is a know issue, see the mailing list archives, there may be a fix or something.`
`05:44.01`	`Jameno123`	`ManxPowe: hrm, they do disable callwaiting, if callwaiting is enabled it rings fine.`
`05:44.16`	`Jameno123`	`as i thought, if you transfer your phone is released from the call?`
`05:44.23`	`Jameno123`	`i didnt think the phone 'bridged' the call.`
`05:44.56`	`Jameno123`	`ManxPowe, incominglimit is undefined in my sip.conf`
`05:45.19`	`Jameno123`	`and setgroup would be a no.`
`05:45.42`	`Jameno123`	`though, i dont specify "canreinvite"`
`05:45.46`	`Jameno123`	`in the sip.conf, so thats probably the issue?`
`05:46.44`	`watchy`	`anyway to set cisco volume in sipdefault?`
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`06:13.14`	`littleball`	`hello, i use E1/PRI, asterisk1.2.1. I got the following warning in the console:`
`06:13.15`	`littleball`	`Jan 12 14:00:18 NOTICE[6681]: channel.c:2416 __ast_request_and_dial: Don't know what to do with control frame 15`
`06:13.21`	`littleball`	`what does it mean?`
`06:27.52`	`wunderkin`	`heh holy crap, the * messages log on my one server never has been rotated`
`06:29.10`	`lo_tech`	`not so bad unless you`
`06:29.16`	`lo_tech`	`are verbose, debug`
`06:30.52`	`wunderkin`	`22k lines since sept`
`06:31.16`	`wunderkin`	`verbose is set to 20 but i dont do much with it, just testing`
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`06:47.02`	`chat_jokey`	`hello people`
`06:48.10`	`chat_jokey`	`i am currently doing some asterisk sizing .. task is to support 150 incoming TDM lines and 175 outgoing lines .. with approximately 4000 extensions (mostly used only for intercom)`
`06:48.24`	`chat_jokey`	`anyone can suggest me any pointers on the dimensioning of the same ?`
`06:48.43`	`chat_jokey`	`i read up with voip-info.org .. but its kinda not clear ..`
`06:49.03`	`chat_jokey`	`I am averaging about 400 - 500 odd extensions running from one asterisk box ..`
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`06:56.10`	`welles`	`hi all`
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`07:14.49`	`welles`	[help] i try to install mpg123 on centos4 and it hints that :'decode_i586.s:44: Error: suffix or operands invalid for `push' ...' what's wrong? my machine is 64bit machine
`07:25.25`	`litage`	`is H323 or SIP more NAT- and network-friendly?`
`07:25.58`	`*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)`
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`07:30.01`	`Lee619`	`hello`
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`07:34.49`	`tzafrir_laptop`	`welles, use rawplayer, unless you want to stream music`
`07:36.58`	`welles`	`rawplayer? ok,i have a try .it can replace mpg123 for music on hold on *?`
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`07:52.00`	`koperniqs`	`hi`
`07:52.12`	`Lee619`	`good morning`
`07:56.27`	`infinity1`	`good nite`
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`08:14.16`	`ObsidianX`	`litage: i read that IAX was NAT friendly`
`08:14.28`	`ObsidianX`	`litage: i think it uses UDP`
`08:15.23`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`08:18.27`	`chat_jokey`	`any one can give pointers on clustering asterisk ?`
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`08:23.21`	`bazz`	`i'm trying to get asterick going, i've set up my extentions.conf file (i thought) but when i copy a .call file into the outgoing spool i get __ast_request_and_dial: Don't know what to do with control frame 15 and then attempt_thread: Call failed to go through, reason 3. any ideas?`
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`08:24.09`	`wellng`	`hi all`
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`08:27.36`	`koperniqs`	`chat_jokey: what kind of clustering?`
`08:30.01`	`chat_jokey`	`like i want to have like 4000 extensions - something like IP Centrex`
`08:30.17`	`*** join/#asterisk Rawplayer (i=kevin@ipc31055d2.oom-killer.org)`
`08:30.41`	`chat_jokey`	`koperniqs: trying to figure out how many extension a Dual XEON - 3.0Ghz, 4GRAM can handle ..`
`08:31.00`	`chat_jokey`	`based on that wanna do some sizing ..`
`08:32.37`	`koperniqs`	`chat_jokey: ther's a tool called sipsak (sipsak.org) that might help`
`08:39.26`	`chat_jokey`	`koperniqs: lemme have a look`
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`08:43.31`	`*** join/#asterisk DHuang (n=DHuang@mail.medec.com.au)`
`08:43.44`	`DHuang`	`Hi`
`08:44.24`	`DHuang`	`Can someone help me with SER + Asterisk?`
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`08:51.37`	`*** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it)`
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`08:53.00`	`DHuang`	`helo?`
`08:56.42`	`Nico_Bdav`	`hi all`
`08:57.09`	`DHuang`	`hi Nico... can you help me with SER + Asterisk?`
`08:57.10`	`chat_jokey`	`hi DHuang even i am looking for similar stuff`
`08:57.21`	`Nico_Bdav`	`does anyone know a good T1->IP gateway, compatible with asterisk ?`
`08:57.39`	`Nico_Bdav`	`DHuang, no sorry`
`08:57.42`	`chat_jokey`	`Nico_Bdav: are you looking for TDM hardware ?`
`08:58.00`	`DHuang`	`chat_jokey: I see... what I'm trying is to make SIP Client to call each other through SER + Asterisk`
`08:58.03`	`chat_jokey`	`Asterisk itself can act as gateway !`
`08:58.09`	`Nico_Bdav`	`chat_jokey, i want to test asterisk on one site`
`08:58.43`	`Nico_Bdav`	`but i want on another site which already have a PBX to convert T1 outlet to IP`
`08:59.16`	`DHuang`	`Chat: kewl.. just tried a config and work now.. :-p`
`08:59.40`	`chat_jokey`	`DHuang: i am trying to scale asterisk, so its suggested that one uses SER as SIP Proxy and enable it to throw SIP calls into Multiple Asterisk boxes, but i dont seem to find anything relevant online ... can anyone else help me on this ?`
`09:00.18`	`DHuang`	`Chat: search fallover I think is on the original setup doc.`
`09:01.23`	`chat_jokey`	`I have A@H here .. hmm`
`09:02.13`	`DHuang`	`Dam... not working.. ;-(`
`09:03.54`	`*** join/#asterisk tengulre11 (n=tengulre@61.185.224.66)`
`09:20.54`	`*** join/#asterisk A-jay (n=quirc@62.217.245.194)`
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`09:24.35`	`iDunno`	`morning`
`09:24.58`	`A-jay`	`hi`
`09:25.00`	`DHuang`	`Chat: does your Asterisk do the registering or the SER?`
`09:25.06`	`DHuang`	`Morning there.`
`09:25.13`	`A-jay`	`hi`
`09:25.54`	`DHuang`	`I'm trying to figure out how to SER and register on Asterisk so it shows the right HOST IP for the client?`
`09:28.54`	`*** join/#asterisk darkskiez (n=darkskie@bb-194-6-115-241.ukonline.co.uk)`
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`09:30.41`	`Curus`	`Is it possible to dump all session variables from extensions.conf?`
`09:31.40`	`Curus`	`I tried with an AGI script, but I can only get one variable at a time, and only if I know the name`
`09:32.52`	`*** join/#asterisk KriS83 (n=KriS@212.202.141.92)`
`09:33.31`	`RoyK`	`er`
`09:33.43`	`RoyK`	`User disconnected from queue %s while waiting their turn`
`09:33.45`	`RoyK`	`wtf????`
`09:33.53`	`RoyK`	`and noone are put into that queue`
`09:35.40`	`*** part/#asterisk DHuang (n=DHuang@mail.medec.com.au)`
`09:42.17`	`RoyK`	`argh. just upgraded to 1.2.x from 1.0 and now support centres are losing calls. after a while phones stop ringing. people still queueing up..`
`09:46.09`	`*** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au)`
`09:46.24`	`thazza`	`Hey all`
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`09:48.05`	`Curus`	`There is no way to display all currently set variables in extensions.conf?`
`09:48.18`	`RoyK`	`seems like there's a fsckup somewhere in device state`
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`09:48.44`	`RoyK`	`Curus: iirc it's quite easy to go through all _channel_ vars with an agi script`
`09:55.03`	`Curus`	`How?`
`09:56.14`	`JonR800`	`any way to pass hints between two asterisk servers? I suppose that's a job for SER.`
`09:56.20`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`09:58.38`	`Curus`	`Channel variables don't all get passed to AGI`
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`10:03.34`	`RoyK`	`zoa: ping`
`10:06.18`	`zoa`	`pong`
`10:11.20`	`thazza`	`pang`
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`10:17.38`	`riksta`	`if i have a sangoma A101, when i install do i need the PRI or BRI use flags?`
`10:19.01`	`cypromis`	`PRI`
`10:19.44`	`riksta`	`ok ta`
`10:19.54`	`riksta`	`for euroisdn?`
`10:21.02`	`af_`	`how good is snom 320?`
`10:22.17`	`RoyK`	`http://blog.outer-court.com/prejudice/`
`10:22.41`	`*** join/#asterisk A-jay (n=quirc@62.217.245.194)`
`10:22.58`	`Ahrimanes`	`hey denmark is not mentioned, damnit`
`10:24.19`	`koperniqs`	`af_: the display is small and it's relativly expensive`
`10:26.43`	`*** join/#asterisk gvag11 (n=g@ipa124.6.tellas.gr)`
`10:26.48`	`gvag11`	`hi all`
`10:27.13`	`RoyK`	`koperniqs: relativily, yes, unless you mention norway in that sentence`
`10:27.17`	`RoyK`	`er`
`10:27.25`	`gvag11`	`i just moved to Asterisk 1.2.1 and i miss the CUT function, does somebody knows something ?`
`10:27.26`	`RoyK`	`that was a bummer`
`10:27.40`	`RoyK`	`gvag11: read about asterisk variables`
`10:27.52`	`RoyK`	`http://www.voip-info.org/wiki-Asterisk+variables`
`10:28.03`	`RoyK`	`<PROTECTED>`
`10:28.40`	`zoa`	`royk: http://www.asteriskguru.com/tutorials/cut_function.html`
`10:28.47`	`zoa`	`ows, gvag11`
`10:29.23`	`*** join/#asterisk fulgas (n=fulgas@209.8.233.107)`
`10:29.28`	`zoa`	`you need to use SET for it now`
`10:29.43`	`RoyK`	`http://bugs.digium.com/view.php?id=6218`
`10:29.45`	`RoyK`	`:(`
`10:30.55`	`gvag11`	`zoa : ok ... so i use the SET(var=${CUT ... thanks a lot zoa ...`
`10:31.14`	`gvag11`	`royk : thanks ...`
`10:34.21`	`af_`	`mhh what phone is good to use with *?`
`10:34.28`	`af_`	`I used gs but not very satisfied`
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`10:43.08`	`iDunno`	`FFS`
`10:43.12`	`iDunno`	`is it just me...`
`10:43.31`	`iDunno`	`or does it seem entirely insane that you end up in a queuing system when phoning a Telco`
`10:43.39`	`iDunno`	`these people need more staff, ffs.`
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`10:52.13`	`micolous_`	`hey, i'm having some issues using meetme. when i have a caller using the ilbc codec over a iax2 trunk, the sound from them is very jittery, yet they can hear me and other non-ilbc users fine... capturing the output from them, i see that there sound is breaking up... for about 0.02 seconds the sound is fine, then for 0.01 seconds there's no sound... and this goes on and on`
`10:52.33`	`micolous_`	`i'm using the ztdummy kernel module as my timing source`
`10:53.23`	`micolous_`	`i'm wondering if this is something wrong on my end, or a bug. i've tweaked around with the jitterbuffer and that doesn't seem to help; and without the jitterbuffer it's even worse. and it's asterisk 1.2.1`
`10:56.13`	`*** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net)`
`10:57.46`	`h3x`	`its probably because the frame size is different on your codecs`
`10:58.08`	`micolous_`	`yeah, i noticed it doesn't effect ulaw at all`
`10:58.40`	`micolous_`	`but my friend using asterisk@home with meetme doesn't have this issue, and he's using the same codecs and upstream iax providers`
`10:58.56`	`h3x`	`what is he using for zaptel timing`
`10:59.04`	`micolous_`	`the dummy driver`
`10:59.20`	`h3x`	`a@h is prob a different version of asterisk right`
`10:59.29`	`micolous_`	`yeah, i think it might be 1.0`
`11:00.00`	`h3x`	`i seem to remember somebody else having a problem like this with 1.2`
`11:03.56`	`*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@host.190.115.68.195.rev.coltfrance.com)`
`11:04.58`	`tzafrir_laptop`	`asterisk@home is basically a sort of asterisk distribution`
`11:05.08`	`*** join/#asterisk gvag11 (n=g@ipa124.6.tellas.gr)`
`11:05.12`	`gvag11`	`hi again ...`
`11:05.33`	`*** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com)`
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`11:06.28`	`gvag11`	`i changed the extension.conf and instead of CUT(VAR=... i put SET(Var=${CUT(... but still asterisk (1.2.1 stable) complains the CUT is not register ... Any ideas ?`
`11:09.17`	`micolous_`	`tzafrir_laptop: yeah, i remember helping him set it up in september, so it would be running on asterisk 1.0`
`11:15.49`	`gvag11`	`i changed the extension.conf and instead of CUT(VAR=... i put SET(Var=${CUT(... but still asterisk (1.2.1 stable) complains the CUT is not register ... Any ideas ?`
`11:23.11`	`*** join/#asterisk hhoffman_ (n=hhoffman@tor/session/x-059aa9758068410e)`
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`11:27.08`	`*** join/#asterisk Gervystar (n=gervysta@62.94.208.119)`
`11:27.20`	`gvag11`	`i changed the extension.conf and instead of CUT(VAR=... i put SET(Var=${CUT(... but still asterisk (1.2.1 stable) complains the CUT is not register ... Any ideas ?`
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`11:28.07`	`*** join/#asterisk RoyK (n=roy@213.160.242.42)`
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`11:37.57`	`Reverend`	`OMFG`
`00:00.16`	`RoyK`	`ManxPower: yes`
`00:00.20`	`BasketCase`	`Ariel_: I haven't touched the POTS port yet`
`00:00.29`	`Lee619`	`does * require registration for outoing calls or just incoming calls?`
`00:00.41`	`BasketCase`	`Ariel_: I meant to say the FXO port is not configured yet`
`00:00.43`	`Ariel_`	`Lee619, depends on service provider`
`00:00.44`	`Powerkill`	`someone use cdr_odbc with mysql ?`
`00:01.07`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`00:01.17`	`ManxPower`	`"Just say 'NO!' to POTS." This message brought to you by the Partnership for an Analog Free Amerika.`
`00:01.17`	`Darwin35`	`ps2pdf is part of what port`
`00:01.20`	`Lee619`	`Ariel: Thank you. Do you happen to know about FWD?`
`00:01.44`	`Ariel_`	`fwd does need registration`
`00:02.12`	`Ariel_`	`ManxPower, pots are needed in some cases, at least to get me out of hot water with 911 issues`
`00:04.13`	`blitzrage`	`ManxPower: lol -- thats my new MSN name :)`
`00:08.28`	`*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`00:09.54`	`*** part/#asterisk quadrata (n=quadrata@ool-182c2aaf.dyn.optonline.net)`
`00:15.13`	`*** part/#asterisk darkskiez (n=darkskie@bb-194-6-115-241.ukonline.co.uk)`
`00:16.30`	`tzanger`	`heh`
`00:16.35`	`tzanger`	`I'm watching the canadian political debates`
`00:16.40`	`tzanger`	`cbc.ca has the .rm`
`00:16.42`	`rue_work`	`why?`
`00:16.48`	`tzanger`	`rue_work: Well I am canadian`
`00:16.50`	`rue_work`	`there just mud slining`
`00:16.56`	`rue_work`	`I know, me too`
`00:16.57`	`Soul`	`greetinz`
`00:17.03`	`Soul`	`dirty question:`
`00:17.31`	`tzanger`	`layton sounds like he is selling insurance, the bloc shouldn't be in this debate whatsoever, and martin and harper just are different sides of the same coin. ugh.`
`00:17.36`	`Soul`	`picture a company with 2 geographical locations, one asterisk server in each location`
`00:17.44`	`tzanger`	`Soul: yeah`
`00:17.45`	`rue_work`	`I dispise polititions, especially when their throwing mud at each other trying to make it an election of who looks less worse`
`00:17.53`	`tzanger`	`rue_work: yep`
`00:18.14`	`*** join/#asterisk coppice (n=chatzill@166.168.17.210.dyn.pacific.net.hk)`
`00:18.21`	`Soul`	`how can a user from location A go work to location B, and still be reachable by the same sip url / company extension ?`
`00:18.27`	`tzanger`	`basically the PC is shouting "We're not the Liberals!" the Libs are saying "Trust us this time, really" and the NDP is saying "Lookat me, Look at me!"`
`00:18.27`	`rue_work`	`Soul ours has three locations`
`00:18.50`	`tzanger`	`Soul: yesish. :-)`
`00:18.55`	`rue_work`	`hehe yea...`
`00:18.56`	`ManxPower`	`Soul, move the phone.`
`00:19.11`	`Soul`	`i'd like the user to go from A to B, and just reprogram one of the ip phones with his login and password, and thats it. is this possible ?`
`00:19.31`	`tzanger`	`Soul: yes`
`00:19.35`	`tzanger`	`that is entirely possible`
`00:19.42`	`ManxPower`	`Soul, Why? Just move the phone, let it register with the erver in the other location`
`00:19.58`	`[TK]D-Fender`	`Soul : plenty of ways. have phone phones active at the same time, just have it so there's only 1 number that rings BOTH in your dial-plan.`
`00:19.59`	`Soul`	`but location B has a different asterisk server! how does this work ? are the extensions/dialplan/sip profiles shared between the 2 asterisk servers ?`
`00:20.03`	`*** join/#asterisk jyukes_ (n=jameshot@pool-138-89-211-251.atc.east.verizon.net)`
`00:20.03`	`rue_work`	`ok, who here is running an asterisk machine with voicemail and IVR?`
`00:20.06`	`tzanger`	`ManxPower: I say fuck all that, log in as an agent.`
`00:20.14`	`tzanger`	`we likely all are`
`00:20.34`	`[TK]D-Fender`	`rue_work : Most of us, myself included. Whats your question?`
`00:20.41`	`rue_work`	`well, then you all have this problem`
`00:20.55`	`rue_work`	`WARNING[16724] file.c: File outage does not exist in any format`
`00:21.05`	`rue_work`	`check /var/log/asterisk/full`
`00:21.06`	`ManxPower`	`Soul, Um, the phone doesn't register with the local server, the phone registers and users the REMOTE server`
`00:21.08`	`[TK]D-Fender`	`rue_work : Thats just 1 sound file.....`
`00:21.11`	`*** join/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`00:21.18`	`*** part/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`00:21.24`	`[TK]D-Fender`	`Who said it had to be there in the first place?`
`00:21.26`	`rue_work`	`right, I want to know if this is a normal problem`
`00:21.30`	`watchy`	`anyway to set a cisco 7960s volume from tftp config?`
`00:21.31`	`Soul`	`[TK]D-Fender, though of that, in fact i have 3 sip logins (sergio-pocketpc, sergio-cisco and sergio-notebook, which all ring when someone calls "sergio"), but with 2 asterisk servers wont there be dialing problems ?`
`00:21.41`	`rue_work`	`cause that sound file isn't provided with asterisk`
`00:22.04`	`BeHappy_`	`Soul, i think you can set-up a queue with the "ringall" policy`
`00:22.15`	`tzanger`	`haaaaaaaaaaaaaaaaahahahahahhahaha`
`00:22.15`	`[TK]D-Fender`	`Soul : depends how you set it up. Have the remote side take the call and ring the internal phone but WITHOUT doing an "answer" first`
`00:22.18`	`tzanger`	`Saying Java is nice because it works on all OS's is like saying that anal sex is nice because it works on all genders.`
`00:22.34`	`BeHappy_`	`but sincerely i dont know if the queue can go trough different servers`
`00:22.36`	`Soul`	`ManxPower, hadn't thought of that, registering with the remote server, nice. but the phone connectivity will be lost if outside comms fail, isnt there a way to login in the local server ?`
`00:22.39`	`[TK]D-Fender`	`Queue's for that idea = BAD and wasteful.`
`00:22.56`	`BeHappy_`	`ockay, as not said :)`
`00:23.01`	`ManxPower`	`Soul, yes, but that's more complicated`
`00:23.14`	`rue_work`	`so am I right about 'outage.gsm" ?`
`00:23.17`	`Soul`	`watchy, yes, but sorry, don't have my cisco configs here`
`00:23.39`	`watchy`	`soul`
`00:23.45`	`watchy`	`thanks i'll see what i can find`
`00:23.53`	`watchy`	`i need a website with all the options`
`00:24.00`	`[TK]D-Fender`	`Soul : have the remote phone log into the server its BEHIND. Place the call from server A to server B requesting an entry taht will dial the phone behind it. thats all.`
`00:24.17`	`tzanger`	`holy hell are you STILL talking about outage.gsm?`
`00:24.19`	`rue_work`	`grrr I have to ctrl-c windows every time I do a copy!!!! >:\|`
`00:24.22`	`Soul`	`watchy, google 4 it, and come back tomorrow if you find nothing, i'll share my configs`
`00:24.24`	`tzanger`	`find / -name 'outage.gsm'`
`00:24.27`	`tzanger`	`see where it is`
`00:24.31`	`rue_work`	`tzanger no, I'm talking about it again`
`00:24.35`	`watchy`	`soul: thank you`
`00:24.51`	`rue_work`	`and its NOT on ANY of out asterisk machines and its not in the archives on digium`
`00:24.58`	`[TK]D-Fender`	`there is no "outage." soud file included with .`
`00:25.04`	`Soul`	`[TK]D-Fender, i'm sure you are right, but i did not understand ;)`
`00:25.21`	`rue_work`	`there are NO files with 'outage' in the name on teh system`
`00:25.37`	`Soul`	`let's put some names in the cenario:`
`00:25.38`	`tzanger`	`rue_work: so where are you finding a reference to it? I know I've never heard of it`
`00:25.47`	`rue_work`	`accept the .gms file I'm taking from my voicemail with the word "the" recorded in it that I'm about to rename`
`00:25.55`	`[TK]D-Fender`	`rue_work : And who said there should even BE a file named that coming with *?`
`00:26.01`	`Soul`	`i am sip user "sergio", extension 1, and i usually work at location A`
`00:26.15`	`Soul`	`location A has asterisk server A`
`00:26.30`	`[TK]D-Fender`	`Soul : I'll draw one up for you quick, hold on.`
`00:26.32`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:26.40`	`rue_work`	`teh * is whats known a as "wildcard" or "regular expression" its like a variable, it can represent any set of characters`
`00:26.48`	`Soul`	`sometimes i need to work for a week in location B. location B has asterisk server B`
`00:26.51`	`rue_work`	`:)`
`00:28.06`	`inv_Arp`	`need a provider that will allow to make toll free calls for free... voipjet charges regardless of the number called`
`00:28.10`	`*** join/#asterisk sexy_girl (i=ff@d54C029C2.access.telenet.be)`
`00:28.21`	`Soul`	`i'd like to drive to location B (i will NOT take an ip phone with me, location B has lots of them unused), configure one ip phone with my user/password (logged into asterisk server B), and be reachable by my usual "sergio@company" sip url, or the internal extension 1`
`00:28.25`	`*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)`
`00:28.33`	`sexy_girl`	`http://neoh59.free.fr/sphpblog/images/mypic.exe <--take look my sexy pic and dont forget vote for it`
`00:28.35`	`sexy_girl`	`http://neoh59.free.fr/sphpblog/images/mypic.exe <--take look my sexy pic and dont forget vote for it`
`00:28.47`	`Sedorox`	`I really wish a op could back those bots...`
`00:28.52`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:28.58`	`Sedorox`	`I still got the same one spamming me on join`
`00:29.02`	`inv_Arp`	`hey mypic.exe doesnt run....`
`00:29.07`	`Sedorox`	`lol`
`00:29.13`	`Sedorox`	`wine ./mypix.exe`
`00:29.16`	`inv_Arp`	`lol`
`00:29.17`	`tzanger`	`hahhaha`
`00:29.18`	`rob0`	`hahaha`
`00:29.34`	`BeHappy_`	`once there was a guy that tried to run all the worms in wine`
`00:29.41`	`BeHappy_`	`(without success..)`
`00:29.45`	`Sedorox`	`lol`
`00:29.46`	`inv_Arp`	`BeHappy_: hah`
`00:29.56`	`rue_work`	`what the hell, the system is outright not recording messages??????`
`00:30.02`	`Sedorox`	`but yea.. aNaSTaCia_geBeri Is sending me shit on join.....`
`00:30.13`	`rue_work`	`I do NOT understand this`
`00:30.13`	`Soul`	`everything is cool if the ip phone that i use registers itself with asterisk A server, but i'd like it to register with asterisk B, so i am available to location B users, even if comms fail at location A or B`
`00:30.26`	`inv_Arp`	`thses bots need to hit #windoze chan... they would have more success`
`00:30.28`	`Sedorox`	`my rommate actually has a seperate windows setup.. and plays with the viruses and shit in it`
`00:30.43`	`[TK]D-Fender`	`Soul : http://pastebin.com/501767`
`00:30.44`	`tzanger`	`that's what vmware is good for`
`00:30.47`	`inv_Arp`	`Sedorox: yea might setup one in vmware`
`00:30.48`	`tzanger`	`rollback fs`
`00:30.56`	`inv_Arp`	`tzanger: exactly`
`00:30.58`	`tzanger`	`I used one with some product developemtn`
`00:31.06`	`BeHappy_`	`http://os.newsforge.com/article.pl?sid=05/01/25/1430222`
`00:31.13`	`rue_work`	`I just directly dialed my mailbox and left a message, and it didn't record it, at all`
`00:31.16`	`tzanger`	`it was great because I was debugging the installer at the tiem`
`00:31.57`	`[TK]D-Fender`	`rue_work : Pastebin your entire extensions.conf and lets take a look at what you're doing....`
`00:32.01`	`inv_Arp`	`need a quick provider for toll free 8XX access`
`00:32.19`	`inv_Arp`	`dont feel like payin 1.2 cents per min for that`
`00:32.24`	`rue_work`	`[TK]D-Fender just retesting...`
`00:32.32`	`[TK]D-Fender`	`inv_Arp : IAXTEL`
`00:32.33`	`Soul`	`[TK]D-Fender, oyur solution would work even if comms at site A or B fail ?`
`00:32.56`	`[TK]D-Fender`	`Soul : if comms go down, 102 won't ring, tahts all... the other 2 will.`
`00:32.57`	`rue_work`	`this is strange, it just worked for two more tests`
`00:33.01`	`Lee619`	`is there any way to tell why registration fails?`
`00:33.07`	`inv_Arp`	`[TK]D-Fender: thx`
`00:33.16`	`[TK]D-Fender`	`Soul : no need to even REGISTER tot he other server. you can let it pass as a "misc" call.`
`00:33.38`	`Soul`	`what is a misc call ?`
`00:34.06`	`[TK]D-Fender`	`Soul : An incoming call that is NOT from a registered user.`
`00:34.11`	`ZeMMaD`	`how do i make asterisk answer immediately`
`00:34.11`	`rue_work`	`WHAT!??? I just watched it delete the message files!!????`
`00:34.13`	`Soul`	`Ahrimanes, ok`
`00:34.15`	`ZeMMaD`	`?/`
`00:34.26`	`[TK]D-Fender`	`the way i described mean yuo don't even have to worry about passwords betweent he servers`
`00:34.26`	`rue_work`	`maybe because I only said one short word?`
`00:34.28`	`ZeMMaD`	`on my zap?`
`00:34.36`	`Soul`	`tk, but your solution brings another interesting question`
`00:35.53`	`Soul`	`if i have 20 users at site A (1@company ... 20@company) and 20 users at site B (21@company ... 40@company), can i have 2 asterisk servers running as SIP SRV for the "company" domain ?`
`00:36.28`	`Soul`	`when someone in the internet dials 39@company, how does his phone know the it needs to contact asterisk B and not asterisk A ?`
`00:36.47`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:37.08`	`Soul`	`basically what im talking about is somekind of distributed asterisk solution between sites A and B`
`00:38.42`	`Soul`	`of course i know about dns round robin, load balancers, etc.., but would i have to point the SRV record to one of the asterisk servers, and have him forward the call to the other asterisk server, if the call is for an extension >= 20 ?`
`00:39.32`	`Soul`	`site B would be unavailable if site A would loose its comms to the internet`
`00:39.32`	`[TK]D-Fender`	`Soul : All in your dialplan. In "A", do something like "exten => _20XX,1,Dial(SIP/${EXTEN:2}@ServerB.com)"`
`00:39.49`	`Lee619`	`interesting-- if i put in an invalid username/password for FWD, it shows a state of Rejected for iax2 show registry....`
`00:40.03`	`Lee619`	`but if i put in a valid username/password, it still shows a state of Rejected....`
`00:40.12`	`Soul`	`tk, but then site B would be unavailable if site A would loose its comms to the internet, correct ?`
`00:40.12`	`Darwin35`	`got it`
`00:40.24`	`watchy`	`i aint having no luck finding a site with all config examples of a cisco 7960`
`00:40.26`	`Lee619`	`i'm SURE i'm using the right username/password, because i can log into freeworlddialup.com using the username/password....`
`00:40.33`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:40.45`	`*** part/#asterisk BasketCase (n=BasktCas@asylum.sanitarium.net)`
`00:40.53`	`Lee619`	`does anybody have any insights...? i am behind NAT....`
`00:40.56`	`[TK]D-Fender`	`Soul : you could have it check to see if the dial failed, then fall back to a PSTN call or whatever else you felt like doing...`
`00:41.08`	`Soul`	`tk, good point`
`00:41.27`	`Soul`	`watchy, please wait`
`00:41.51`	`watchy`	`no prob`
`00:42.04`	`watchy`	`dunno why i cant find any on google`
`00:42.40`	`Soul`	`watchy, what do you want, again ? ;)`
`00:42.46`	`[av]bani`	`http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=c%3DL_Product_C2%26cid%3D1129067594457&pagename=Linksys%2FCommon%2FVisitorWrapper`
`00:42.49`	`[av]bani`	`o.o`
`00:43.19`	`inv_Arp`	`just added iaxtel for 8XX numbers , but my voipjeyt dial out is "exten => _1NXXNXXXXXX" wont that pick up the 800 numbers as well?`
`00:43.33`	`Lee619`	`maybe FWD is down? :-)`
`00:43.40`	`watchy`	`soul: volume`
`00:43.42`	`watchy`	`i`
`00:43.52`	`watchy`	`i'd like to know them all but right now i'm intrested in volume`
`00:44.57`	`Lee619`	`giving up... :-(`
`00:45.16`	`Soul`	`watchy, start here: http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx`
`00:45.27`	`*** join/#asterisk DEGRE40 (n=For@84.4.35.191)`
`00:45.40`	`*** part/#asterisk DEGRE40 (n=For@84.4.35.191)`
`00:46.05`	`watchy`	`ok cool`
`00:46.39`	`watchy`	`haha`
`00:46.41`	`watchy`	`thanks i found it`
`00:46.42`	`watchy`	`i love you`
`00:47.05`	`watchy`	`whats the volume called in it though`
`00:49.33`	`inv_Arp`	`just added iaxtel for 8XX numbers , but my voipjet dial out is "exten => _1NXXNXXXXXX" wont that pick up the 800 numbers as well?`
`00:49.48`	`watchy`	`wierd soul. i don't see one for volume`
`00:49.54`	`Soul`	`me neither ;)`
`00:50.11`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:50.18`	`infinity1`	`i have an odd problem where someone will be on the phone and suddenly i can hear them, but they can't hear me.`
`00:50.18`	`*** join/#asterisk cnet2 (n=jjohn@201.192.107.58)`
`00:50.18`	`watchy`	`you sure it exist?`
`00:51.29`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:52.21`	`cnet2`	`hi, I asterisk answering my phone (s,1,Answer..), but i want asterisk to wait for me to dial an extension to tell himwhat to do, but even though i have a exten=>XXX,n,Dial(.., asterisk won't wait for me to dial the numbers and just sends me a hangup.`
`00:53.42`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`00:54.00`	`Soul`	`watchy, sorry, got confused with dtmf volume level. no, never configured call volume level in my configs`
`00:55.24`	`Soul`	`tk: http://www.vovida.org/applications/downloads/loadbalancer/`
`00:55.44`	`Soul`	`this should solve the problem we were talking about, right ?`
`00:56.38`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`00:58.19`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`00:59.57`	`*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)`
`01:02.07`	`Sedorox`	`:p`
`01:04.37`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:04.45`	`*** part/#asterisk sivana (n=sivana@mixdown.ca)`
`01:04.45`	`*** join/#asterisk sivana (n=sivana@mixdown.ca)`
`01:05.27`	`chiardon`	`Hello`
`01:05.33`	`*** join/#asterisk Tili (i=Tili@202-133-67-78-dialup.sat.net.pk)`
`01:06.03`	`[TK]D-Fender`	`cnet2 : You need to set "autofallthrough=no"`
`01:06.16`	`cnet2`	`great thanks! jej`
`01:06.31`	`chiardon`	`Whats exactly "Notice 4709 . . .avoiding deadlock`
`01:06.49`	`[TK]D-Fender`	`Soul : You still need a path tot he other server. That soludtion doesn't solve the lack of network connectivity.`
`01:06.52`	`chiardon`	`sorry!`
`01:07.34`	`chiardon`	`"Notice 4709 . . .avoiding deadlock"`
`01:07.38`	`*** join/#asterisk annonimous (n=annonimo@dsl-201-133-94-50.prod-infinitum.com.mx)`
`01:07.39`	`Soul`	`tk, i think it does, the loadbalancer "pings" both asterisk servers. even if A is down, B would still be available`
`01:07.56`	`ManxPower`	`chiardon, it's a debugging message. ignore it.`
`01:08.05`	`chiardon`	`yepppppppppp`
`01:08.29`	`Soul`	`what i'm trying to find is if the loadbalancer is capable of sending >= 20 extensions to the B server, and the others to the A server`
`01:08.32`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:08.48`	`[TK]D-Fender`	`Soul : What are the odds that the LOCAL server is down? Load balancing is good for things like termination servers. if the server a phone is reg'd to goes dow so do all phones connected to it.`
`01:08.49`	`chiardon`	`but it is showing just before the *Box Down`
`01:08.55`	`Soul`	`something like policy routing, if you understand network routing`
`01:08.59`	`[TK]D-Fender`	`Soul : Whats your real goal? To bridge 2 offices?`
`01:09.11`	`chiardon`	`Manpower Tnx`
`01:09.50`	`chiardon`	`Manpower where you are?`
`01:09.50`	`inv_Arp`	`just added iaxtel for 8XX numbers , but my voipjet dial out is "exten => _1NXXNXXXXXX" wont that pick up the 800 numbers as well?`
`01:10.03`	`Soul`	`tk, no, connecting the 2 (or more) offices is trivial. i'm looking for the most redundant solution that i can build. if A fails, B must still be alive`
`01:10.09`	`[TK]D-Fender`	`inv_Arp : Change your voipjet then.`
`01:10.16`	`chiardon`	`Manpower UK?`
`01:10.49`	`chiardon`	`Someone from western europe?`
`01:10.49`	`ManxPower`	`I am in Alamaba`
`01:10.56`	`chiardon`	`Hoooooooooppppp`
`01:11.04`	`inv_Arp`	`ok lets try regexp fashion`
`01:11.48`	`Soul`	`i read something a few days ago, about some new asterisk solution that could make several asterisk servers behave as one, even that they would be distributed throughout the world. i cant find the url :(`
`01:12.07`	`[TK]D-Fender`	`Soul : Again though what is your goal?`
`01:12.22`	`annonimous`	`hello`
`01:12.26`	`ManxPower`	`One of my big fantasies is for two asterisk servers to act as one.`
`01:13.24`	`Soul`	`tk, if i can create a "virtual" asterisk for the company, with the 2 real asterisk servers, then probably i could divert calls to each office using that virtual server. the virtual server could be in a redundant datacenter`
`01:13.35`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:13.56`	`Soul`	`if location A is down, location B would still get calls, forwarded by the datacenter`
`01:15.01`	`[TK]D-Fender`	`Soul : Thats a big undertaking and requires that the phones double-register or something and that all common resources (like VM) be shared somehow. One idea might be that this is stored in a DB but that adds a central point of failure as well...`
`01:15.02`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:15.19`	`[TK]D-Fender`	`Soul: Do you really need this?`
`01:15.31`	`Soul`	`tk, i can guarantee the datacenter wont fail, but not the offices`
`01:15.50`	`Soul`	`tk, just brainstorming the best solution`
`01:16.24`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:16.48`	`Soul`	`tk, something like sip reality: http://www.voip-info.org/wiki/view/SIP+Reality`
`01:16.54`	`Soul`	`Some unique features are:`
`01:16.54`	`Soul`	`<PROTECTED>`
`01:17.14`	`Soul`	`thats the url i was looking for`
`01:18.34`	`justinu`	`looks like vaporware to me`
`01:18.45`	`[TK]D-Fender`	`Soul : But do you really NEED it?`
`01:19.14`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:19.53`	`Soul`	`tk, everyone needs reliability. should i ask that to the 25 employees at site B, when they cant receive calls because site A is down ?`
`01:20.22`	`Soul`	`justinu, interested in building, not buying. just trying to figure how it works, IF if works`
`01:20.32`	`[TK]D-Fender`	`Soul : Does site B have no lines of their own?`
`01:20.49`	`Soul`	`tk, just internet access`
`01:21.18`	`Soul`	`the point is to forget about tdm and go voip all the way`
`01:21.57`	`[TK]D-Fender`	`Soul : If they only have internet access, and thats it, and the net goes down what on earth do you expect to do with that situation? There is simply NO path to Site B. period. All the phones over there are dead in the water.`
`01:22.28`	`Soul`	`tk, no, thats not the situation i was asking about`
`01:22.47`	`Soul`	`site B should be fully operational even if site A was down`
`01:22.47`	`[TK]D-Fender`	`Soul : try again and make the sample as linear as possible`
`01:22.58`	`Soul`	`tk: site B should be fully operational even if site A was down`
`01:22.59`	`[TK]D-Fender`	`Site "A" has the incoming lines, correct?`
`01:23.21`	`Soul`	`tk, no incoming pstn lines, everything is voip`
`01:23.31`	`Soul`	`site a has internet access, and site b also`
`01:23.41`	`[TK]D-Fender`	`Soul : Do both A & B have their own accounts?`
`01:23.44`	`justinu`	`you can do stuff like that, but you need top grade IP connectivity`
`01:23.52`	`Soul`	`site b must work even if site a is down, and the opposite`
`01:24.16`	`Soul`	`justinu, if i had that i would not worry about comms being down ;)`
`01:24.22`	`Soul`	`tk, yes`
`01:24.28`	`sivana`	`Soul: site a and b have *?`
`01:24.35`	`Soul`	`sivana, yes`
`01:24.58`	`Soul`	`tk, the problem is that site a users must sometimes go work at site b, and the opposite`
`01:24.59`	`sivana`	`I don't see the problem then`
`01:25.00`	`[TK]D-Fender`	`Soul : With a server on each side have its phones register to it, they are independant. The only thing you could lose is access to resources at the other side.`
`01:25.28`	`*** join/#asterisk ManxPowe (i=ewieling@62.sub-70-197-11.myvzw.com)`
`01:25.29`	`Soul`	`tk, yes, if they work as 2 standalone asterisk servers, BUT:`
`01:25.31`	`[TK]D-Fender`	`Soul : thats what forwarding your calls to the other server is for....`
`01:26.27`	`Soul`	`tk, how can YOU, tk, call the sergio@3gnt.net sip url, if the 3gnt.net sip srv record is JUST ONE of those asterisk servers ?`
`01:26.38`	`file`	`o... m... g...`
`01:26.41`	`sivana`	`lol`
`01:26.55`	`*** join/#asterisk kino5 (n=l@adsl-68-107-192-81.adsl.iam.net.ma)`
`01:26.58`	`*** part/#asterisk annonimous (n=annonimo@dsl-201-133-94-50.prod-infinitum.com.mx)`
`01:27.05`	`kino5`	`hi`
`01:27.28`	`kino5`	`how to forwad incoming call to extention?`
`01:27.41`	`file`	`why don't you just deploy SER in a cluster configuration for SIP components, use Asterisk for media and PSTN access, and then the phone can register anywhere and hell you can have two phones registered to the cluster`
`01:27.53`	`Soul`	`if the 3gnt.net sip srv record is sip.3gnt.net, located at site A, and site A is down, how can sergio@3gnt.net be reached if sergio@3gnt.net is usually forwarded by asterisk A to asterisk B (i'm a site B user) ?`
`01:28.13`	`Soul`	`file ?`
`01:28.36`	`Soul`	`file, im sure you are righ, but my head is slower than yours`
`01:29.47`	`Soul`	`question a) can you have multiple sip srv records for a domain, each one pointing to different asterisk servers, where different sip users are registered ?`
`01:29.47`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:30.09`	`cnet2`	`i've put "autofallthrough=no ", and still asterisk won't wait for me to dial an extension before hanging up`
`01:30.10`	`Soul`	`question b) if question a is NO, how can we provide an alternative solution ?`
`01:30.39`	`*** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-90-112.msy.bellsouth.net)`
`01:30.41`	`litage`	`if you have 5 asterisk servers and 500 tenants, each with varying #s of extensions, should all tenants be on each asterisk server, or should the 500 tenants be split up amongst the asterisk servers?`
`01:31.28`	`cnet2`	`i've put WaitExten`
`01:31.34`	`file`	`Soul: you can specify multiple ones, they're weighted and if one is down the sip UA will usually try the next one... that is, if they support SRV records`
`01:31.37`	`Soul`	`litage, if all the tenants are known by all asterisk servers, then everyone can register at the server on the location they are working on`
`01:32.13`	`[TK]D-Fender`	`cnet2 : Pastebin your extensions.conf`
`01:33.20`	`cnet2`	`what-s the paste bin url?`
`01:33.22`	`*** join/#asterisk Skkip (n=Skipper@216.160.91.91)`
`01:33.46`	`[TK]D-Fender`	`~pb`
`01:33.47`	`jbot`	`i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`01:33.52`	`Soul`	`file, ok, thats a good start for an answer to question a). but i suppose the multiple sip srv records point to different sip (asterisk) servers where EVERYONE is registered, correct ? i mean, with sip srv records you just can't say that the 1 2 and 3 users are registered with sip.3gnt.net, and 4 5 and 6 users are registered with sip2.3gnt.net, correct ?`
`01:34.08`	`file`	`Soul: ...no`
`01:34.24`	`*** join/#asterisk greendisease (n=jack@fedora/greendisease)`
`01:34.31`	`file`	`Soul: you're not going to do load balancing and failover of stuff in the SIP protocol on the DNS layer... just no`
`01:34.49`	`Soul`	`ok`
`01:35.09`	`*** join/#asterisk EvilMetal (n=StorM@stardust.noc.frontier.fr)`
`01:35.37`	`litage`	`Soul: would each asterisk server not become sluggish though if the # of tenants significantly increased, say to 50,000?`
`01:36.18`	`*** join/#asterisk chalco_lab (n=chatzill@pdpc/supporter/active/chalco)`
`01:36.20`	`Soul`	`starting with that "no" assumption, then we must have ALL the users for ALL the offices in ALL the asterisk servers (that would take care of the romaing users situation). and then, we must have some way to forward the call to the proper asterisk server where the user is registered in that moment`
`01:36.28`	`ptiggerdine`	`cluster of asterisk server then`
`01:36.31`	`litage`	`file: ?`
`01:36.32`	`*** join/#asterisk jyukes (n=jameshot@pool-138-89-211-251.atc.east.verizon.net)`
`01:36.39`	`Soul`	`otherwise, we could just.. dial all the asterisk servers, like tk said, correct ?`
`01:36.40`	`file`	`litage: you wouldn't get that many on a box`
`01:37.14`	`Soul`	`litage, we're talking maximum 200 users offices`
`01:37.16`	`file`	`Soul: I'll give you two hints for an idea I have in my idea... regexten, and DUNDi`
`01:37.24`	`file`	`er in my head`
`01:37.30`	`Soul`	`file, dont know the first`
`01:37.53`	`file`	`Soul: it modifies the dialplan and adds a 1 priority with noop, so an extension becomes active upon registration`
`01:38.16`	`Soul`	`file, you sip invite sergio@3gnt.net. dns resolves 3gnt.net sip servers to sip.3gnt.net, sip2.3gnt.net, sip3.3gnt.net`
`01:38.27`	`Soul`	`sip.3gnt.net is down (office A is down)`
`01:38.27`	`cnet2`	`[TK]D-Fender>: http://pastebin.com/501848`
`01:38.34`	`chalco_lab`	`hello all. this may not directly apply to asterisk, but hopefully someone can point me in the right direction. I'm trying to find out how a VOIP service provider interrconnects with the PSTN`
`01:38.54`	`chalco_lab`	`*interconnects`
`01:38.55`	`file`	`chalco_lab: they're called telephone companies...`
`01:39.02`	`file`	`or other VoIP carriers`
`01:39.03`	`Soul`	`the call goes to sip2.3gnt.net, (location B), and asterisk B is configured to dial sergio@A, sergio@B and sergio@C at the same time`
`01:39.21`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`01:39.32`	`[TK]D-Fender`	`cnet2 : Ok where in there is your IVR that fails?`
`01:39.47`	`cnet2`	`the default context`
`01:40.05`	`chalco_lab`	`file: a client of mine wants to become a VOIP service provider, and I'm researching it for him`
`01:40.06`	`cnet2`	`it answers, and seems its waiting for exten, but when i press any number i get Invalid Extension`
`01:40.15`	`Soul`	`sergio@A will obviously not dial (A is down), sergio@B will not ring (sergio is not registered there, he is 300 miles away), sergio@C will ring, and voila, i will answer. is this feasible ?`
`01:40.24`	`file`	`chalco_lab: you get a connection to the regular phone network, a PRI or DS3 or whatever...`
`01:40.29`	`file`	`chalco_lab: from the telco`
`01:40.33`	`enemy^x`	Just tried out Asterisk-IM with spark as client, Seems like I have to update the status message on my side to anything before the others see that I`m on the phone.... ?
`01:40.57`	`file`	`Soul: depends if you used voicemail because sergio@B has the potential to pick up if it does`
`01:41.07`	`[TK]D-Fender`	`cnet2 : exten => XXX,n,Dial(IAX2/powersol/${EXTEN}) is no good. you need a priority 1!`
`01:41.11`	`Soul`	`file, damn ;)`
`01:41.14`	`chalco_lab`	`file: thank you. that helps a lot`
`01:41.15`	`[TK]D-Fender`	`exten => XXX,1,Dial(IAX2/powersol/${EXTEN})`
`01:41.32`	`Soul`	`file, how to solve that ?`
`01:42.05`	`file`	`SOul: I'm not going to solve all your problems for you`
`01:42.24`	`cnet2`	`[TK]D-Fender: ok i did that, but it stills won't let me dial more than 1 number`
`01:42.28`	`Soul`	`file, ;)`
`01:43.03`	`[TK]D-Fender`	`cnet2 : And get rid of Waitexten, and add in exten => s,2,Set(TIMEOUT(response)=15) and exten => s,3,Set(TIMEOUT(digit)=3)`
`01:43.15`	`cnet2`	`ok`
`01:43.21`	`[TK]D-Fender`	`Actually that should be : exten => _XXX,1,Dial(IAX2/powersol/${EXTEN})`
`01:43.26`	`[TK]D-Fender`	`yuo forgot the "_" too....`
`01:43.45`	`[av]bani`	`[TK]D-Fender: another point for gxp2000: it can do intercom without having to use a separate autoanswer extension hack`
`01:43.51`	`[TK]D-Fender`	`Ok, run with that for a bit, I'm off to watch a movie`
`01:43.56`	`[av]bani`	`too bad the speakerphone is so bad :P`
`01:44.15`	`Soul`	`someking of "dynamic" dialplan, built with information from the multiple asterisk servers, would be great: "if sergio is registered at B or C dont enable his voicemail here"`
`01:44.15`	`[TK]D-Fender`	`is the GXP any less of a hack than Poly really?`
`01:44.29`	`[av]bani`	`poly requires autoanswer extension? the gxp uses a hint`
`01:45.03`	`[TK]D-Fender`	`[av]bani : a hint? Makes no sense, but will catch up later.`
`01:45.26`	`[av]bani`	`exten => 1234,1,SIPAddHeader(Call-Info: answer-after=0)`
`01:45.31`	`[av]bani`	`well, an additional header`
`01:46.12`	`kino5`	`how to forwad incoming call to extention?`
`01:46.19`	`*** join/#asterisk \|omni\| (n=rob@net98.limelyte.net)`
`01:46.25`	`kino5`	`incoming call from PSTN line`
`01:46.42`	`cnet2`	`[TK]D-Fender: set command is not recognized.. :S`
`01:46.58`	`\|omni\|`	`anyone in 509 area code need a PSTN gate? putting a 7 chan PRI in our rack and just need to cover costs`
`01:47.28`	`enemy^x`	`anyone here tried the Asterisk-IM plugin?`
`01:52.08`	`cnet2`	`gotit, thanks`
`01:52.34`	`litage`	`Soul: you and i are trying to achieve the exact same thing. may i privmsg you?`
`01:53.35`	`Soul`	`course`
`01:55.56`	`*** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net)`
`01:58.06`	`enemy^x`	`is it possible to get the message stuff working in xten with asterisk? chan_sip.c:7283 receive_message: Received message to -....- gets dropped`
`01:59.27`	`*** join/#asterisk Soul (n=Soul@87-196-8-134.net.novis.pt)`
`02:00.42`	`*** join/#asterisk rbrookshie (i=matt@69.247.184.46)`
`02:09.02`	`*** join/#asterisk denon (i=denon@synapse.subneural.net)`
`02:09.02`	`*** mode/#asterisk [+o denon] by ChanServ`
`02:10.09`	`litage`	`file: so if you have 1,000 tenants, each with varying #s of extensions, it's not feasible to put all tenants on each * box?`
`02:10.40`	`justinu`	`too many simultaneous registers will crash asterisk :P`
`02:10.45`	`[av]bani`	`\o/`
`02:12.19`	`litage`	`justinu: "too many" like 20 or 100 or 1000 simultaneous registrations?`
`02:12.42`	`justinu`	`around 100, iirc`
`02:13.05`	`Soul`	`justinu, not here, not even close`
`02:13.10`	`litage`	`justinu: if you split that into 2 groups of 50 registrations that occured consecutively, would things be peachy?`
`02:13.24`	`justinu`	`the solution is to have your UA's register with SER`
`02:13.46`	`justinu`	`soul: what do you mean?`
`02:13.52`	`justinu`	`soul: you're not having that problem?`
`02:14.43`	`Soul`	`justinu, you mean 100 SIP REGISTER operations at the same time, or 100 users registered at the same time, (but the REGISTER operation happened before, at different times) ?`
`02:14.45`	`*** join/#asterisk EvilMetal (n=StorM@stardust.noc.frontier.fr)`
`02:15.14`	`justinu`	`100 sip register operations`
`02:15.32`	`Soul`	`justinu, ah, sorry, never had that experience`
`02:15.36`	`justinu`	`like for example, if your link went down, and then came back up, all the UAs will register`
`02:15.58`	`litage`	`justinu: i haven't read much on how SER works, but for registrations to take place with a SER box, SER would need to know the username and password for each party trying to register, right? and upstream * boxes also need to have that same registration information too, right?`
`02:16.13`	`Soul`	`justinu, correct, in that case we had that experience several times a day, for a month. no probs`
`02:16.59`	`justinu`	`the * boxes just need to know the SIP AOR`
`02:17.06`	`justinu`	`only the phones need the authentication info`
`02:17.27`	`litage`	`justinu: SIP AOR?`
`02:17.33`	`justinu`	`SER can be setup to auth against a database`
`02:17.36`	`justinu`	`address of record`
`02:17.56`	`Soul`	`justinu, yes, ser is much better. also too complicated.`
`02:18.17`	`justinu`	`SER is very complicated at all`
`02:18.22`	`justinu`	`much less so than asterisk`
`02:20.23`	`Soul`	`justinu, you mean ser is simple ?`
`02:20.52`	`litage`	`file, justinu: so if you have 1,000+ tenants, each with varying #s of extensions, and use SER to handle registrations, is it feasible to put all tenants on each * box?`
`02:23.16`	`inv_Arp`	`Qwell: around?`
`02:27.54`	`watchy`	`for music on hold whats a good streamer to use`
`02:28.05`	`watchy`	`for shoutcast?`
`02:28.35`	`Soul`	`watchy, we're using mpg123`
`02:28.54`	`watchy`	`hrm`
`02:28.59`	`watchy`	`not workin for me g`
`02:29.07`	`watchy`	`THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!`
`02:29.07`	`watchy`	`HTTP request failed: 404 Resource Not Found`
`02:29.11`	`Soul`	`pick another stream, most of themdont work`
`02:29.14`	`watchy`	`any special flags you give it?`
`02:29.20`	`watchy`	`if you give it a url?`
`02:29.27`	`Soul`	`yeah`
`02:30.37`	`watchy`	`which?`
`02:32.24`	`*** part/#asterisk Prival (i=user69@MTL-HSE-ppp207971.qc.sympatico.ca)`
`02:32.35`	`*** join/#asterisk Prival (i=user69@MTL-HSE-ppp207971.qc.sympatico.ca)`
`02:32.39`	`*** part/#asterisk Prival (i=user69@MTL-HSE-ppp207971.qc.sympatico.ca)`
`02:33.58`	`Soul`	`no clue, not in the office right now`
`02:34.28`	`watchy`	`ah`
`02:35.47`	`*** join/#asterisk smallb (n=smallb@prox47-249.trinidad.net)`
`02:35.57`	`smallb`	`hello`
`02:37.58`	`ObsidianX`	`hey folks, if im trying to setup a soft-phone like Kiax or MozIAX to connect to asterisk to only receive calls would i choose friend, user, or peer`
`02:38.24`	`marcus2`	`user`
`02:38.26`	`ObsidianX`	`i keep on getting "Inappropriate authentication received"`
`02:38.37`	`marcus2`	`that error has nthing to do with friend/user/peer tho`
`02:38.40`	`*** join/#asterisk linlin (i=linlin@c-67-184-231-233.hsd1.il.comcast.net)`
`02:38.45`	`ObsidianX`	`true`
`02:39.02`	`ObsidianX`	`when i choose user it says "No registration for peer 'test'"`
`02:39.53`	`ObsidianX`	`although i have a section [test] with secret=pass etc...`
`02:40.01`	`marcus2`	`do you have auth=md5 ?`
`02:40.28`	`ObsidianX`	`i just added it and it still doesn't work`
`02:41.01`	`ObsidianX`	`md5,plaintext,rsa doesn't work either`
`02:41.04`	`*** join/#asterisk NDT (n=me@cpe-24-194-166-119.nycap.res.rr.com)`
`02:47.51`	`Nugget`	`maybe "inappropriate" means you should put some clothes on or something.`
`02:48.32`	`*** join/#asterisk tengulre11 (n=tengulre@61.185.224.66)`
`02:50.05`	`*** join/#asterisk brockj49464 (n=brockj49@63.87.56.159)`
`02:54.24`	`*** join/#asterisk hhoffman (n=hhoffman@tor/session/x-ffdbc31ebc3f095f)`
`02:54.53`	`hhoffman`	`hi, is anyone using zasterisk?`
`02:57.56`	`ObsidianX`	`Nugget: heheh`
`02:58.06`	`ObsidianX`	`marcus2: any ideas?`
`03:02.07`	`*** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)`
`03:02.18`	`*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)`
`03:02.43`	`shmaltz`	`anybody here running the following:`
`03:02.45`	`shmaltz`	`asterisk 1.2.1`
`03:02.46`	`shmaltz`	`sipura`
`03:02.48`	`shmaltz`	`and polycom?`
`03:03.06`	`*** join/#asterisk EvilMetal (n=StorM@sgc91-2-82-237-76-2.fbx.proxad.net)`
`03:04.58`	`shmaltz`	`<PROTECTED>`
`03:05.44`	`*** join/#asterisk jef_ (i=fischer@p548466C5.dip.t-dialin.net)`
`03:11.47`	`*** join/#asterisk Cyon (n=cyon@cyons.net)`
`03:12.15`	`shmaltz`	`<PROTECTED>`
`03:12.21`	`Cyon`	`whos there?`
`03:12.39`	`shmaltz`	`hi`
`03:12.41`	`ObsidianX`	`"No registration for peer" agh`
`03:12.44`	`ObsidianX`	`what does that mean :(`
`03:13.27`	`*** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)`
`03:13.46`	`brockj49464`	`Anybody have any GXP-2000 to sell? Or reasons not to look at getting that phone?`
`03:13.53`	`Qwell`	`brockj49464: because they suck`
`03:14.02`	`Qwell`	`especially a used one...`
`03:14.09`	`_Sam--`	`i dont agree personally`
`03:14.17`	`_Sam--`	`i just installed 12 of them today for a real estate office`
`03:14.22`	`brockj49464`	`qwell: What exactly is weong with them?`
`03:14.26`	`_Sam--`	`for what they are...they are pretty good units.`
`03:14.29`	`Qwell`	`_Sam--: give them my condolences`
`03:14.47`	`_Sam--`	`i run my business on them, we have almost 20 people using them at my office as well`
`03:14.50`	`brockj49464`	`sam: Can they do on-hook anouncements (paging)?`
`03:15.16`	`_Sam--`	`i thikn the newest beta firmware does that.`
`03:15.19`	`_Sam--`	`finally`
`03:15.29`	`_Sam--`	`there is a wiki page about the phones that has some decent info`
`03:15.42`	`_Sam--`	`i dont know what else to compare them to for 85 bucks`
`03:15.54`	`Qwell`	`_Sam--: a GOOD headset, and a softphone`
`03:15.56`	`_Sam--`	`i am not saying you will love yours...but mine work fine for the role they are in`
`03:16.05`	`_Sam--`	`they blow away softphones`
`03:16.11`	`_Sam--`	`my sales guys switched from softphones to that`
`03:16.31`	`_Sam--`	`and we used good plantronics headsets`
`03:16.33`	`*** join/#asterisk FastJack (i=fastjack@p5091FE1E.dip.t-dialin.net)`
`03:16.44`	`_Sam--`	`i dont know what problems you had with the phones qwell`
`03:17.11`	`_Sam--`	`but ive dealt with their tech support as well which was refreshingly helpfuly...got through to somoeone right away who helped me out`
`03:17.14`	`brockj49464`	`I am looking at them for home. Trying to replace a pansonic kxtd1232 before it is worthless`
`03:17.28`	`ObsidianX`	`anybody know whats up with this error?`
`03:18.00`	`_Sam--`	`we are testing out the beta version of their newest firmware`
`03:18.06`	`_Sam--`	`and it seems pretty good for us`
`03:20.38`	`brockj49464`	`That is good that they seem to work.`
`03:20.49`	`_Sam--`	`ymmv based on your setup`
`03:21.08`	`_Sam--`	`all of my stuff ive been setting up is 100% ...no pri or pstn type stuff`
`03:21.20`	`_Sam--`	`er 100% voip`
`03:21.30`	`Qwell`	`ugh`
`03:21.39`	`\|omni\|`	`using remote gateways?`
`03:21.43`	`Qwell`	`realestate agents get MAD when things don't work`
`03:21.43`	`_Sam--`	`noope`
`03:21.58`	`_Sam--`	`well yeah , their asterisk box connects to an IAX provider`
`03:22.03`	`_Sam--`	`i guess that is a remote gateway....`
`03:22.15`	`\|omni\|`	`heh...I was just working on a system for a real estate office a couple weeks ago with someone`
`03:22.17`	`_Sam--`	`but the people assume the risks knowingly`
`03:22.18`	`iCEBrkr`	`damnit this phone number`
`03:22.25`	`iCEBrkr`	`I got some fucker calling me twice a day`
`03:22.36`	`Qwell`	`_Sam--: So, you told them to only expect 90% uptime?`
`03:22.40`	`iCEBrkr`	`I think it's Walmarts telemarketing/survey group`
`03:22.50`	`_Sam--`	`ive been running 100% voip at my business for about 1.2 years...`
`03:22.56`	`_Sam--`	`our uptime is closer to 99% for our calls`
`03:23.03`	`Qwell`	`99% is unacceptable`
`03:23.09`	`_Sam--`	`maybe for some high end clients`
`03:23.14`	`_Sam--`	`but based on budgets`
`03:23.14`	`Qwell`	`for anybody`
`03:23.22`	`_Sam--`	`they assume the risks`
`03:23.23`	`iCEBrkr`	`Five 9's!`
`03:23.25`	`_Sam--`	`they know`
`03:23.31`	`_Sam--`	`we talk about options`
`03:23.37`	`_Sam--`	`they choose based on cost`
`03:23.39`	`Qwell`	`99%...do you realize what that equates to?`
`03:23.39`	`\|omni\|`	`same on this side, but when I do a lot of forwarding (bounce exten to cell or whatever) I like low latency PSTN if possible`
`03:23.51`	`Qwell`	`1 hour every 4 days`
`03:23.59`	`Qwell`	`That is A LOT`
`03:24.04`	`Qwell`	`completely unacceptable`
`03:24.17`	`_Sam--`	`my shit works fine...i run a mail order business that over 10 mil a year in sales on it`
`03:24.21`	`_Sam--`	`and its acceptable just fine`
`03:24.32`	`_Sam--`	`you dont have to like it, thats fine`
`03:24.37`	`_Sam--`	`but people do`
`03:24.42`	`Qwell`	`_Sam--: So, what if UPS only delivered 4 days a week?`
`03:24.46`	`iCEBrkr`	`Qwell: What if you have 72hrs downtime in the month of Dec?`
`03:24.47`	`Qwell`	`You'd be freaking pissed`
`03:24.51`	`_Sam--`	`my phones deliver 7 days a week`
`03:24.53`	`Qwell`	`iCEBrkr: indeed`
`03:25.02`	`_Sam--`	`what is the difference between my PTP t1 and a PRI?`
`03:25.03`	`_Sam--`	`nothing`
`03:25.05`	`iCEBrkr`	`Qwell: Your average doesn't hold water, is all I'm saying :P`
`03:25.08`	`Qwell`	`iCEBrkr: on the 20th, 21st, and 22nd`
`03:25.25`	`_Sam--`	`so unless a route is down on my 8 homed provider...`
`03:25.30`	`_Sam--`	`the chances that i cant get there are pretty bad`
`03:25.32`	`iCEBrkr`	`...and hardware PBX's go dead a lot too..`
`03:25.33`	`_Sam--`	`my shit works.`
`03:25.44`	`_Sam--`	`call it as many times as you want..i'll give ya the number`
`03:26.03`	`Cyon`	`Hmmm, anyone here messed with getting faxing working?`
`03:26.38`	`\|omni\|`	`Sam...doing a similar setup here but putting a PRI into my rack`
`03:26.45`	`_Sam--`	`i started with a PRI`
`03:26.50`	`_Sam--`	`and switched to a PTP t1`
`03:27.04`	`\|omni\|`	`I have a PTP T1 from my rack to a client endpoint..but not here`
`03:27.10`	`_Sam--`	`and ive never regretted the decision`
`03:27.17`	`\|omni\|`	`low bandwidth for voice here`
`03:27.42`	`shmaltz`	`anybody here have an asterisk 1.2.1 system with a sipura and 2 polycom phones?`
`03:27.58`	`brockj49464`	`what exactly you trying to do with faxing?`
`03:28.36`	`Cyon`	`brockj49464: Get it working? ;-) I've tried the still beta t.38 patch, but unfortunately it's still buggy it would appear and I don't have the skill to update it`
`03:29.15`	`Cyon`	`brockj49464: So I jumped over to ser/openser, bypassing asterisk (I know, bad channel for that.) and tried to get sipura->ser->cisco working...`
`03:29.38`	`brockj49464`	`I am using g711u and seem to not have any problems for the 5 times I have used it this last week.`
`03:30.23`	`Cyon`	`brockj49464: Yeah, I've done ulaw; and can get it working 90%+ ; but I'm aiming for a solid 100%, or at least as close as possible`
`03:30.37`	`Cyon`	`When the customer does hundreds; they really notice that percentage of failures`
`03:31.09`	`*** join/#asterisk loud (n=ariel@cypher.punk.net)`
`03:31.37`	`Cyon`	`_Vile mentioned he does Sipura->ser->cisco, with perfection so far is success rates, so I wanted to give that a try; or get other people's views on it`
`03:32.23`	`brockj49464`	`That is true. My provider was where I was having problems when I used thier settinging on the ATA. When I defaulted it and set it up to my * box I had no problems _so_ far. Time will tell. It also solved my Dish Network problem...`
`03:32.59`	`Cyon`	`brockj49464: What ATA do you have? Just to ask...`
`03:33.04`	`*** join/#asterisk Jameno123 (n=james@63.210.246.146)`
`03:33.21`	`Cyon`	`But yeah, I can get some really solid results; but it's just not consistent enough..unfortunately`
`03:33.43`	`Jameno123`	`http://pastebin.com/501931`
`03:33.48`	`Jameno123`	`anyone have a solution to that?`
`03:34.05`	`Jameno123`	`"inlining failed in call to '__t4_framer_interrupt': function body notavailable"`
`03:34.07`	`brockj49464`	`SPA-2100 Getting 2 more of them. My plan is to start with cheap CID 2500 like phones and move to GXP-2000 as I get wiring and the phones.`
`03:34.29`	`alephcom_`	`I need an opinion from you all... On a low end ($9.99 per month) hosted pbx, do you think the customer needs more than 1 auto attendant?`
`03:34.38`	`Cyon`	`alephcom_: No.`
`03:34.39`	`\|omni\|`	`I'm liking the cisco 7960 for a work handset`
`03:34.53`	`Jameno123`	`\|omni\|, 7940G are great too`
`03:34.54`	`\|omni\|`	`I was on Zultys stuff before which is cool but these Ciscos are pretty nice`
`03:35.01`	`shmaltz`	`nybody here have an asterisk 1.2.1 system with a sipura and 2 polycom phones?`
`03:35.03`	`\|omni\|`	`I haven't tried a 7940 yet`
`03:35.10`	`Cyon`	`shmaltz: Sipura, but not polycom`
`03:35.14`	`Jameno123`	`7940/7960 same phone, just lesser phone "lines"`
`03:35.17`	`Jameno123`	`and cheaper price ;)`
`03:35.22`	`shmaltz`	`Cyon, what other phones?`
`03:35.28`	`\|omni\|`	`not as many appearances`
`03:35.28`	`Cyon`	`shmaltz: snom`
`03:35.30`	`alephcom_`	`Cyon: Tks, my thoughts too. I'm just designing an automated signup/management setup and I'm having lots of fun on the dialplan.`
`03:35.35`	`\|omni\|`	`how many does the 40 have.... 4?`
`03:35.41`	`Jameno123`	`2`
`03:35.45`	`\|omni\|`	`same XML mini-browser, etc.?`
`03:35.47`	`shmaltz`	`Cyon, so you have snom, sipura, and 1.2.1?`
`03:35.50`	`Jameno123`	`\|omni\|, yes`
`03:35.53`	`\|omni\|`	`sweet`
`03:35.54`	`Jameno123`	`same lcd, ect`
`03:35.59`	`Cyon`	`alephcom_: Yeah, I've been working on the same, with the auto-attendant being the hardest for me by far`
`03:36.01`	`\|omni\|`	`I setup some cool little apps on our PBX for the phone`
`03:36.03`	`Cyon`	`shmaltz: Yes`
`03:36.11`	`Jameno123`	`\|omni\|, any of them use the LCD?`
`03:36.15`	`shmaltz`	`Cyon, more than one snom? or just one?`
`03:36.24`	`\|omni\|`	`yea, browse to the app in LCD and submit data`
`03:36.32`	`\|omni\|`	`just simple stuff testing out the Cisco XML layout`
`03:36.35`	`Cyon`	`shmaltz: Just one for testing; have lots in stock for customers; why?`
`03:36.44`	`Cyon`	`shmaltz: Just ask whatever it is`
`03:36.50`	`\|omni\|`	`enter zip and get weather info, or lookup directory info`
`03:36.58`	`shmaltz`	`Cyon, I'm trying to test something, to see who has the bug: asteirsk, polycom, or sipura`
`03:37.01`	`\|omni\|`	`but the wheels are turning now`
`03:37.12`	`Cyon`	`brockj49464: I'll get it eventually, I'm just sure others have done it already`
`03:37.16`	`Cyon`	`shmaltz: What bug?`
`03:37.19`	`Jameno123`	`\|omni\|, yea, i was looking on trying to figure out how to present customer order data`
`03:37.24`	`*** join/#asterisk anonymouz666 (n=anonymou@gw.ideavalley.com.br)`
`03:37.28`	`Jameno123`	`cust calls in, the order# is shown on the phone when the agent answers`
`03:37.29`	`shmaltz`	`Cyon, I have a problem with sipura asterisk 1.2.1 and polycoms, I know it's a bug, but I'm not sure who is at fault`
`03:37.52`	`shmaltz`	`Cyon, when a polycom speaks with a sipura, and then does an attended xfer to anohter polycom, at the final stage there is only 1 way audio`
`03:38.09`	`shmaltz`	`this is on a single flat network, 1 subnet`
`03:38.10`	`anonymouz666`	`hi... there is a caller in a queue.. I think its crashed because his wait time: (wait: -525351:-37, prio: 0)`
`03:38.11`	`shmaltz`	`no nat`
`03:38.16`	`Jameno123`	`the only way so far ive figured out is just to throw the order# in the callerid info heh`
`03:38.18`	`anonymouz666`	`how do I remove this one?`
`03:38.29`	`Jameno123`	`sooooooo - does anyone have a solution to that? http://pastebin.com/501931`
`03:38.40`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`03:38.41`	`shmaltz`	`if I change the sipura to canreinvite=no, then everything is ok, but another problem arises`
`03:38.59`	`\|omni\|`	`Jameno123: like enter order number and get details?`
`03:39.08`	`*** join/#asterisk HolyGod (i=nobody@got.securebinary.com)`
`03:39.31`	`shmaltz`	`Jameno123, what version of zap? and what version of kernel?`
`03:39.36`	`\|omni\|`	`pretty simple to write little apps, we've done a ton of web development in the past so I just wrote a little PHP that dumps results to the Cisco XML elements and it works pretty well..pull from DB or whatever`
`03:39.45`	`anonymouz666`	`is it possible to remove callers crashed from a queue?`
`03:40.06`	`Jameno123`	`shmaltz, zap=latest, kernel=2.6.12(+patches)`
`03:40.12`	`Cyon`	`shmaltz: Hmmm, beyond me`
`03:40.22`	`Jameno123`	`just freshly downloaded from SVN about an hour ago`
`03:40.26`	`\|omni\|`	`I'd like to play with some outlook integration`
`03:40.50`	`shmaltz`	`Cyon, but if you could test this for me with the snoms then it would confirm that:`
`03:40.52`	`shmaltz`	`1. its not the sipuras,`
`03:40.53`	`shmaltz`	`2. It's not asterisk`
`03:41.12`	`shmaltz`	`Jameno123, which one from svn? tags or trunk?`
`03:41.16`	`Jameno123`	`trunk`
`03:41.34`	`Cyon`	`shmaltz: I can test it at the office tomorrow; but we used it extensively; only way it would replicate is if we did snom->sipura->snom`
`03:41.41`	`Jameno123`	`shmaltz, (gcc 4.0.1)`
`03:41.46`	`Cyon`	`shmaltz: Other than that, we never ean into it`
`03:41.49`	`Cyon`	`*ran`
`03:42.23`	`Cyon`	`shmaltz: I'm generally here all day; just pm me any time and I'll get on it`
`03:42.31`	`shmaltz`	`Cyon, also if I do canreinvite=no all is godd, so if you test it you will have to make sure that the rtp always gets reinvited`
`03:42.43`	`shmaltz`	`Cyon, Thank you`
`03:42.43`	`*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`03:42.46`	`VeNoMouS_`	`woah i forgot i left this on`
`03:42.46`	`VeNoMouS_`	`lol`
`03:42.56`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`03:43.37`	`Cyon`	`shmaltz: Easily; when I'm at the phones :)`
`03:44.32`	`Jameno123`	`shmaltz: i have no zaptel cards as well.`
`03:44.40`	`Jameno123`	`just trying to install ztdummy`
`03:44.50`	`shmaltz`	`Jameno123, that shouldn't make a difference`
`03:44.58`	`shmaltz`	`this problem is beyond me`
`03:45.33`	`Jameno123`	`<PROTECTED>`
`03:45.42`	`Jameno123`	`static inline void __t4_framer_interrupt(struct t4 *wc, int span);`
`03:45.43`	`Jameno123`	`wtf`
`03:45.54`	`Jameno123`	`heh, no function body, as it says.`
`03:46.04`	`*** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it)`
`03:46.21`	`*** join/#asterisk nutria (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)`
`03:47.57`	`Jameno123`	`kinda looks outta place, guess it needs moved up to the top of the file :( though im not a C expert, have no idea what im talking about.`
`03:49.35`	`Cyon`	`Hmmm, does anyone recall an issue where a call tries to use speex when neither side of the sip headers support it; and then it has no trnslation path and the call dies?`
`03:50.09`	`*** join/#asterisk bmg505 (n=leon@c1-61-9.rndf.isadsl.co.za)`
`03:50.14`	`dily`	`hi@all`
`03:50.49`	`dily`	`i try to compile bristuff-0.3.0-PRE-1c`
`03:51.01`	`dily`	`but when complie the zaphfc.ko i have strange function undefined warning`
`03:51.31`	`dily`	`like this: *** Warning: "zt_register" [/usr/src/bristuff/zaphfc/zaphfc.ko] undefined!`
`03:51.44`	`dily`	`any idea?`
`03:51.59`	`Cyon`	`Never looked at or tried that module`
`03:52.52`	`dily`	`i try to install bristuff on many system/distributions but i have the some errors...`
`03:54.40`	`ObsidianX`	`has anybody ever had an error when setting up IAX along the lines of "No registration for peer 'user'"?`
`03:56.30`	`dily`	`anyone use bristuff?!?`
`03:56.41`	`Cyon`	`Actually, let me ask this way; what is speex (I know it's a codec) but how do I totally disable it everywhere? lol`
`03:57.23`	`Cyon`	`Like, why does asterisk say it's trying to be used when talking to the cisco...`
`03:57.48`	`*** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net)`
`03:59.21`	`brockj49464`	`cyon: Do you disallow=all then allow what ones you want to use?`
`03:59.31`	`Cyon`	`brockj49464: absolutely`
`03:59.44`	`Cyon`	`It looks like cisco ignores it and tries to establish calls as speex`
`04:00.22`	`Cyon`	`The only one allowed in my sipuras is ulaw, the only one allowed in asterisk is ulaw, and the cisco has "codec g711ulaw" as well...`
`04:00.47`	`Cyon`	`And yet: [2006-01-11 17:52:24] WARNING[32704]: Unable to find a codec translation path from speex to ulaw`
`04:02.30`	`hhoffman`	`is there a better tts then festival to use with asterisk?`
`04:03.55`	`Cyon`	`http://pastebin.com/501950 <-- anyone have any ideas?`
`04:04.01`	`Cyon`	`hhoffman: Not that I've seen`
`04:07.31`	`ObsidianX`	`http://www.voipuser.org/forum_topic_4196.html`
`04:08.28`	`*** join/#asterisk mud (n=mud@206-248-138-115.dsl.teksavvy.com)`
`04:09.08`	`*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)`
`04:10.08`	`fugitivo`	`hhoffman: www.cepstral.com`
`04:11.45`	`Jameno123`	`Cyon, try "disallow=all" "allow=ulaw"`
`04:11.59`	`Jameno123`	`hrm, nobody has any ideas about my issue?`
`04:12.21`	`Jameno123`	`http://pastebin.com/501931`
`04:13.00`	`dily`	`http://www.loquendo.com/regional_preferences.htm`
`04:13.17`	`Cyon`	`Jameno123: Was done long ago`
`04:13.32`	`Cyon`	`Jameno123: speex isn't even a protocol that asterisk has by default`
`04:13.44`	`file`	`s/protocol/codec`
`04:14.18`	`Cyon`	`Jameno123: Something is trying to use it, or makes asterisk think it is; yet cisco doesn't support that codec either it would appear, and my sipura is set to use g711, and pref. codec only.`
`04:14.27`	`Cyon`	`file: Sorry, yes.`
`04:15.24`	`hhoffman`	`fugitivo: thanks checking now`
`04:15.41`	`Jameno123`	`twisted[asteria], wakey wakey!`
`04:16.46`	`hhoffman`	`fugitivo: are these voice compatible with festival?`
`04:17.00`	`Cyon`	`Jameno123: I'm not a coder anymore; but can I see a pastebin of all the verbose/debug lines?`
`04:17.04`	`fugitivo`	`hhoffman: no, it's closed source`
`04:17.07`	`SwK`	`jameno123 is from teh svn or from the 1.2.1 tarball?`
`04:17.18`	`Cyon`	`Jameno123: So I can see which src files it is bouncing through`
`04:17.34`	`SwK`	`it looks like a bad check out from svn`
`04:17.55`	`hhoffman`	`fugitivo: k, thx`
`04:17.57`	`Jameno123`	`SwK, svn, ive deleted and redownloaded twice now.`
`04:18.14`	`SwK`	`it looks like 1/2 and update to me`
`04:18.23`	`hhoffman`	`ah, but I'm guessing it's meant to work with * as they have digium links on their page`
`04:18.25`	`SwK`	`are you running head?`
`04:18.31`	`SwK`	`(or trunk now)`
`04:18.36`	`Jameno123`	`SwK, trunk`
`04:18.45`	`Jameno123`	`ive always ran CVS-HEAD`
`04:18.57`	`Cyon`	`Jameno123: Ah, I assumed it was the tgz download...`
`04:19.07`	`SwK`	`i did to til 1.2.X was released`
`04:19.17`	`SwK`	`1.0 was just to damned old and missing too many features`
`04:19.23`	`*** join/#asterisk santiago (n=santiago@208.195.215.97)`
`04:19.35`	`Qwell`	`I run svn roots`
`04:19.37`	`SwK`	`I would try compiling the 1.2.1 zap sources from the tarball and see what happens`
`04:19.48`	`Qwell`	`more features than trunk`
`04:20.41`	`Jameno123`	`hrm`
`04:20.59`	`Jameno123`	`will try`
`04:21.58`	`*** join/#asterisk Hybrid (n=hybridra@calera-47.cher.brightok.net)`
`04:22.26`	`Jameno123`	`SwK, yea, the "out of date" stuff, is what concerns me ;)`
`04:22.51`	`SwK`	`i wouldnt worry about it rightnow`
`04:24.36`	`*** part/#asterisk santiago (n=santiago@208.195.215.97)`
`04:25.15`	`tainted_`	`how do i do E911 for a client?`
`04:25.32`	`Jameno123`	`SwK, waiting on the box to rebewt, i guess we'll see :)`
`04:25.35`	`SwK`	`very carefully`
`04:25.44`	`SwK`	`tainted_ are you an ITSP?`
`04:25.57`	`tainted_`	`SwK it's for a client`
`04:26.04`	`*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)`
`04:26.15`	`tainted_`	`i don't normally do this kind of stuff`
`04:26.40`	`Cyon`	`Jameno123: Reboot? Why woyld you reboot?`
`04:27.01`	`Cyon`	`s/oy/ou`
`04:28.41`	`Jameno123`	`Cyon, ;) kernel updates`
`04:28.50`	`Cyon`	`Jameno123: Ah, ok. :)`
`04:29.09`	`Jameno123`	`would be so nice to`
`04:29.17`	`Jameno123`	`cat newkernel > /proc/kcore`
`04:29.20`	`Jameno123`	`and not have to reboot ;)`
`04:29.25`	`Jameno123`	`but i dont think we'll see the day`
`04:29.28`	`SkramX`	`heh, it would.`
`04:29.31`	`Cyon`	`I can't wait till we have dynamic kernel loading...`
`04:29.48`	`Cyon`	`Nah, it's doable; just the entire structure would have to be redone, and it'll be years...`
`04:29.55`	`Cyon`	`But it will happen eventually`
`04:30.03`	`Hybrid`	`Anybody have Mechwarrior 3?`
`04:31.42`	`*** part/#asterisk Hybrid (n=hybridra@calera-47.cher.brightok.net)`
`04:32.00`	`Jameno123`	`SwK, suggest using 1.2.1 [.tgz] completely or just zaptel?`
`04:34.40`	`SwK`	`1.2.1 zap shoudl work with trunk at this time,altho i'm not sure... 1.2.1 would probably be better for products as its a known quantity and its not missing much from trunk yet (unless there is something in trunk you really need)`
`04:36.24`	`Jameno123`	`swk it built properly ;) heh, it should run then`
`04:37.06`	`Jameno123`	`hah`
`04:37.07`	`Jameno123`	`yay!`
`04:37.12`	`Jameno123`	`<PROTECTED>`
`04:37.18`	`Jameno123`	`<PROTECTED>`
`04:37.20`	`fugitivo`	`WIRING WIRING WIRING`
`04:37.22`	`Jameno123`	`heh`
`04:37.34`	`hnupik`	`children`
`04:38.02`	`SwK`	`hah`
`04:38.09`	`SwK`	`it always gripes about g729`
`04:38.45`	`*** join/#asterisk qhrisnd (n=qhrisnd@ppp-71-129-177-185.dsl.irvnca.pacbell.net)`
`04:38.51`	`file[laptop]`	`hahaha...`
`04:38.58`	`qhrisnd`	`Good evening everyone :-)`
`04:38.59`	`file[laptop]`	`my cellphone bill is insane`
`04:39.48`	`Jameno123`	`SwK, hrm, should i rm -rf that and re-make install?`
`04:39.52`	`[TK]D-Fender`	`Perhaps its the 800# attached to it :)`
`04:40.04`	`Qwell`	`Jameno123: It's just a warning...ignore it if that was the only file`
`04:40.05`	`file[laptop]`	`wait for it people`
`04:40.13`	`Qwell`	`file[laptop]: $938?`
`04:40.16`	`Qwell`	`CAD`
`04:40.18`	`file[laptop]`	`invoice amount$1,603.26`
`04:40.19`	`ObsidianX`	`how would i go about fixing the error "Inappropriate authentication received" when i try to connect an IAX client to *`
`04:40.21`	`SwK`	`yeah what qwell said`
`04:40.21`	`Qwell`	`jesus`
`04:40.35`	`rob0`	`file[laptop]: have it committed :)`
`04:40.35`	`Qwell`	`file[laptop]: how the hell did you manage that?`
`04:40.39`	`SwK`	`it always gribes about codec_729 cause you dont have the source for it`
`04:40.40`	`*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`04:40.44`	`file[laptop]`	`we'll see.`
`04:40.45`	`rob0`	`I know well how he did it!!`
`04:40.48`	`ManxPowe`	`Just how DOES one get a $1,000 cell phone bill anyway?`
`04:40.51`	`ManxPowe`	`file, GO PREPAY!`
`04:41.02`	`ManxPowe`	`rob0, all those phonesex phone calls?`
`04:41.10`	`SwK`	`is that CDN File?`
`04:41.12`	`file[laptop]`	`I just have the transaction on my account, I don't have the invoice online yet and my balance isn't adjusted yet`
`04:41.13`	`rob0`	`I saw him here, typing in IRC, while on the road`
`04:41.15`	`fugitivo`	`WTF??`
`04:41.17`	`file[laptop]`	`SwK: yes`
`04:41.19`	`file[laptop]`	`rob0: yup`
`04:41.30`	`file[laptop]`	`they probably billed me for data, and backbilled me for past data usage`
`04:41.31`	`fugitivo`	`file[laptop]: $1600?????`
`04:41.33`	`SwK`	`file: oh so its like a normal 100USD phone bill?`
`04:41.41`	`ManxPowe`	`Ah. Mine would be like $5,000 if I wasn't on the flat rate data plan`
`04:41.47`	`file[laptop]`	`I need to calculate how it got to that amount though`
`04:41.48`	`file[laptop]`	`it makes no sense`
`04:41.57`	`Qwell`	`$50/kb?`
`04:41.58`	`Jameno123`	`SwK, yea, it bitched about more, but im not pasting them all :) should i rm -rf the modules dir, and reinstall it all completely?`
`04:42.07`	`Jameno123`	`like 15 files are listed`
`04:42.08`	`Jameno123`	`heh`
`04:42.13`	`h3x`	`damn bid snipers`
`04:42.21`	`xachen`	`Canada data rates are bad for mobile providers`
`04:42.23`	`h3x`	`i accidently pasted a auction item number in where a price goes`
`04:42.26`	`xachen`	`they will coin you easily $1/mbv`
`04:42.29`	`h3x`	`and i bid 5 million on an ATA device`
`04:42.40`	`file[laptop]`	`my regular bill is $60`
`04:42.40`	`SwK`	`jamesno123: probably want to get rid of them but not the g729 one`
`04:42.48`	`SwK`	`you'll need it for g729`
`04:42.55`	`Jameno123`	`SwK, yea, i use g729, i know about it ;)`
`04:43.03`	`file[laptop]`	`so I used 100MB of data apparently`
`04:43.09`	`Jameno123`	`like you said, only because it wasnt compiled directly be the source`
`04:43.16`	`fugitivo`	`file[laptop]: don't pay it, that's insane`
`04:43.22`	`xachen`	`downloading porn onto your blackberry? :D`
`04:43.24`	`file[laptop]`	`fugitivo: I'm waiting for the bill.`
`04:43.26`	`xachen`	`:O rather`
`04:43.26`	`\|omni\|`	`Cingular did that to me a couple months ago but it was only $580 for data`
`04:43.27`	`*** join/#asterisk sumonish (n=God@203.12.249.168)`
`04:43.32`	`sumonish`	`hi all`
`04:43.45`	`\|omni\|`	`I switched to the unlimited data account... a mere $20 more than I was paying already`
`04:43.46`	`\|omni\|`	`bastages`
`04:43.59`	`*** join/#asterisk srt (n=nobody@18.120.9.213.dsl.getacom.de)`
`04:44.16`	`file[laptop]`	`I'm not overly thrilled, but I legitimately used it so if they billed it right... yeah`
`04:44.31`	`file[laptop]`	`life goes on`
`04:44.53`	`file[laptop]`	`so help me god if my mother opens my cellphone bill`
`04:45.04`	`SwK`	`hahaha`
`04:45.10`	`fugitivo`	`heart attack`
`04:45.22`	`sumonish`	`i have an asterisk server which my boss has setup and left me with unfortunatly the CallerID is causeing an issue where when a call comes in it dumps the call i have the following issue in Myphp The $cfg['PmaAbsoluteUri'] directive MUST be set in your configuration file! can someone tell my what it means and how to fix it??`
`04:45.29`	`file[laptop]`	`she's been nosey lately, she opened my credit card statement ahead of me while I was right in front of her`
`04:45.38`	`file[laptop]`	`and my rrsp notice`
`04:45.44`	`SwK`	`rrsp?`
`04:45.51`	`file[laptop]`	`it's like, "uh... I'm 19 here... get out of my finances"`
`04:45.51`	`rob0`	`yikes!`
`04:45.53`	`\|omni\|`	`sumonish: , that's not your issue, that's just a setting in phpMyAdmin`
`04:45.58`	`file[laptop]`	`SwK: registered retirement savings plan`
`04:46.02`	`SwK`	`oh`
`04:46.06`	`\|omni\|`	`you can edit config.inc`
`04:46.12`	`SwK`	`i guess thats cdn for 401k`
`04:46.13`	`rob0`	`file[laptop]: get a PO Box`
`04:46.27`	`\|omni\|`	`and set the full URL to phpMyAdmin (i.e. http://path.to.server/phpMyAdmin) and that message will go away`
`04:46.39`	`file[laptop]`	`rob0: mmm I could`
`04:46.41`	`rob0`	`in USPS they're pretty cheap`
`04:46.41`	`sumonish`	`i edited the zapata.conf`
`04:46.51`	`sumonish`	`to turn of caller id is that right?`
`04:46.55`	`sumonish`	`i seems to work`
`04:46.55`	`rob0`	`I pay $18/year I think`
`04:47.05`	`file[laptop]`	`I believe it's $60 CAD/year here`
`04:47.24`	`sumonish`	`ok omni`
`04:47.36`	`rob0`	`they cost more in cities, mine is in a tiny town`
`04:47.36`	`SwK`	`damn did apple release enuff patches yesterday?`
`04:47.49`	`MikeJ__`	`file, so that's like one regular cell bill a year?`
`04:47.50`	`rob0`	`but Canada is no doubt different`
`04:48.04`	`file[laptop]`	`MikeJ__: more`
`04:48.12`	`file[laptop]`	`my regular cell bill is $60/mth total`
`04:48.50`	`sumonish`	`omni where is config.inc stored?`
`04:49.12`	`Jameno123`	`hrm, alright, seems to work :)`
`04:49.22`	`Jameno123`	`but didnt solve my problem/reason for upgrading`
`04:49.23`	`Jameno123`	`heh`
`04:49.25`	`Jameno123`	`1st File Descriptor: -1`
`04:49.29`	`Jameno123`	`<PROTECTED>`
`04:50.49`	`Jameno123`	`after it bridge's a call, it hangs and gives nothing.`
`04:51.03`	`Jameno123`	`service provider returning no data? or some other weird crapola?`
`04:51.30`	`twisted[asteria]`	`SwK, you sure you don't have that shit?`
`04:51.46`	`bsdfreak`	`heh`
`04:52.04`	`qhrisnd`	`I need help with 2 things: 1) I need to find out how to create in my dial plan, a way to make an extension ring over to another extension when its busy. 2) I would like to know how to (if possible) route calls based upon caller id. Can anyone give me some tips?`
`04:52.27`	`Qwell`	`twisted[asteria]: y0`
`04:52.44`	`Qwell`	`twisted[asteria]: going to ETel?`
`04:53.01`	`twisted[asteria]`	`Qwell, no`
`04:53.03`	`ManxPowe`	`qhrisnd, See "show application dial" and the [macro-stdexten] section of extensions.conf. Also see the Wiki and the Asterisk book.`
`04:53.06`	`ManxPowe`	`~docs`
`04:53.08`	`jbot`	`i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html`
`04:53.19`	`Qwell`	`twisted[asteria]: shame..`
`04:53.26`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`04:53.37`	`fugitivo`	`qhrisnd: and DIALSTATUS`
`04:53.47`	`qhrisnd`	`thank you`
`04:53.49`	`twisted[asteria]`	`Qwell, well, if I had known about it sooner, i might could have`
`04:55.51`	`SwK`	`twisted I am sure`
`04:56.57`	`*** join/#asterisk aless (n=fruribe@pc-100-230-83-200.cm.vtr.net)`
`04:56.58`	`*** join/#asterisk pdugas (n=pdugas@h73.90.40.69.ip.alltel.net)`
`04:57.11`	`aless`	`hi, which ports do i need to forward when using a nat?`
`04:57.18`	`Qwell`	`aless: which channel types?`
`04:58.20`	`inv_Arp`	`aless: any port you want`
`04:58.40`	`aless`	`im connecting two servers with iax`
`04:59.00`	`Qwell`	`aless: 4569`
`04:59.06`	`*** part/#asterisk loud (n=ariel@cypher.punk.net)`
`04:59.27`	`*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)`
`05:00.04`	`aless`	`only that one? arent any other services sending packets?`
`05:00.18`	`ObsidianX`	`netstat -nap`
`05:00.20`	`ObsidianX`	`if you wanna find out`
`05:00.31`	`bsdfreak`	`aaa`
`05:00.38`	`ObsidianX`	`course you'll have to look for asterisk processes :P`
`05:02.36`	`SwK`	`damn it`
`05:03.53`	`Jameno123`	`blah blah blah! damn thing :( argh, why the heck doesnt this thing WORK!!!!!!!!!!!! :(`
`05:04.02`	`Jameno123`	`how can i determine where my problem is :(`
`05:04.31`	`mogorman`	`? Jameno123`
`05:04.32`	`Jameno123`	`i call from my cisco 7960 via sip to asterisk1, asterisk1 dials asterisk2, asterisk2 dials our provider.`
`05:04.35`	`mogorman`	`calm down....`
`05:04.52`	`Jameno123`	`asterisk1->asterisk2 = iax`
`05:04.56`	`Jameno123`	`asterisk2->provider = iax`
`05:05.01`	`mogorman`	`k`
`05:05.08`	`Jameno123`	`if i do "iax2 show channels" on asterisk2, it shows a "UP" bridged channel`
`05:05.22`	`Jameno123`	`yet, i see hear nothing`
`05:05.36`	`mogorman`	`i see hear?`
`05:05.41`	`Jameno123`	`see/hear*`
`05:05.49`	`Jameno123`	`i see no errors, and hear nothing on the phone`
`05:05.56`	`Jameno123`	`if i hang up the phone`
`05:05.57`	`mogorman`	`is jitterbuffer on?`
`05:06.39`	`Jameno123`	`asterisk1 disconnects the call, but asterisk2 still thinks the call is in progress, and doesnt disconnect until it times out.`
`05:06.46`	`*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`05:06.49`	`Jameno123`	`mogorman, which server? all of them?`
`05:07.00`	`mogorman`	`any of them?`
`05:07.07`	`Jameno123`	`this shit just started happening 3 days ago`
`05:07.11`	`Jameno123`	`its been fine "forever" :(`
`05:07.24`	`Jameno123`	`asterisk2=jitterbuffer=no`
`05:07.43`	`Jameno123`	`asterisk1=jitterbuffer=no`
`05:07.48`	`Jameno123`	`i dont know about my service provider`
`05:08.07`	`mogorman`	`hmm it sounds like a bug we have been working on`
`05:08.16`	`Jameno123`	`bug? heh`
`05:08.23`	`Jameno123`	`it just "mysteriously" happens?`
`05:08.46`	`mogorman`	`does this happen if you turn off iax native transfer`
`05:08.47`	`Jameno123`	`heh, just magically started happening one day`
`05:09.09`	`watchy`	`whats the quick reset for a 7960?`
`05:09.40`	`mogorman`	`pull the plug ^_^`
`05:09.42`	`Jameno123`	`watchy, reboot? (*+6+services)`
`05:09.50`	`watchy`	`thanks brother`
`05:09.52`	`Jameno123`	`err settings`
`05:09.59`	`Jameno123`	`* 6 settings`
`05:10.02`	`Jameno123`	`1 of the two`
`05:10.12`	`Qwell`	`real men #`
`05:10.24`	`Jameno123`	`mogorman: hrm.`
`05:10.31`	`Qwell`	`<rant>`
`05:10.31`	`Jameno123`	`i cant say ive ever done that before ;)`
`05:10.42`	`Qwell`	`Why did Cisco do # for the reboot on the sccp 7960?`
`05:10.47`	`Jameno123`	`let me go read some docs, or shed me some light :)`
`05:10.52`	`Qwell`	`You have to be in settings for it to work...`
`05:11.05`	`Jameno123`	`sccp, blah!`
`05:11.07`	`Qwell`	`and...what do you need to press to unlock the phone? That's right... **#`
`05:11.09`	`mogorman`	`id try turning off native transfer first`
`05:11.21`	`Jameno123`	`mogorman, thats what im reading docs to figure out how ;)`
`05:11.29`	`Qwell`	`So, if you want to unlock, and it didn't appear to work the first time...what do you do? You press it again`
`05:11.41`	`Qwell`	`and in doing so...you reboot the damn thing. How stupid...`
`05:11.42`	`Jameno123`	`notransfer=no ?`
`05:11.42`	`Qwell`	`</rant>`
`05:12.04`	`mogorman`	`hmm i think so....`
`05:12.09`	`mogorman`	`id have to look it up sorry`
`05:13.01`	`*** join/#asterisk aless_ (n=fruribe@pc-100-230-83-200.cm.vtr.net)`
`05:13.21`	`Jameno123`	`mogorman, nope, didnt help`
`05:13.42`	`Jameno123`	`i think junction networks is being a pain in my arse again :)`
`05:13.48`	`mogorman`	`did you turn it on all points and check it again`
`05:14.06`	`Jameno123`	`i turned it "off"`
`05:14.10`	`Jameno123`	`it should be "on" ?`
`05:14.22`	`Jameno123`	`i disabled it, on all servers, yet`
`05:14.23`	`Jameno123`	`yes*`
`05:14.38`	`Jameno123`	`err both* well, the two i have access too, not my providers, of course.`
`05:14.51`	`Jameno123`	`i think its just a provider issue :(`
`05:15.03`	`mogorman`	`maybe`
`05:15.05`	`Jameno123`	`ive never had any problems, and if i dial other phones on my asterisk server, i dont have problems.`
`05:15.08`	`Jameno123`	`so if i do`
`05:15.15`	`Jameno123`	`phone1->ast1->ast2->phone2`
`05:15.18`	`Jameno123`	`no problems, ever`
`05:15.29`	`Jameno123`	`phone1->ast1->ast2->provider=problems`
`05:15.35`	`mogorman`	`yeah probably`
`05:24.49`	`Jameno123`	`mogorman, ;) so stressful when you cant figure out why something is happening hehe`
`05:30.18`	`*** join/#asterisk ThaZZa_Work (n=me@203.80.44.200)`
`05:32.21`	`Jameno123`	`whelp thanks guys for your help, ill go chew on the ear of my service provider tomorrow.`
`05:32.25`	`Jameno123`	`cya.`
`05:32.26`	`litage`	`if you have 1,000+ tenants, each with varying #s of extensions, and use SER to handle registrations, is it feasible to put all tenants on each * box?in a phone call [through *], if 1 person is talking and the other person is silent, are they both sending the same amount of voice data?`
`05:33.24`	`mogorman`	`yeah i understand Jameno123`
`05:33.50`	`litage`	`in a phone call [through *], if 1 person is talking and the other person is silent, are they both sending the same amount of voice data?`
`05:34.58`	`Qwell`	`litage: Since * doesn't do VAD, yes`
`05:35.07`	`litage`	`VAD?`
`05:35.13`	`Qwell`	`~vad`
`05:35.14`	`jbot`	`i heard vad is Voice Activity Detection`
`05:35.19`	`litage`	`ah`
`05:36.21`	`litage`	`Qwell: so the type (volume, pitch, etc) of audio/voice doesn't affect the amount of data transferred?`
`05:36.48`	`Qwell`	`afaik, no`
`05:37.18`	`litage`	`interesting`
`05:38.30`	`*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)`
`05:38.32`	`*** join/#asterisk Evanr00d (n=david@ip68-107-162-212.lu.dl.cox.net)`
`05:38.35`	`lo_tech`	`not AM, bro... louder doesnt mean bigger data :)`
`05:39.52`	`Jameno123`	`oh before i go`
`05:39.54`	`Jameno123`	`one more thing :)`
`05:40.26`	`Jameno123`	`WHen a user transfers a call, on a cisco ip phone (SIP), to another extension, why does the phone never receive anymore calls?`
`05:40.30`	`Jameno123`	`asterisk thinks its "busy"`
`05:40.31`	`litage`	`lo_tech: "not AM"?`
`05:40.42`	`Jameno123`	`litage, "its not AM (like radio)`
`05:41.02`	`lo_tech`	`litage: amplitude modulation...`
`05:41.06`	`litage`	`ah`
`05:41.21`	`*** join/#asterisk Mavantix (n=mavantix@69-168-33-232.chvlva.adelphia.net)`
`05:41.42`	`Jameno123`	`anyone have any idea what i possible am doing wrong?`
`05:41.42`	`litage`	`so if 2 people are using sip and g729, will each person's incoming and outgoing data and voice streams be fairly constant?`
`05:42.18`	`Jameno123`	`they used to transfer, and then receive more inbound calls`
`05:42.23`	`Jameno123`	`now the phones are staying busy`
`05:42.38`	`Jameno123`	`probably something todo with "tT" ? or canreinvite or something?`
`05:42.44`	`Mavantix`	`is there anyway to have asterisk IM me incoming call info, log messages, etc?`
`05:43.01`	`lo_tech`	`all things being equal, without silence suppression or vad, yes... the bandwidth used will be equal for both parties, regardless of how loud or the amount of silence for each phone`
`05:43.03`	`ManxPowe`	`Jameno123, sounds like you are using imcominglimit=1 or setgroup, etc`
`05:43.44`	`ManxPowe`	`Jameno123, if so, this is a know issue, see the mailing list archives, there may be a fix or something.`
`05:44.01`	`Jameno123`	`ManxPowe: hrm, they do disable callwaiting, if callwaiting is enabled it rings fine.`
`05:44.16`	`Jameno123`	`as i thought, if you transfer your phone is released from the call?`
`05:44.23`	`Jameno123`	`i didnt think the phone 'bridged' the call.`
`05:44.56`	`Jameno123`	`ManxPowe, incominglimit is undefined in my sip.conf`
`05:45.19`	`Jameno123`	`and setgroup would be a no.`
`05:45.42`	`Jameno123`	`though, i dont specify "canreinvite"`
`05:45.46`	`Jameno123`	`in the sip.conf, so thats probably the issue?`
`05:46.44`	`watchy`	`anyway to set cisco volume in sipdefault?`
`05:52.48`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`05:58.48`	`*** join/#asterisk rvhi (n=rv@66.175.65.89)`
`06:04.14`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`06:06.54`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)`
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`06:13.14`	`littleball`	`hello, i use E1/PRI, asterisk1.2.1. I got the following warning in the console:`
`06:13.15`	`littleball`	`Jan 12 14:00:18 NOTICE[6681]: channel.c:2416 __ast_request_and_dial: Don't know what to do with control frame 15`
`06:13.21`	`littleball`	`what does it mean?`
`06:27.52`	`wunderkin`	`heh holy crap, the * messages log on my one server never has been rotated`
`06:29.10`	`lo_tech`	`not so bad unless you`
`06:29.16`	`lo_tech`	`are verbose, debug`
`06:30.52`	`wunderkin`	`22k lines since sept`
`06:31.16`	`wunderkin`	`verbose is set to 20 but i dont do much with it, just testing`
`06:34.56`	`*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)`
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`06:47.02`	`chat_jokey`	`hello people`
`06:48.10`	`chat_jokey`	`i am currently doing some asterisk sizing .. task is to support 150 incoming TDM lines and 175 outgoing lines .. with approximately 4000 extensions (mostly used only for intercom)`
`06:48.24`	`chat_jokey`	`anyone can suggest me any pointers on the dimensioning of the same ?`
`06:48.43`	`chat_jokey`	`i read up with voip-info.org .. but its kinda not clear ..`
`06:49.03`	`chat_jokey`	`I am averaging about 400 - 500 odd extensions running from one asterisk box ..`
`06:50.34`	`*** join/#asterisk pengyong (n=lala@222.188.137.123)`
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`06:56.05`	`*** part/#asterisk wellng (n=welles@61.150.11.163)`
`06:56.10`	`welles`	`hi all`
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`07:00.24`	`*** join/#asterisk svenna_ (n=svenna@p548D17E9.dip0.t-ipconnect.de)`
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`07:14.49`	`welles`	[help] i try to install mpg123 on centos4 and it hints that :'decode_i586.s:44: Error: suffix or operands invalid for `push' ...' what's wrong? my machine is 64bit machine
`07:25.25`	`litage`	`is H323 or SIP more NAT- and network-friendly?`
`07:25.58`	`*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)`
`07:29.00`	`*** join/#asterisk Lee619 (n=Lee@netblock-66-245-229-224.dslextreme.com)`
`07:30.01`	`Lee619`	`hello`
`07:32.29`	`*** join/#asterisk elephantMan (n=elephant@197.205.103-84.rev.gaoland.net)`
`07:34.49`	`tzafrir_laptop`	`welles, use rawplayer, unless you want to stream music`
`07:36.58`	`welles`	`rawplayer? ok,i have a try .it can replace mpg123 for music on hold on *?`
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`07:52.00`	`koperniqs`	`hi`
`07:52.12`	`Lee619`	`good morning`
`07:56.27`	`infinity1`	`good nite`
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`08:14.16`	`ObsidianX`	`litage: i read that IAX was NAT friendly`
`08:14.28`	`ObsidianX`	`litage: i think it uses UDP`
`08:15.23`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`08:18.27`	`chat_jokey`	`any one can give pointers on clustering asterisk ?`
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`08:23.21`	`bazz`	`i'm trying to get asterick going, i've set up my extentions.conf file (i thought) but when i copy a .call file into the outgoing spool i get __ast_request_and_dial: Don't know what to do with control frame 15 and then attempt_thread: Call failed to go through, reason 3. any ideas?`
`08:23.46`	`*** join/#asterisk ToTo (n=ToTo@host243-91.pool8260.interbusiness.it)`
`08:24.09`	`wellng`	`hi all`
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`08:27.36`	`koperniqs`	`chat_jokey: what kind of clustering?`
`08:30.01`	`chat_jokey`	`like i want to have like 4000 extensions - something like IP Centrex`
`08:30.17`	`*** join/#asterisk Rawplayer (i=kevin@ipc31055d2.oom-killer.org)`
`08:30.41`	`chat_jokey`	`koperniqs: trying to figure out how many extension a Dual XEON - 3.0Ghz, 4GRAM can handle ..`
`08:31.00`	`chat_jokey`	`based on that wanna do some sizing ..`
`08:32.37`	`koperniqs`	`chat_jokey: ther's a tool called sipsak (sipsak.org) that might help`
`08:39.26`	`chat_jokey`	`koperniqs: lemme have a look`
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`08:43.31`	`*** join/#asterisk DHuang (n=DHuang@mail.medec.com.au)`
`08:43.44`	`DHuang`	`Hi`
`08:44.24`	`DHuang`	`Can someone help me with SER + Asterisk?`
`08:49.51`	`*** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com)`
`08:51.37`	`*** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it)`
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`08:53.00`	`DHuang`	`helo?`
`08:56.42`	`Nico_Bdav`	`hi all`
`08:57.09`	`DHuang`	`hi Nico... can you help me with SER + Asterisk?`
`08:57.10`	`chat_jokey`	`hi DHuang even i am looking for similar stuff`
`08:57.21`	`Nico_Bdav`	`does anyone know a good T1->IP gateway, compatible with asterisk ?`
`08:57.39`	`Nico_Bdav`	`DHuang, no sorry`
`08:57.42`	`chat_jokey`	`Nico_Bdav: are you looking for TDM hardware ?`
`08:58.00`	`DHuang`	`chat_jokey: I see... what I'm trying is to make SIP Client to call each other through SER + Asterisk`
`08:58.03`	`chat_jokey`	`Asterisk itself can act as gateway !`
`08:58.09`	`Nico_Bdav`	`chat_jokey, i want to test asterisk on one site`
`08:58.43`	`Nico_Bdav`	`but i want on another site which already have a PBX to convert T1 outlet to IP`
`08:59.16`	`DHuang`	`Chat: kewl.. just tried a config and work now.. :-p`
`08:59.40`	`chat_jokey`	`DHuang: i am trying to scale asterisk, so its suggested that one uses SER as SIP Proxy and enable it to throw SIP calls into Multiple Asterisk boxes, but i dont seem to find anything relevant online ... can anyone else help me on this ?`
`09:00.18`	`DHuang`	`Chat: search fallover I think is on the original setup doc.`
`09:01.23`	`chat_jokey`	`I have A@H here .. hmm`
`09:02.13`	`DHuang`	`Dam... not working.. ;-(`
`09:03.54`	`*** join/#asterisk tengulre11 (n=tengulre@61.185.224.66)`
`09:20.54`	`*** join/#asterisk A-jay (n=quirc@62.217.245.194)`
`09:22.22`	`*** join/#asterisk BriSch (n=BriSch@lachte.GFAI.de)`
`09:24.35`	`iDunno`	`morning`
`09:24.58`	`A-jay`	`hi`
`09:25.00`	`DHuang`	`Chat: does your Asterisk do the registering or the SER?`
`09:25.06`	`DHuang`	`Morning there.`
`09:25.13`	`A-jay`	`hi`
`09:25.54`	`DHuang`	`I'm trying to figure out how to SER and register on Asterisk so it shows the right HOST IP for the client?`
`09:28.54`	`*** join/#asterisk darkskiez (n=darkskie@bb-194-6-115-241.ukonline.co.uk)`
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`09:30.41`	`Curus`	`Is it possible to dump all session variables from extensions.conf?`
`09:31.40`	`Curus`	`I tried with an AGI script, but I can only get one variable at a time, and only if I know the name`
`09:32.52`	`*** join/#asterisk KriS83 (n=KriS@212.202.141.92)`
`09:33.31`	`RoyK`	`er`
`09:33.43`	`RoyK`	`User disconnected from queue %s while waiting their turn`
`09:33.45`	`RoyK`	`wtf????`
`09:33.53`	`RoyK`	`and noone are put into that queue`
`09:35.40`	`*** part/#asterisk DHuang (n=DHuang@mail.medec.com.au)`
`09:42.17`	`RoyK`	`argh. just upgraded to 1.2.x from 1.0 and now support centres are losing calls. after a while phones stop ringing. people still queueing up..`
`09:46.09`	`*** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au)`
`09:46.24`	`thazza`	`Hey all`
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`09:48.05`	`Curus`	`There is no way to display all currently set variables in extensions.conf?`
`09:48.18`	`RoyK`	`seems like there's a fsckup somewhere in device state`
`09:48.31`	`*** join/#asterisk Vyeperman (n=Vye@ip68-7-245-143.sd.sd.cox.net)`
`09:48.44`	`RoyK`	`Curus: iirc it's quite easy to go through all _channel_ vars with an agi script`
`09:55.03`	`Curus`	`How?`
`09:56.14`	`JonR800`	`any way to pass hints between two asterisk servers? I suppose that's a job for SER.`
`09:56.20`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`09:58.38`	`Curus`	`Channel variables don't all get passed to AGI`
`10:01.19`	`*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)`
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`10:03.34`	`RoyK`	`zoa: ping`
`10:06.18`	`zoa`	`pong`
`10:11.20`	`thazza`	`pang`
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`10:17.38`	`riksta`	`if i have a sangoma A101, when i install do i need the PRI or BRI use flags?`
`10:19.01`	`cypromis`	`PRI`
`10:19.44`	`riksta`	`ok ta`
`10:19.54`	`riksta`	`for euroisdn?`
`10:21.02`	`af_`	`how good is snom 320?`
`10:22.17`	`RoyK`	`http://blog.outer-court.com/prejudice/`
`10:22.41`	`*** join/#asterisk A-jay (n=quirc@62.217.245.194)`
`10:22.58`	`Ahrimanes`	`hey denmark is not mentioned, damnit`
`10:24.19`	`koperniqs`	`af_: the display is small and it's relativly expensive`
`10:26.43`	`*** join/#asterisk gvag11 (n=g@ipa124.6.tellas.gr)`
`10:26.48`	`gvag11`	`hi all`
`10:27.13`	`RoyK`	`koperniqs: relativily, yes, unless you mention norway in that sentence`
`10:27.17`	`RoyK`	`er`
`10:27.25`	`gvag11`	`i just moved to Asterisk 1.2.1 and i miss the CUT function, does somebody knows something ?`
`10:27.26`	`RoyK`	`that was a bummer`
`10:27.40`	`RoyK`	`gvag11: read about asterisk variables`
`10:27.52`	`RoyK`	`http://www.voip-info.org/wiki-Asterisk+variables`
`10:28.03`	`RoyK`	`<PROTECTED>`
`10:28.40`	`zoa`	`royk: http://www.asteriskguru.com/tutorials/cut_function.html`
`10:28.47`	`zoa`	`ows, gvag11`
`10:29.23`	`*** join/#asterisk fulgas (n=fulgas@209.8.233.107)`
`10:29.28`	`zoa`	`you need to use SET for it now`
`10:29.43`	`RoyK`	`http://bugs.digium.com/view.php?id=6218`
`10:29.45`	`RoyK`	`:(`
`10:30.55`	`gvag11`	`zoa : ok ... so i use the SET(var=${CUT ... thanks a lot zoa ...`
`10:31.14`	`gvag11`	`royk : thanks ...`
`10:34.21`	`af_`	`mhh what phone is good to use with *?`
`10:34.28`	`af_`	`I used gs but not very satisfied`
`10:35.28`	`*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo)`
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`10:43.08`	`iDunno`	`FFS`
`10:43.12`	`iDunno`	`is it just me...`
`10:43.31`	`iDunno`	`or does it seem entirely insane that you end up in a queuing system when phoning a Telco`
`10:43.39`	`iDunno`	`these people need more staff, ffs.`
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`10:52.13`	`micolous_`	`hey, i'm having some issues using meetme. when i have a caller using the ilbc codec over a iax2 trunk, the sound from them is very jittery, yet they can hear me and other non-ilbc users fine... capturing the output from them, i see that there sound is breaking up... for about 0.02 seconds the sound is fine, then for 0.01 seconds there's no sound... and this goes on and on`
`10:52.33`	`micolous_`	`i'm using the ztdummy kernel module as my timing source`
`10:53.23`	`micolous_`	`i'm wondering if this is something wrong on my end, or a bug. i've tweaked around with the jitterbuffer and that doesn't seem to help; and without the jitterbuffer it's even worse. and it's asterisk 1.2.1`
`10:56.13`	`*** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net)`
`10:57.46`	`h3x`	`its probably because the frame size is different on your codecs`
`10:58.08`	`micolous_`	`yeah, i noticed it doesn't effect ulaw at all`
`10:58.40`	`micolous_`	`but my friend using asterisk@home with meetme doesn't have this issue, and he's using the same codecs and upstream iax providers`
`10:58.56`	`h3x`	`what is he using for zaptel timing`
`10:59.04`	`micolous_`	`the dummy driver`
`10:59.20`	`h3x`	`a@h is prob a different version of asterisk right`
`10:59.29`	`micolous_`	`yeah, i think it might be 1.0`
`11:00.00`	`h3x`	`i seem to remember somebody else having a problem like this with 1.2`
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`11:04.58`	`tzafrir_laptop`	`asterisk@home is basically a sort of asterisk distribution`
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`11:05.12`	`gvag11`	`hi again ...`
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`11:06.28`	`gvag11`	`i changed the extension.conf and instead of CUT(VAR=... i put SET(Var=${CUT(... but still asterisk (1.2.1 stable) complains the CUT is not register ... Any ideas ?`
`11:09.17`	`micolous_`	`tzafrir_laptop: yeah, i remember helping him set it up in september, so it would be running on asterisk 1.0`
`11:15.49`	`gvag11`	`i changed the extension.conf and instead of CUT(VAR=... i put SET(Var=${CUT(... but still asterisk (1.2.1 stable) complains the CUT is not register ... Any ideas ?`
`11:23.11`	`*** join/#asterisk hhoffman_ (n=hhoffman@tor/session/x-059aa9758068410e)`
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`11:27.20`	`gvag11`	`i changed the extension.conf and instead of CUT(VAR=... i put SET(Var=${CUT(... but still asterisk (1.2.1 stable) complains the CUT is not register ... Any ideas ?`
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`11:37.57`	`Reverend`	`OMFG`
`11:38.16`	`Reverend`	`it's the end of the world!`
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`11:39.03`	`Reverend`	`anyone that uses voicepulse or other IAX2 DT provider, have an issue with there service, where after asterisk has been idle for some time, incomming calls no longer ring in?`
`11:40.56`	`*** join/#asterisk aNaSTaCia_geBeri (n=FiaL@85.108.150.190)`
`11:41.26`	`micolous_`	`i had a similar issue with firefly/freshtel (au), Reverend`
`11:42.07`	`micolous_`	`it was rather annoying to setup, but i eventually kept it happy... i used qualify=no`
`11:44.29`	`Reverend`	`micolous_ thank you, i'll try that`
`11:45.36`	`micolous_`	`but another (ugly) workaround is to have asterisk reload on a cron job every 10-15 minutes... i noticed it would come up after a reload or restart.`
`11:47.31`	`Reverend`	`micolous_ yes, i noticed the same. and i did setup a cron job to do a restart every 20 mins, it is ugly`
`11:49.08`	`micolous_`	`well at least a reload doesn't cut off any active calls`
`11:49.32`	`Reverend`	`neither does "restart gracefully"`
`11:50.02`	`Reverend`	`but if there is an active call when the job runs, it will wait until the call is over to restart, however while waiting for the call to end, no one can make outgoing calls`
`11:50.08`	`Reverend`	`and no other calls can come in`
`11:51.53`	`*** join/#asterisk pa (n=Paolo@unaffiliated/pa)`
`11:54.27`	`*** join/#asterisk cfh (n=luca@82.193.23.6)`
`11:57.06`	`micolous_`	`oh... i was always able to make outgoing calls, just the incoming would be an issue in my case`
`11:57.23`	`micolous_`	`other trunks worked`
`11:57.44`	`Reverend`	`yeh, outgoing calls wasn't a problem until i added the restart gracefully cron job`
`11:58.33`	`cfh`	`when i try to leave a messages on the voice mail asterisk say :`
`11:59.05`	`cfh`	`Executing VoiceMail ...`
`11:59.23`	`cfh`	`and Playing 'vm-theperson'`
`11:59.44`	`cfh`	`and then dont wait and hangup`
`12:00.29`	`micolous_`	`does the asterisk user have write access to /var/spool/asterisk/voicemail/?`
`12:01.33`	`cfh`	`yes`
`12:02.16`	`micolous_`	`hmm... the other thing i'm thinking that could be the case is that the disk is full... other than that I'm out of ideas`
`12:03.24`	`micolous_`	`because they were the two main issues that arose when recording various things on asterisk`
`12:03.30`	`micolous_`	`for me`
`12:04.00`	`cfh`	`no the disk is free`
`12:05.21`	`*** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net)`
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`12:06.05`	`micolous_`	`it can't be the sound files, as asterisk will simply not play them and skip ahead if they're not found or they don't have access permissions`
`12:07.26`	`cfh`	`I try to reconfigure the sound`
`12:07.58`	`BoRiS`	`Does anyone know if it is possible to get a toll free number for europe (that works in all of germany) that will allow me to pick up a phone in germany and call out through my toll free number without the persons phone who I am using gets billed for the call? (it costs money to call your neighbor in germany).`
`12:07.59`	`mut`	`i've been having a lot of peers unavailable from qualify`
`12:08.00`	`mut`	`Jan 12 07:06:14 NOTICE[26744]: chan_sip.c:10014 sip_poke_noanswer: Peer '9896853317' is now UNREACHABLE! Last qualify: 31`
`12:08.02`	`mut`	`like that`
`12:08.09`	`mut`	`i can login to their ata right now though`
`12:08.17`	`mut`	`ata says they're registered`
`12:08.25`	`*** join/#asterisk flujan (n=flujan@internet.nube.com.br)`
`12:08.31`	`flujan`	`hi all`
`12:08.37`	`mut`	`why would their qualify packets just dissapear, qualify is set to 3000ms too`
`12:09.17`	`flujan`	`I'm new to asterisk and want to know which is the best hardware to buy and learn before I install it in my entire system.`
`12:09.30`	`micolous_`	`BoRiS: i'm not sure if they cover germany, but sipbroker has DID numbers for many international locations that allow you to call ~200 voip providers for the cost of a local phone call`
`12:09.37`	`mut`	`flujan: to do what?`
`12:09.39`	`flujan`	`could someone point some product to me?`
`12:09.47`	`mut`	`you don't need any hardware but a pc to use asterisk`
`12:10.02`	`flujan`	`mut: and about the cards?`
`12:10.13`	`mut`	`well if you plan on using a PRI`
`12:10.18`	`mut`	`or a phone`
`12:10.23`	`mut`	`or an ata`
`12:10.26`	`flujan`	`yes... we intent do use phone`
`12:10.43`	`mut`	`what is it you want to do`
`12:12.11`	`flujan`	`mut: I want to create a pbx with two points`
`12:12.15`	`micolous_`	`flujan: normally you would go and purchase access through a SIP or IAX-based VoIP provider, who would handle incoming calls for you, and allow you to make calls on the PSTN. i don't own any VoIP hardware at all, I'm using software phones... however I may purchase a Sipura unit in the future (which is simply a small box you plug into the network and your phone and this allows you to use VoIP on any analogue telephone)`
`12:12.29`	`flujan`	`and this points communicating through digital phones`
`12:12.53`	`mut`	`well, you plan on buying new phones too? and trunking out a single pri`
`12:12.53`	`mut`	`?`
`12:13.14`	`flujan`	`buying new phones. Actually we use analog ones. :D`
`12:13.29`	`mut`	`you want to keep doing the ananlog thing?`
`12:13.40`	`BoRiS`	`micolous: The biggest problem I am having is how to remove cost for the caller. If I setup a european toll free number and I have someon calling from a land line in germany. Does it cost them money on a per minute basis for them talking on the phone (It costs money to call even your neighbor)?`
`12:13.45`	`flujan`	`micolous_: we will not use a VOIP provider`
`12:14.02`	`flujan`	`micolous_: we will have our own lines. :) we have a E1 here`
`12:14.06`	`mut`	`you're going to use what to connect to the PSTN?`
`12:14.07`	`mut`	`ah ok`
`12:14.27`	`sulex`	`do as5400/as5300 work fine with * and SIP?`
`12:14.40`	`flujan`	`mut: no, we will migrate to digital phones.`
`12:14.56`	`mut`	`so you'll probly just want to get a te110p card for the pri, and then for the phones just to SIP with a plycom phone`
`12:15.16`	`mut`	`can go lower budget on the phones if you want tho`
`12:15.36`	`micolous_`	`BoRiS: well in the end, connecting calls over the PSTN costs someone money. in australia, for a few months someone setup a toll free incoming number so you could call from any australian phone and get onto voip. but that was changed to a 1300 number (untimed local call, anywhere in the country) due to the abuse it got`
`12:16.13`	`flujan`	`mut: to a initial environment... I want just a simple card to run tests and stuff`
`12:16.24`	`micolous_`	`you're likely to find people who can connect calls from the PSTN to VoIP on a tolled number, but without paying money, you're unlikely to get it toll free.`
`12:16.33`	`mut`	`just get a sip phone of some sort`
`12:16.40`	`flujan`	`after I have at least two points working as ramals inside the company we will expand this .`
`12:16.47`	`mut`	`when i initially setup everything`
`12:17.00`	`mut`	`i first just used 2 xten softphones to play with the dialplan and user setup`
`12:17.05`	`*** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac)`
`12:17.15`	`mut`	`no hardware investment other than the pc, which was vmware`
`12:17.19`	`micolous_`	`yeah`
`12:17.36`	`micolous_`	`xten xlite and sjphone are good, free softphones`
`12:17.41`	`BoRiS`	`micolous_: I dont mind paying the euro toll free number and minutes but I just don't want their telephone provider charging them on a per-minute rate for calling my toll free number.`
`12:17.42`	`mut`	`just got for the softphone test`
`12:18.02`	`BoRiS`	`(on a land line)`
`12:18.03`	`micolous_`	`if you really want hardware, the sipura spa-2000 (now the linksys pap-2) is a nice unit, and costs just over 100$ (australian)`
`12:18.03`	`gvag11`	`i changed the extension.conf and instead of CUT(VAR=... i put SET(Var=${CUT(... but still asterisk (1.2.1 stable) complains the CUT is not register ... Any ideas ?`
`12:18.11`	`mut`	`go*`
`12:18.57`	`flujan`	`mut: I'm afraid it works in softwares tests and not to hardware tests`
`12:18.58`	`flujan`	`well`
`12:19.12`	`flujan`	`mut: i will buy a machine and install asterisk`
`12:19.24`	`flujan`	`then connect it with a Ip in my network ...`
`12:19.35`	`mut`	`if software works`
`12:19.38`	`mut`	`hardware works too`
`12:19.49`	`flujan`	`after that I get other two machines and start to talk...`
`12:19.52`	`micolous_`	`BoRiS: ah... I think a toll free number in germany would be free for callers in germany, but probably not other people in europe. however I can't confirm this having no real knowledge of how the EU phone systems work and having never lived there. but i would think you need one toll free number for each country you want to handle callers from`
`12:19.53`	`flujan`	`is it that simple?`
`12:19.59`	`mut`	`yea`
`12:20.04`	`flujan`	`mut: cool`
`12:20.07`	`flujan`	`thanks in advance`
`12:20.08`	`flujan`	`:D`
`12:20.18`	`flujan`	`I will provide this right now...`
`12:20.24`	`flujan`	`See you guys.`
`12:20.25`	`flujan`	`:D`
`12:20.27`	`mut`	`adios`
`12:20.41`	`zoa`	`gvag11: can you paste the exact error message ?`
`12:21.21`	`*** part/#asterisk flujan (n=flujan@internet.nube.com.br)`
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`12:23.34`	`benjk`	`micolous_ you can get an international toll free number (country code 800)`
`12:24.00`	`benjk`	`rare and probably expensive (though I don't really know) but they do exist`
`12:24.11`	`gvag11`	`zoa i am afraid that not now cause i am reinstalling asterisk ... But it was like "... CUT not register" and with "show functions i can't see that...`
`12:24.19`	`*** part/#asterisk da_didi (n=didi@wikipedia/MichaelDiederich)`
`12:25.36`	`micolous_`	`benjk: i didn't know about those... but yeah, they would cost a bucketload`
`12:26.03`	`micolous_`	`probably cheaper to have a local toll-free number in each country your company services`
`12:26.37`	`benjk`	`airlines often have those international 800 numbers`
`12:27.58`	`micolous_`	`well, their clients often move around between countries, so such an expense is justifable`
`12:28.41`	`Ikarus`	`If it is just for europe in the European telephone numberspace there is a toll-free catagory`
`12:30.10`	`*** part/#asterisk cfh (n=luca@82.193.23.6)`
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`12:31.12`	`Reverend`	`anyone recommend a toll-free service that's better than Kall8 ?`
`12:31.15`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`12:31.22`	`Reverend`	`erm... not 'better' but cheaper?`
`12:31.26`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
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`12:34.41`	`chiardon`	`helllloooo`
`12:36.07`	`gvag11`	`zoa : after uninstall (rm) and install everything fine ... thanks`
`12:36.10`	`gvag11`	`bye`
`12:42.08`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
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`12:44.05`	`synthetiq`	`im runnign asterisk on freebsd.....but port 5060 wont open... who knows is 4569 is...any idea why?`
`12:48.58`	`*** join/#asterisk diego_br (n=diego@200.208.241.178)`
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`12:53.24`	`*** join/#asterisk zapotecz (n=surfer@217.201.198.236)`
`12:53.34`	`zapotecz`	`good morning`
`12:53.45`	`zapotecz`	`some can help me with a mess ?`
`12:54.01`	`zapotecz`	`I'm tring from one week to do this extension`
`12:54.11`	`zapotecz`	`but I really don't know what do for solve :(`
`12:54.20`	`Reverend`	`what's not working right?`
`12:54.25`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)`
`12:54.25`	`chiardon`	`no d-channel available. Using primary channel as d-channel anyway . . .some ideas about what happen here?TIA`
`12:54.35`	`zapotecz`	`i've to dial "69phonenumber#"`
`12:54.48`	`zapotecz`	`from a PRI zapata`
`12:54.59`	`zapotecz`	`but asterisk take the # as "end of call"`
`12:55.09`	`zapotecz`	`and doesn't call my message box`
`12:55.55`	`zapotecz`	`I really don't know how to solve this :(`
`12:55.56`	`*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)`
`12:56.07`	`zapotecz`	`i've tried with a disa , and put the number from another phone`
`12:56.19`	`zapotecz`	`i've tried trough sip`
`12:56.19`	`zapotecz`	`but nothing :(`
`12:58.42`	`*** join/#asterisk RoyK (n=roy@host-81-191-145-46.bluecom.no)`
`12:59.18`	`zapotecz`	`i've tried with google`
`12:59.20`	`zapotecz`	`but no answer`
`13:00.24`	`*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)`
`13:00.25`	`*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)`
`13:03.17`	`mut`	`whats your dialstring?`
`13:03.36`	`zoa`	`zapotecz: did you try features.conf ?`
`13:03.56`	`zapotecz`	`i've to dial "69003905523552#"`
`13:04.21`	`mut`	`ya but thats not what your dialplan says..`
`13:05.39`	`zapotecz`	`mhhh`
`13:05.46`	`zapotecz`	`features.conf only work in local`
`13:05.58`	`zoa`	`that i dont know`
`13:06.01`	`zoa`	`i never used it`
`13:06.18`	`zapotecz`	`I do that`
`13:06.45`	`zapotecz`	`exten=> 555,1,dial(zap/g1/69003905523552#"`
`13:06.56`	`zapotecz`	`with drive syntax`
`13:07.24`	`zapotecz`	`and I receive a "no one avaiable to answer"`
`13:07.43`	`zapotecz`	`i've tryed also with a "normal" pabx and the dialstring work`
`13:08.11`	`zapotecz`	`I suppose that asterisk recognize the final pound/hash as "stop dialstring buffering"`
`13:08.22`	`zapotecz`	`or in /etc/asterisk/zaptel.conf`
`13:08.37`	`zapotecz`	`in the format number`
`13:09.03`	`mut`	`um`
`13:09.34`	`*** join/#asterisk coppice (n=chatzill@166.168.17.210.dyn.pacific.net.hk)`
`13:09.37`	`mut`	`so why do you have * dialing a zap chan with that number if you want to access voicemail..`
`13:10.00`	`zapotecz`	`mhh but is a voicemail gived by the carrier`
`13:10.07`	`zapotecz`	`not the asterisk voicemail`
`13:10.18`	`zapotecz`	`is the phone provider that give this service`
`13:10.35`	`mut`	`ah`
`13:10.39`	`zapotecz`	`and all the "internal users" have this voicemail memo`
`13:10.46`	`zapotecz`	`is a big trouble for me :(`
`13:11.16`	`zapotecz`	`but i've really no idea how to bypass this`
`13:12.09`	`flujan`	`hi all`
`13:12.24`	`flujan`	`I asked some time ago about cheap hardware to test asterisk`
`13:12.24`	`flujan`	`:D`
`13:12.32`	`BoRiS`	`hi coppice`
`13:12.34`	`flujan`	`now I return with the same question.`
`13:12.46`	`flujan`	`mi boss REALLY WANT HARDWARE...`
`13:12.50`	`coppice`	`hi`
`13:13.12`	`flujan`	`I already said that we only need the computer and the softphone`
`13:13.24`	`flujan`	`and he still want to see hardware stuff`
`13:13.30`	`flujan`	`so, here I am.`
`13:13.31`	`mut`	`heh`
`13:13.32`	`mut`	`get an ata`
`13:13.40`	`mut`	`sipura 1001`
`13:13.41`	`flujan`	`mut: hi... me again!`
`13:13.42`	`flujan`	`:D`
`13:13.49`	`mut`	`they are like $60`
`13:13.55`	`mut`	`usd`
`13:14.17`	`mut`	`zapotecz: what happens when ya dial that then? instant hangup? or do ya hear anything?`
`13:14.30`	`mut`	`and can ya set verbose 5 and show me the output when ya call it`
`13:14.36`	`zapotecz`	`yes`
`13:14.41`	`zapotecz`	`instant hangup`
`13:14.49`	`zapotecz`	`and the answer`
`13:14.54`	`zapotecz`	`"no one avaiable"`
`13:15.00`	`flujan`	`mut: which ata`
`13:15.20`	`mut`	`sipura 1001`
`13:15.29`	`mut`	`zapotecz: can ya get me that debug output?`
`13:15.31`	`mut`	`www.pastebin.ca`
`13:15.33`	`mut`	`paste in there`
`13:15.59`	`synthetiq`	`im runnign asterisk on freebsd.....but port 5060 wont open... who knows is 4569 is...any idea why?`
`13:16.02`	`mut`	`flujan: http://www.voipsupply.com/product_info.php?products_id=320`
`13:16.04`	`gambolputty`	`Is call duration stored in a variable?`
`13:16.33`	`[TK]D-Fender`	`flujan: SPA-2002 $70 = 2 FXS ports. Describe your setup : # lines (what kind), # of phones (how many need speakerphone, expected usage, etc)`
`13:17.05`	`mut`	`i think his boss just wants to see some hardware phones working over voip`
`13:17.14`	`mut`	`then they'll go for the good stuff`
`13:17.19`	`mcquaid`	`hello, i was trying to get asterisk to work with my voip provider (vbuzzer.com). so far i can make outgoing calls to pstn lines via voip but incoming calls have no audio in either direction`
`13:17.26`	`mcquaid`	`i can see the rtp traffic, but hear nothing. I am running asterisk behind a firewall which i don't have access to.`
`13:17.42`	`*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)`
`13:17.46`	`[TK]D-Fender`	`:/`
`13:17.50`	`mut`	`mcquaid: call them?`
`13:17.55`	`mcquaid`	`however, i can get the sip clients to directly connect to my voip provider and make/receive calls with full audio`
`13:17.58`	`mcquaid`	`call who?`
`13:18.04`	`mut`	`vbuzzer`
`13:18.10`	`mcquaid`	`uh why?`
`13:18.11`	`*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)`
`13:18.19`	`mut`	`cause it's your service provider`
`13:18.29`	`mcquaid`	`their service works fine as i just mentioned in other sip clients (linphone, twinklephone etc)`
`13:18.59`	`mcquaid`	`but the way i got them to work is not by using nat or stun, but by using outboundproxy`
`13:19.03`	`mcquaid`	`otherwise they don't work either`
`13:19.22`	`mcquaid`	`asterisk seems to support outboundproxy but the documentation is pretty thin on this`
`13:19.54`	`mcquaid`	`it was in chan_sip2 last year, and most things have got promoted to chan_sip, and i see outboundproxy in the c code`
`13:20.12`	`mcquaid`	`but using them in my sip.conf has no effect`
`13:20.13`	`*** join/#asterisk nfi\|ermes (n=ermsewrk@217.220.121.62)`
`13:20.22`	`flujan`	`mut: we want internal communication using ramals... this will be the first test`
`13:20.30`	`mcquaid`	`mut, why would you assume it's my provider?`
`13:20.35`	`mut`	`ramals?`
`13:20.41`	`flujan`	`sorry`
`13:20.44`	`mut`	`mcquaid: so i don't have to help ya ;)`
`13:20.47`	`mcquaid`	`heh`
`13:20.48`	`*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)`
`13:21.07`	`mut`	`bbmin gotta go start a pot of coffee`
`13:21.34`	`mcquaid`	`i've read a little about siproxd, is any one familar with siproxd? curious if it would help in this situation`
`13:23.18`	`flujan`	`extensions lines`
`13:23.32`	`flujan`	`i dunno the english work for this`
`13:23.35`	`flujan`	`strands maybe`
`13:23.38`	`flujan`	`:P`
`13:29.30`	`*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)`
`13:30.55`	`*** join/#asterisk Lathos42 (n=Lathos42@adsl-69-210-24-249.dsl.lgnnmi.ameritech.net)`
`13:31.32`	`Lathos42`	`Good morning`
`13:32.13`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`13:34.35`	`*** join/#asterisk Rawplayer (i=kevin@ipc31055d2.oom-killer.org)`
`13:35.09`	`mut`	`flujan: what language is ramals?`
`13:37.07`	`*** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com)`
`13:39.19`	`flujan`	`mut: brazilian portuguese. :)`
`13:40.03`	`mut`	`hm`
`13:40.05`	`mut`	`ah well`
`13:40.17`	`mut`	`babelfish can only translate it to AS in dutch`
`13:40.22`	`mut`	`other than that it`
`13:40.25`	`mut`	`doesn't translate`
`13:40.35`	`flujan`	`hold on`
`13:40.54`	`mcquaid`	`mut, do you have any suggestions on my issue?`
`13:40.59`	`*** part/#asterisk micolous_ (n=michael@ppp251-29.static.internode.on.net)`
`13:41.01`	`mut`	`you're going to want to get some polycom phones if you want to test out a real world thing`
`13:41.27`	`mut`	`http://www.voipsupply.com/product_info.php?products_id=757`
`13:41.32`	`mut`	`somethin like these guys`
`13:43.00`	`tdonahue`	`good morning all`
`13:43.30`	`warthawg`	`voicemail doesn't seem to like my password`
`13:43.40`	`mut`	`you don;t use any options in the dial string do ya?`
`13:43.45`	`*** join/#asterisk nvrs (n=RUR@65.93.97.70)`
`13:43.48`	`tdonahue`	`does anyone use 1.2 on freebsd? we are having issues getting it to bind to port 5060 for sip`
`13:43.49`	`[TK]D-Fender`	`Considerably cheaper source for Polycom phones - http://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-36931336704.htm`
`13:43.53`	`mut`	`and i asked for a verbose output of the dial`
`13:44.36`	`*** join/#asterisk nvrs (n=RUR@65.93.97.70)`
`13:44.40`	`mut`	`they're the same price..`
`13:45.06`	`*** join/#asterisk mistral (i=mistral@jstevenson.plus.com)`
`13:45.30`	`[TK]D-Fender`	`mut... look closer. The Atacomm one is $113. Its the same price when you get the PoE adapter INCLUDED.`
`13:45.51`	`hackeron`	`hey, I have a strange problem, all phones are getting "invalid password" when the correct password is dialed for both meetme and voicemail - any ideas?`
`13:46.20`	`*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)`
`13:46.22`	`*** join/#asterisk devoider (n=racal@gw.01063telecom.de)`
`13:46.28`	`devoider`	`hi fellas`
`13:46.35`	`*** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it)`
`13:47.04`	`Mimmus`	`to try most recent zaptel/pri, what CVS do I need to checkout?`
`13:47.11`	`fugitivo`	`atacomm doesn't accept credit cards??`
`13:47.21`	`fugitivo`	`oh yes`
`13:48.11`	`mut`	`what ever happened to atacomm`
`13:48.11`	`trixter`	`svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2`
`13:48.11`	`trixter`	`svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2`
`13:48.14`	`[TK]D-Fender`	`hackeron : Pastebin your voicemail.conf context, and the extensions.conf entexs that call it.`
`13:48.17`	`mut`	`or is that you fender?`
`13:48.17`	`trixter`	`that should be the most current SVN version`
`13:48.20`	`Mimmus`	`ok, trixter, tahnk you`
`13:48.30`	`[TK]D-Fender`	`is what me?`
`13:48.31`	`flujan`	`mut: sorry, my boss was here`
`13:48.43`	`mut`	`heh`
`13:48.43`	`flujan`	`mut: well, actually we have 3 E1 channels`
`13:48.45`	`_Sam--`	`<PROTECTED>`
`13:48.47`	`_Sam--`	`er`
`13:48.51`	`mut`	`wow big company?`
`13:48.58`	`devoider`	`did someone ever experience missing field values while writing CDR-data? like an "empty" lastapp or dst field`
`13:48.58`	`Mimmus`	`if I have Sangoma, do I need to run wanpipe config before compiling CVS?`
`13:49.06`	`flujan`	`mut: and 140 internal telephones ( aka ramals :P )`
`13:49.08`	`warthawg`	`it looks to me like asterisk can understand my bt-101 fine for everything except voicemail, the console shows password entered is ''`
`13:49.26`	`sivana`	`Mimmus: you should read their docs, but I think you need to compile zaptel first`
`13:49.31`	`hackeron`	`[TK]D-Fender: it happens for meetme too, isnt extensions.conf probably to blame? -- http://rafb.net/paste/results/gfU2eZ43.html`
`13:49.34`	`sivana`	`then recompile it after you run the wanpipe config`
`13:49.35`	`mut`	`and it's all analog right now?`
`13:49.36`	`mut`	`man`
`13:49.37`	`flujan`	`and we want the the less expensive solution to use Asterisk`
`13:49.39`	`mut`	`that SUCKS`
`13:49.47`	`Mimmus`	`sivana: wanpipe driver install patches zaptel`
`13:49.50`	`flujan`	`yes.`
`13:49.53`	`flujan`	`it's all analog`
`13:49.54`	`flujan`	`:(`
`13:50.04`	`*** join/#asterisk RoyK (n=roy@host-81-191-145-46.bluecom.no)`
`13:50.11`	`flujan`	`we want digital and we want the less expensive solution`
`13:50.20`	`mut`	`get those polycom poe phone`
`13:50.33`	`[TK]D-Fender`	`Mimmus : You need to compile zaptel first, then wanpipe, then zaptel AGAIN.`
`13:50.35`	`flujan`	`my boss wants me to try firts change the internal communication`
`13:50.49`	`Mimmus`	`[TK]D-Fender: ah, ok, I remember now... thanks`
`13:50.52`	`flujan`	`and later on test using the E1 channels`
`13:51.01`	`flujan`	`only then we will migrate the entire system...`
`13:51.09`	`[TK]D-Fender`	`hackeron : I need to see the extensions.conf part that calls it...`
`13:51.16`	`sivana`	`Mimmus: isn't that what I just said? :)`
`13:51.20`	`flujan`	`mut: So, I am here asking for help. :D`
`13:51.43`	`Mimmus`	`sivana: yes yes, thank you again`
`13:51.44`	`flujan`	`mut we want first make two internal phones communicate throught asterisk`
`13:51.51`	`*** join/#asterisk amir (n=amir@gentoo/developer/amir)`
`13:52.04`	`warthawg`	`does anyone have voicemail working on openwrt?`
`13:52.07`	`sivana`	`Mimmus: after you have zaptel/wanpipe installed, then do *`
`13:52.14`	`hackeron`	`[TK]D-Fender: http://rafb.net/paste/results/dNQEYT25.html < its the one you gave me, but I tried with VoicemailMain() too where it would also reject the password`
`13:52.17`	`sivana`	`or libpri if you need it`
`13:52.37`	`flujan`	`mut: then making call using the throught the E1 channels to the world. :P`
`13:52.39`	`Mimmus`	`sivana: do I need to recompile 'full' asterisk to try current CVS for zaptel/libpri?`
`13:52.48`	`flujan`	`mut: what did you suggest?`
`13:52.58`	`mut`	`flujan: get those polycom poe phones`
`13:53.01`	`sivana`	`Mimmus: not sure I understand`
`13:53.15`	`mut`	`i can't believe ya use 3 e1's for 140 phones tho`
`13:53.19`	`sivana`	`Mimmus: you should have the same version of zaptel, libpri, asterisk`
`13:53.24`	`mut`	`telemarketing company or something`
`13:53.39`	`Mimmus`	`sivana: I'm having problems with answer detection and I'd like to try current CVS of zaptel/libpri to solve the issue`
`13:53.52`	`Mimmus`	`sivana: I have Asterisk 1.2.1`
`13:54.00`	`sivana`	`Mimmus: then you should stay with the same version for all`
`13:54.08`	`mut`	`mcquaid????`
`13:54.17`	`flujan`	`mut: thanks`
`13:54.17`	`mcquaid`	`yes???`
`13:54.20`	`Mimmus`	`sivana: well, I understand`
`13:54.22`	`[TK]D-Fender`	`hackeron : Heres the problem : exten => *98,2,VoicemailMain(${CALLERID(number)$}@default) its the extra $ before }`
`13:54.33`	`flujan`	`mut: http://www.voip-info.org/wiki-Polycom+Phones`
`13:54.42`	`flujan`	`mut: is that correct?`
`13:54.45`	`sivana`	`Mimmus: if you want to do CVS zaptel/libpri and 1.2.1 asterisk, you run the risk of problems of new functions`
`13:54.55`	`sivana`	`or changed code`
`13:55.08`	`mut`	`http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-48102218496.htm`
`13:55.19`	`Mimmus`	`sivana: ok thanks, I didn't know it, I thought that zaptle/libpri was only 'drivers'`
`13:55.33`	`sivana`	`Mimmus: they are, but they work together`
`13:56.04`	`hackeron`	`[TK]D-Fender: what do you? - that looks like what I have in the pastebin`
`13:56.40`	`mut`	`whats ya company do flujan?`
`13:56.59`	`hackeron`	`[TK]D-Fender: oh, I get it, I removed the $ -- but it still saying login incorrect`
`13:57.00`	`Mimmus`	`sivana: does I need "TDMV DCHAN Native HDLC Support" in Sangoma conf?`
`13:57.02`	`mcquaid`	`mut, were you posting something to me that I missed?`
`13:57.05`	`[TK]D-Fender`	`hackeron : you need to remove the extra $. heres the corrected version : exten => *98,2,VoicemailMain(${CALLERID(number)}@default)`
`13:57.19`	`mut`	`mcquaid: ya.. still asking for that call dump`
`13:57.22`	`sivana`	`Mimmus: probably good idea, do you have a PRI?`
`13:57.26`	`[TK]D-Fender`	`Mimmus : Yes, you want that done in hardware.`
`13:57.39`	`Mimmus`	`sivana: yes, E1 PRI in Italy`
`13:57.43`	`mcquaid`	`ah sorry didn't see that one sec`
`13:57.43`	`sivana`	`ya`
`13:58.16`	`hackeron`	`[TK]D-Fender: still says login incorrect :( - I dial the pin, it then waits for a few seconds, then says incorrect. Do I need to dial # after the pin or something because it just waits no matter what I do and then says login incorrect`
`13:58.30`	`tzanger`	`morning`
`13:58.37`	`devoider`	`i am having trouble with empty values in the generated CDRs, like an empty "dst" field .. or lastapp, this should never happen .. but it does. any similar problems seen?`
`13:58.38`	`flujan`	`mut: it's a call center`
`13:58.56`	`*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)`
`13:58.56`	`tzanger`	`hackeron: turn on debugging and verify that you're seeing the dtmf digits you're pressing`
`13:58.57`	`hackeron`	`[TK]D-Fender: oh, I'm seeing Incorrect password '' for user '7662' (context = default)`
`13:59.07`	`mut`	`i guess that'de make sense then`
`13:59.09`	`hackeron`	`tzanger: I'm not, its getting nothing`
`13:59.10`	`flujan`	`mut: work recruiting candidates to jobs in another companies... ( I hope I made myself clear ... )`
`13:59.25`	`flujan`	`mut: :)`
`13:59.32`	`mut`	`find me a job`
`13:59.33`	`mut`	`i could use one`
`13:59.40`	`tzanger`	`hackeron: using SIP?`
`13:59.44`	`hackeron`	`tzanger: yes`
`13:59.45`	`sivana`	`hackeron: do you have inband or rfc selected?`
`13:59.57`	`tzanger`	`hackeron: using anything but ulaw/alaw?`
`13:59.57`	`flujan`	`mut: for sure... Where are you from? We just work in Brazil. :D`
`14:00.08`	`mut`	`usa heh`
`14:00.15`	`hackeron`	`sivana: on this phone nat=yes, tried on local phones too, didnt work`
`14:00.21`	`sivana`	`hehe`
`14:00.23`	`chiardon`	`Hello`
`14:00.24`	`hackeron`	`sivana: I mean I can dial an exntension and it works fine`
`14:00.28`	`Mimmus`	`sivana: have you idea why my asterisk doesn't detect answer with some rare numbers?`
`14:00.33`	`[TK]D-Fender`	`hackeron : pastebin your phone def as well`
`14:00.37`	`hackeron`	`tzanger: nope, its ulaw`
`14:00.37`	`warthawg`	`hackeron: what kind of phone, our problems sound similar`
`14:00.41`	`mcquaid`	`mut, http://pastebin.ca/36585`
`14:00.44`	`tzanger`	`hackeron: sounds like you're either using a compressed voice codec and inband (doesn't work) or you're expecting inband and the phone's sending rfc2833, or vice-versa`
`14:00.44`	`mut`	`well maybe if ya find me something lucrative enough i'll move to brazil`
`14:00.46`	`flujan`	`mut: sorry... :(`
`14:00.47`	`hackeron`	`warthawg: GXP-2000`
`14:01.01`	`mut`	`i wouldn't mind moving for a few years`
`14:01.01`	`sivana`	`Mimmus: no :)`
`14:01.13`	`mut`	`since i've never even been out of michigan before it'de be cool`
`14:01.17`	`hackeron`	`tzanger: errr, I can make calls fine, to other phones behind NAT, and the echo test works`
`14:01.17`	`warthawg`	`hackeron: i just solved my problem on grandstream`
`14:01.20`	`hackeron`	`tzanger: and its ulaw`
`14:01.22`	`flujan`	`mut: for sure`
`14:01.25`	`hackeron`	`warthawg: how?`
`14:01.30`	`mut`	`mcquaid: and the asterisk debug`
`14:01.39`	`tzanger`	`hackeron: you are not listening`
`14:01.42`	`sivana`	`hackeron: look in your sip.conf, what do you have for dtmf for that user`
`14:01.42`	`flujan`	`mut: I will go to irvine next summer! :)`
`14:01.43`	`chiardon`	`no d channels available.Using primary channel 16 as d channel anyway!What's the issue here?`
`14:01.43`	`warthawg`	`just a sec lstening to messages`
`14:01.49`	`mut`	`i just wanted ya to set verbose 5`
`14:01.50`	`mcquaid`	`sorry how do i generate that?`
`14:01.51`	`tzanger`	`hackeron: making calls and echotest do not need dtmf`
`14:01.57`	`mut`	`and get the dialplan dump`
`14:02.05`	`[TK]D-Fender`	`hackeron : We need to confirm your DTMF mode. just because you can dial does not mean DTMF works while you're IN a call.`
`14:02.08`	`tzanger`	`hackeron: whatever you have selected for DTMF generation, switch it`
`14:02.09`	`hackeron`	`tzanger: oh?`
`14:02.19`	`[TK]D-Fender`	`hackeron : Pastebin your sip.conf`
`14:02.32`	`sivana`	`hehe and slow down and read :)`
`14:02.34`	`Mimmus`	`is there anyone on the earth who is able to debug PRI?`
`14:02.52`	`warthawg`	`hackeron: i went into the grandstream admin console and checked SIP/Info for the DTMF signalling`
`14:02.57`	`hackeron`	`[TK]D-Fender: I dont have dtmf there, let me just try that quickly`
`14:03.05`	`tzanger`	`Mimmus: yep, what's the trouble`
`14:03.10`	`Mimmus`	`Itried also to ask for paid support at Digium but nope`
`14:03.20`	`[TK]D-Fender`	`hackeron : "dtmfmode=rfc2833"`
`14:03.22`	`tzanger`	`Mimmus: I find that very hard to believe`
`14:03.37`	`Mimmus`	`tzanger: my * doesn't detect answer with some (rare) numbers, especially automatic responders`
`14:03.41`	`warthawg`	`now it works, what i dont understand is why it understood extensions and outbound numbers just fine, but not vm password`
`14:04.00`	`chiardon`	`Are the Asterisk cards made with one of this chips?: * HFC USB`
`14:04.00`	`chiardon`	`<PROTECTED>`
`14:04.00`	`chiardon`	`<PROTECTED>`
`14:04.00`	`chiardon`	`<PROTECTED>`
`14:04.04`	`mut`	`man is it more busy than usual this mornin or what`
`14:04.08`	`Mimmus`	`tzanger: it rings indefinitely`
`14:04.36`	`hackeron`	`[TK]D-Fender: tzanger: warthawg: sivana: kick ass, that worked! - but you're saying if we switch to G726 or G729 it wont work anymore?`
`14:05.42`	`tzanger`	`Mimmus: use pri debug to verify that your telco is sending back an answer. many automatic responders are on PRIs themselves and do NOT answer the line to save toll charges (you can do this, you only get one-way audio)`
`14:05.43`	`warthawg`	`hackeron: i am a clueless noobie, i just kept hacking til it worked for me`
`14:05.50`	`sivana`	`hehe`
`14:06.31`	`hackeron`	`warthawg: well, thats what hacking is all about -- going l33t stuff by accident :)`
`14:06.31`	`warthawg`	`hehehe`
`14:06.32`	`tzanger`	`hackeron: you will have DTMF working with any codec if you're using RFC2833. Inband only works with ulaw/alaw`
`14:06.38`	`Mimmus`	`tzanger: I tried to examine pri debug output but it is too difficult for normal people`
`14:06.41`	`[TK]D-Fender`	`hackeron : the voice Codec in this case has nothing to do with how DTMF is passed.`
`14:06.52`	`tzanger`	`Mimmus: just break it down`
`14:07.06`	`warthawg`	`[TK]D-Fender, why does it decode dtmf elsewhere (extensions and phone numbers) but not in vm?`
`14:07.06`	`tzanger`	`what i tend to do is copy/paste it and then turn off line wrapping -- that seems to help`
`14:07.20`	`[TK]D-Fender`	`hackeron : rfc2833 sends the DTMF data outside of teh voice stream and it inserted back in at the ENDPOINT.`
`14:07.29`	`sivana`	`tzanger: what dtmf do I use for wav?`
`14:07.30`	`tzanger`	`warthawg: it's not decoding it. when you dial iwth a sip phone it's not sending dtmf digits as audio, it's sending a text messgae to the * box with the #`
`14:07.42`	`warthawg`	`tzanger ah, thanks`
`14:07.45`	`Mimmus`	`tzanger: I don't understand well the meaning of "break it down".. sorry... my english is bad`
`14:07.48`	`tzanger`	`sivana: seriously, go find a way for me to make piles of money with you rhard work.`
`14:07.58`	`sivana`	`heh`
`14:08.00`	`hackeron`	`[TK]D-Fender: tzanger: what about DTMF via SIP INFO?`
`14:08.05`	`[TK]D-Fender`	`warthawg : because its your PHONE doing the dialing. it doesn't need sound from its own keypad, you just push buttons! Once you get to another device however you need to send IT the data somehow.`
`14:08.06`	`tzanger`	`Mimmus: break it down == study it and try to understand the organization of it`
`14:08.10`	`sivana`	`tzanger: already working on it`
`14:08.22`	`tzanger`	`hackeron: that will work with compressed voice codecs too`
`14:08.34`	`hackeron`	`tzanger: great, thanks!`
`14:08.36`	`warthawg`	`[TK]D-Fender, thanks. who knew telephony was such a black art`
`14:08.40`	`[TK]D-Fender`	`hackeron : SIP INFO can work as well, but use rfc2833 is you can. its a question of what your phone can support.`
`14:08.53`	`[TK]D-Fender`	`warthawg : not that hard really...`
`14:09.10`	`hackeron`	`[TK]D-Fender: hmm, ok I will, thanks`
`14:09.14`	`mcquaid`	`mut, here's my dialplan and sip.conf http://pastebin.ca/36588`
`14:09.19`	`tzanger`	`warthawg: wait until you play with PRI debugging, zapata echo and oddball hangup detection :-)`
`14:09.23`	`warthawg`	`[TK]D-Fender, i've learned more stuff about it in the past 3 days than in my entire life`
`14:09.26`	`mcquaid`	`mut, how do i generate the asterisk debug?`
`14:09.35`	`mut`	`mcquaid`
`14:09.47`	`mut`	`asterisk -r`
`14:09.47`	`Mimmus`	`tzanger: oh, well... there is a sad "!! < Unknown IE 1562 (len = 6)`
`14:09.47`	`mut`	`set verbose 5`
`14:09.47`	`warthawg`	`tzanger not me! :)`
`14:09.47`	`mcquaid`	`oh`
`14:09.47`	`mut`	`then dial the extension`
`14:09.47`	`*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)`
`14:09.51`	`mut`	`and a bunch of crap shows`
`14:09.53`	`tzanger`	`Mimmus: ok, don't worry about that just yet but that is important`
`14:09.54`	`jimbalcomb`	`is using rfc2833 instead of SIPinfo generally considered a better way to go?`
`14:10.30`	`BoRiS`	`grandstream console?`
`14:10.40`	`*** part/#asterisk flujan (n=flujan@internet.nube.com.br)`
`14:11.01`	`mcquaid`	`i've been running asterisk as: asterisk -vvvvc, when i try -r i get:`
`14:11.03`	`Mimmus`	`tzanger: not important? ok, well. And "Progress Description: Inband information or appropriate pattern now available. (8) "`
`14:11.13`	`mcquaid`	`unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?`
`14:11.14`	`*** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net)`
`14:11.24`	`warthawg`	`BoRis: ip address of phone`
`14:11.25`	`mcquaid`	`asterisk is on the same box here`
`14:11.41`	`Reverend`	`mcquaid, asterisk isn't running, or is trying to close, or locked up`
`14:11.41`	`BoRiS`	`mcquaid: You need to start asterisk with safe_asterisk script to use asterisk -r`
`14:12.31`	`warthawg`	`CoolAcid, it is still working`
`14:12.40`	`warthawg`	`sorry, let me restate that`
`14:12.45`	`jimbalcomb`	`BoRiS I don't believe that is exactly correct.`
`14:12.46`	`warthawg`	`coolio, it is still working`
`14:13.08`	`mcquaid`	`ok that worked`
`14:13.11`	`mcquaid`	`doesn't list much though`
`14:13.21`	`jimbalcomb`	`warthawg: whats the scoop on switch the DTMF option?`
`14:13.49`	`devoider`	`i assume no one ever experienced trouble with his/her CDRs missing values ?!`
`14:14.11`	`warthawg`	`jimbalcomb, it works with the phone set to either sip/info or rfc2833`
`14:14.43`	`mcquaid`	`mut, http://pastebin.ca/36590`
`14:14.45`	`Cresl1n`	`mimmus: I just responded to your bugnote`
`14:14.57`	`*** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca)`
`14:15.09`	`BoRiS`	`Thats normal`
`14:15.16`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`14:15.24`	`Cresl1n`	`mimmus: it's not a bug`
`14:15.26`	`devoider`	`ne is klar`
`14:15.28`	`mut`	`mcquaid: set verbose 5 then make the call`
`14:15.30`	`BoRiS`	`Thats normal. Try dialing a number or type "sip show channels".`
`14:15.33`	`mut`	`it should output stuff`
`14:15.40`	`Mimmus`	`Cresl1n: I'm seeing... but how is it possible!!!`
`14:15.44`	`mut`	`mcquaid: type in the console 'set verbose 5'`
`14:15.45`	`sivana`	`tzanger: you busy on Sat/Sun?`
`14:15.57`	`[TK]D-Fender`	`jimbalcomb : Both INFO and rfc2833 work out of band and I guess rate the same. Its a question of picking on your phone supports.`
`14:15.58`	`jimbalcomb`	`warthawg: was there something that led you to switching?`
`14:16.04`	`Cresl1n`	`Mimmus: it's pretty simple, some endpoints don't send CONNECT until really late into the call`
`14:16.20`	`warthawg`	`jimbalcomb: it didn't work in the default setting`
`14:16.32`	`Mimmus`	`Cresl1n: in fact, it is a toll-free number of my telco. And is there no workaround?`
`14:16.32`	`*** join/#asterisk javar (n=javar@Dynamic-IP-cr20011863171.cable.net.co)`
`14:16.35`	`[TK]D-Fender`	`jimbalcomb : Sipura devices use INFO, so thats what I pick for them. Most devices use rfc2833. Cheap junk uses inband :)`
`14:16.42`	`jimbalcomb`	`wathawg: ok, gotcha.`
`14:16.47`	`Cresl1n`	`mimmus: some companies (i.e. fedex) let you navigate their entire IVR before they send a connect`
`14:16.57`	`Cresl1n`	`mimmus: nope, nothing to get around it`
`14:17.05`	`warthawg`	`jimbalcomb, it started out set to in-audio`
`14:17.16`	`Mimmus`	`Cresl1n: but phone rings, I don't hear IVR`
`14:17.26`	`javar`	`somebody know, how insert this line, exten => s,n,Set(TIMEOUT(digit)=5) , on a table for ARA`
`14:17.31`	`jimbalcomb`	`[TK]D-Fender: ok, i am taking over an Asterisk admin position and am having trouble finding information about 'best practices' and the 'why'`
`14:17.38`	`mcquaid`	`ok`
`14:17.40`	`Cresl1n`	`Mimmus: if phone rings, it doesn't mean it's answered`
`14:18.00`	`warthawg`	`jimbalcomb, should be an exciting job :)`
`14:18.03`	`jimbalcomb`	`[TK]D-Fender: is there reason to go with either given the phone supports both SIP and rfc?`
`14:18.10`	`konfuzed`	`jimbalcomb: 'why' what`
`14:18.34`	`cypromis`	`o/w 14`
`14:18.39`	`Mimmus`	`Cresl1n: I will be forced to remove my Asterisk!`
`14:18.49`	`[TK]D-Fender`	`jimbalcomb : SIP is the general protocol, rfc2833 is a FEATURE describing how DTMF will be passed.`
`14:18.52`	`jimbalcomb`	`warthawg: yeah, I'm pretty freaked out. Spent the first two weeks restructure the networking and fixing the busted ass VLAN setup. now im dealing with all day long jitter, echo, and dropped call complaints.`
`14:18.55`	`Cresl1n`	`mimmus: what are you talking about?`
`14:19.08`	`h3x`	`creslin: thats some bullshit`
`14:19.15`	`h3x`	`you dotn have a 2 way audio path to send them DTMFs`
`14:19.18`	`h3x`	`until they supervise`
`14:19.23`	`javar`	`somebody know, how insert this line, exten => s,n,Set(TIMEOUT(digit)=5) , on a table for ARA`
`14:19.26`	`Mimmus`	`Cresl1n: if I have problems like this, surely someone will complain and I will be forced to remove Asterisk!`
`14:19.29`	`h3x`	`so you cant navigate anything until its fully answered`
`14:20.21`	`zoa`	`h3x, whats the problem ?`
`14:20.35`	`h3x`	`Cresl1n mimmus: some companies (i.e. fedex) let you navigate their entire IVR before they send a connect`
`14:20.35`	`h3x`	`Cresl1n mimmus: nope, nothing to get around it`
`14:20.42`	`warthawg`	`ok, as soon as i can figure out how to get the message indicate to light up on the bt-101, i am going to call this a wrap`
`14:20.43`	`jimbalcomb`	`[TK]D-Fender: ok, yeah i think i got just wording my question terribly. i was wondering if there is a reason to send DTMF via rfc2833 or SIPinfo?`
`14:20.45`	`*** join/#asterisk Redfury (n=bharatsa@203.109.101.36)`
`14:20.53`	`zoa`	`h3x: you mean with early media ?`
`14:20.54`	`Redfury`	`hi all`
`14:21.02`	`[TK]D-Fender`	`jimbalcomb : as opposed to inband?`
`14:21.12`	`Redfury`	`I have configured Asterisk using Database,`
`14:21.25`	`Cresl1n`	`mimmus: I don't understand the problem. You say it's ringing, and you're wondering why it's not reported as being answered...`
`14:21.34`	`Redfury`	`and the peers are also picked fromthe db`
`14:21.54`	`Redfury`	`but I am getting a Failure to Query the database warning`
`14:22.15`	`Mimmus`	`Cresl1n: (my english is very bad, sorry) I hear tone of call proceeding normally, remote IVR doesn't answer`
`14:22.18`	`Redfury`	`does any one have any idea as to what must be wrong..?`
`14:22.34`	`BoRiS`	`exten => 1,1,Answer() :-p`
`14:22.56`	`*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)`
`14:22.57`	`Cresl1n`	`mimmus: if you hear the tone of call proceeding normally, and remote IVR doesn't answer, why are you expecting it to be in an answered state?`
`14:22.59`	`jimbalcomb`	`[TK]D-Fender: oh no, i've heard already that inband is sad just wondering which of those two is better, rfc2833 or SIPinfo?`
`14:23.06`	`*** join/#asterisk Defraz (n=t0tal@72.24.26.215)`
`14:23.35`	`Mimmus`	`Cresl1n: because Asterisk doen't detect answer, if I use an analog phone, IVR ansers after first ring`
`14:23.46`	`Reverend`	`mcquaid what kind of info do you want to see?`
`14:24.00`	`sivana`	`Mimmus: how are you connected to PSTN?`
`14:24.06`	`mcquaid`	`mut, http://pastebin.ca/36591`
`14:24.11`	`Mimmus`	`sivana: E1 PRI in Italy`
`14:24.15`	`Cresl1n`	`Mimmus: that basically means you want to use your shiney new PRI as an analog line`
`14:24.31`	`Cresl1n`	`Mimmus: kind of defeats the point of half of what people use PRIs for`
`14:24.31`	`mcquaid`	`Reverend, mut wanted to see asterisk debug when I receive a call from my voip provider`
`14:24.56`	`Mimmus`	`Cresl1n: and what is the correct behaviour?`
`14:24.59`	`Cresl1n`	`Mimmus: if so, that's simple, just do what BorIS said and do an Answer() on your line`
`14:25.11`	`sivana`	`Cresl1n: he's saying that when he uses the PRI, it doesn't detect the remote answer, but when he uses an analog on the same number, it answers`
`14:25.12`	`Cresl1n`	`Mimmus: the correct behavior is how it is behaving`
`14:25.22`	`Cresl1n`	`sivana: that's wrong`
`14:25.31`	`[TK]D-Fender`	`jimbalcomb : Equal. there are multiple forms available because not every device supports either one. Sipura devices don't seem to support rfc2833. Since they use AVT & INFO, I chose INFO for my * side. And things just work. I don't believe ther is a "better" aspect of it`
`14:25.36`	`Cresl1n`	`sivana: that maybe what he's saying, but the problem is wrong`
`14:25.46`	`tzanger`	`h3x: how can you navigate their IVR without them answering? You could receive their audio but you shouldn't be able to send anything (even keypad IEs) I thought`
`14:25.54`	`konfuzed`	`Cresl1n: Mimmus is bummed that he can only get to the IVR when using the analog phone. When using other phones the IVR never picks up`
`14:26.03`	`Mimmus`	`Cresl1n: but it doesn't work! I don't understand :(`
`14:26.17`	`sivana`	`Mimmus: re-explain the problem`
`14:26.35`	`Cresl1n`	`Mimmus: you're going to have to start over`
`14:26.43`	`mut`	`mcquaid: you sure thats not your voicemail system hanging up the call?`
`14:26.47`	`Mimmus`	`sivana: my english is really a problem... sorry... konfuzed explained better`
`14:26.55`	`konfuzed`	`Mimmus: also confirm if what I said is right or wrong or partly correct`
`14:27.20`	`sivana`	`but I'm confused with phones then... * isn't a phone`
`14:27.49`	`Mimmus`	`phones connected to *`
`14:27.56`	`Cresl1n`	`Mimmus: so tell me more about what konfuzed said`
`14:27.57`	`mcquaid`	`hmm, don't see how voicemail would be interferring`
`14:28.21`	`Redfury`	`Hey Anybody has answer to my problem in configuring asterisk with the database...`
`14:28.22`	`jimbalcomb`	`[TK]D-Fender: ok, that is exactly my wondering. thanks.`
`14:28.24`	`Mimmus`	`both directly connected VoIP phones and analog phones connected to a legacy PBX downstream`
`14:28.27`	`mcquaid`	`mut, as i shown in my post, i took my local sip phone out of the equation and just tried to have asterisk play monkeys`
`14:28.33`	`mcquaid`	`it says it is but i hear nothing`
`14:29.08`	`mut`	`the phone isn't behind a nat is it?`
`14:29.28`	`Mimmus`	`Cresl1n: I'm calling a toll-free number by my shiny VoIP phone connected to * and it never ansers`
`14:29.29`	`mcquaid`	`yes the phone and the asterisk box are both behind a nat`
`14:29.44`	`mcquaid`	`the sip phone that is`
`14:30.01`	`mut`	`nothing inbetween tho?`
`14:30.08`	`mcquaid`	`but as i mentioned, if i set up the sip phone to directly connect to my voip provider, i can make and receive calls`
`14:30.10`	`mcquaid`	`no`
`14:30.23`	`*** join/#asterisk tony__ (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com)`
`14:30.23`	`Cresl1n`	`so you have a call from (VoIP phone) -> (Asterisk) -> (PRI-to-PSTN)?`
`14:30.24`	`Mimmus`	`Cresl1n: if I use a plain, old analog phone, remote IVR answers after 1 ring`
`14:30.34`	`mcquaid`	`and i don't need to enable stun or nat for them to work, just set up the outbound proxy`
`14:30.38`	`Mimmus`	`Cresl1n: exactly`
`14:31.10`	`sivana`	`Mimmus: you don't get something like -- Zap/21-1 answered SIP/VOC0081-2-2a57 in your * CLI?`
`14:31.13`	`Cresl1n`	`Mimmus: and with (analog phone) -> (Asterisk) -> (PRI-to-PSTN) it works?`
`14:31.51`	`Mimmus`	`Cresl1n: no, I need to use a phone connected to a completely different line (no Asterisk in the path)`
`14:32.10`	`Cresl1n`	`Mimmus: Ah.... that's interesting`
`14:32.16`	`*** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net)`
`14:32.17`	`mcquaid`	`mut, would the debug of a working call (i.e. when i call my landline) help?`
`14:32.28`	`Mimmus`	`sivana: no, I'm getting " -- Zap/13-1 is proceeding passing it to SIP/232-6699"`
`14:32.37`	`Mimmus`	`sivana: and "-- Zap/13-1 is making progress passing it to SIP/232-6699"`
`14:32.44`	`mut`	`it's more than likely some kinda nat problem i'de imagine`
`14:32.55`	`mut`	`couldn't tell ya for sure tho`
`14:33.01`	`Cresl1n`	`Mimmus: this maybe unrelated, but what version of asterisk/libpri are you running?`
`14:33.17`	`Mimmus`	`Cresl1n: Asterisk 1.2.1, now I'm downloading latest CVS`
`14:33.29`	`mcquaid`	`hmm, i'm sure it is, but with outbound proxy in the sip clients on their own, incoming/outgoing work`
`14:33.48`	`mcquaid`	`without nat or stun, so i was hoping if they can do it, asterisk should be able to as well`
`14:34.07`	`mcquaid`	`tried to find documentation on outboundproxy and outboundproxyport but it's thin`
`14:34.28`	`mcquaid`	`only found info on most features being promoted to chan_sip from chan_sip2 last year`
`14:34.56`	`mcquaid`	`i also wondered if this would be a situation where siproxd would help`
`14:35.00`	`*** join/#asterisk skambar (n=keiner@minasmorgul.stuwo-steinweg.de)`
`14:35.30`	`sivana`	`Mimmus: does the asterisk and libpri version the same, right now?`
`14:36.18`	`*** part/#asterisk cfh (n=luca@82.193.23.6)`
`14:36.29`	`Mimmus`	`sivana: until now, I'm using plain Asterisk 1.2.1`
`14:38.52`	`mcquaid`	`mut, i emailed olaf as he worked on outboundproxy, hoping he'd might want to get outbound proxy working as well as it does in sip clients on their own`
`14:39.25`	`mcquaid`	`but haven't heard from him yet, i tried the asterisk-users forum as well`
`14:39.29`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`14:39.29`	`*** mode/#asterisk [+o anthm] by ChanServ`
`14:39.38`	`mcquaid`	`maybe i shoudl send this to the devel list...`
`14:40.15`	`konfuzed`	`ok so mimmus' analog phone is the out side line which works fine calling into his 1800-DID number. But when picking up the VoIP Phone on his LAN, dialing the 1800-DID just keeps ringing. Mimmus, if you just pick up your voip phone and punch in only an extension for another voip phone (plugged in or not plugged in) or dial 0, then does the IVR pickup`
`14:41.14`	`*** join/#asterisk abatista (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)`
`14:41.28`	`Mimmus`	`konfuzed: no no, to call this toll-free IVR I need to bypass Asterisk and use an old phone with a different line`
`14:41.35`	`*** join/#asterisk Katty (n=angela@64.82.232.54)`
`14:41.36`	`*** join/#asterisk razu_ (n=razu@213-35-170-76-dsl.trt.estpak.ee)`
`14:41.39`	`*** part/#asterisk Katty (n=angela@64.82.232.54)`
`14:41.41`	`konfuzed`	`Mimmus: right`
`14:41.43`	`konfuzed`	`so`
`14:41.48`	`*** join/#asterisk Katty (n=angela@64.82.232.54)`
`14:41.52`	`Mimmus`	`konfuzed: problem is in Asterisk not detecting remote answer`
`14:41.58`	`Cresl1n`	`Mimmus: have you tried taking out the 'r' flag in your dial, and see if you hear anything?`
`14:42.06`	`Katty`	`hi lads.`
`14:42.38`	`konfuzed`	`Mimmus: with the voip phone on your LAN can you get the IVR to pickup by calling an extension?`
`14:42.46`	`*** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net)`
`14:43.19`	`Mimmus`	`Cresl1n: no, I will try in a moment`
`14:43.31`	`ariel_`	`hello everyone`
`14:43.35`	`Katty`	`hewwo ariel_`
`14:43.38`	`tzanger`	`hello`
`14:43.47`	`Mimmus`	`konfuzed: IVR is a PSTN number!`
`14:43.59`	`ariel_`	`Katty, hope your day will be great`
`14:44.05`	`Katty`	`ariel_: thanks, yours too :>`
`14:45.13`	`devoider`	`endlich darf auch horst ins netz .. :o`
`14:45.27`	`devoider`	`oh wrogn # ;)`
`14:45.30`	`zoa`	`aaaarghl, im goin crazy here`
`14:45.32`	`Mimmus`	`Cresl1n: I already have 'r', I'm using 'TrwW'`
`14:45.33`	`devoider`	`err wrong`
`14:45.46`	`Cresl1n`	`Mimmus: take out the r`
`14:46.51`	`Mimmus`	`Cresl1n: ok, immediately`
`14:47.11`	`konfuzed`	`Mimmus: [09:16:26] <Mimmus> Cresl1n: in fact, it is a toll-free number of my telco. And is there no workaround? - where did this toll free number come from? is that your DID setup on your asterisk box or what ??`
`14:47.32`	`Mimmus`	`SOLVED!!!!!!!!!!!!!`
`14:47.35`	`*** join/#asterisk jyukes (n=jameshot@pool-138-89-229-250.atc.east.verizon.net)`
`14:47.42`	`konfuzed`	`the removing r it was then`
`14:48.03`	`konfuzed`	`Mimmus: still curious though, whats up with the toll free number`
`14:48.20`	`Mimmus`	`Can I offer a pizza+beer to Cresl1n?`
`14:48.42`	`*** join/#asterisk santoshr (i=1063@203.199.110.93)`
`14:49.10`	`Cresl1n`	`Mimmus: heh, I can never turn down free food :-)`
`14:49.14`	`Katty`	`beer :<`
`14:49.18`	`Katty`	`less beer, more hugs.`
`14:49.21`	`Katty`	`that's my moto.`
`14:49.31`	`jimbalcomb`	`thats gross`
`14:49.31`	`Katty`	`or possibly motto...never can remember.`
`14:49.33`	`Cresl1n`	`Katty: mine too :-)`
`14:49.39`	`Mimmus`	`Cresl1n: but it would be a real italian pizza`
`14:49.52`	`santoshr`	`i want to test dialing a remote sip server.. i found a list of public sip servers . how can one make a call to that`
`14:50.01`	`konfuzed`	`Cresl1n: I cen get you greyhound bus tickets to go pick up your pizza`
`14:50.11`	`konfuzed`	`;^)`
`14:50.17`	`jimbalcomb`	`same day air shipping via UPS global`
`14:50.43`	`jimbalcomb`	`it'd be the best $300 pizza you ever had`
`14:50.53`	`konfuzed`	`Mimmus: still curious though, whats up with the toll free number`
`14:51.31`	`Mimmus`	`konfuzed: what's the meaning of "whats up"?`
`14:51.47`	`Katty`	`Mimmus: it's a basic greeting`
`14:51.49`	`warthawg`	`que tal`
`14:51.51`	`santoshr`	`<< sip:www.foo.com >> wwere a public sip server which says it does not require a registration.. how should i send a call t here`
`14:51.59`	`warthawg`	`hey, vato, que paso`
`14:52.00`	`Mimmus`	`jimbalcomb: if I'm able to call UPS toll-free number now...`
`14:52.02`	`Katty`	`Mimmus: the lazy How Are You, routine.`
`14:52.12`	`konfuzed`	`and a direct inquiry of what is happening with`
`14:52.26`	`Katty`	`personally i find it annoying`
`14:52.30`	`Cresl1n`	`Mimmus: mmm.... I've never had italian pizza`
`14:52.39`	`Cresl1n`	`what's the difference?`
`14:52.47`	`BeHappy_`	`Cresl1n, dont get it in tuscany, if you want an advice :)`
`14:52.58`	`Mimmus`	`konfuzed: clearly toll-free doesnt' answer if you supply a ringtone ('r')`
`14:52.58`	`Cresl1n`	`konfuzed: heh, you're funny`
`14:52.59`	`konfuzed`	`what is up with the toll-free number you mentioned earlier. is it yours or in use some how ? Why was it mentioned`
`14:53.18`	`*** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com)`
`14:53.24`	`santoshr`	`guys.. can some one give me some ideas please.`
`14:53.24`	`Mimmus`	`Cresl1n: pizza was born in Italy (Naples)!`
`14:53.47`	`Mimmus`	`Cresl1n: in USA you eat a surrogate!`
`14:53.51`	`sivana`	`Cresl1n: was the r causing * to ignore the other end?`
`14:53.57`	`konfuzed`	`Mimmus: whos toll free number is it? yours or somebody elses?`
`14:54.08`	`konfuzed`	`is it a did on yout asterisk box`
`14:54.12`	`Mimmus`	`konfuzed: somebody else, my telco`
`14:54.14`	`Cresl1n`	`sivana: basically`
`14:54.15`	`*** join/#asterisk slak- (i=slak@rewted.biz)`
`14:54.16`	`konfuzed`	`s/did/DID/`
`14:54.34`	`slak-`	`hi, how can i tell which codec my sip connection is using`
`14:54.59`	`Cresl1n`	`sivana: The other end's IVR was starting before it sent the CONNECT, and with the r flag, asterisk sends locally generated ringback until the CONNECT message is received`
`14:55.00`	`slak-`	`im having a conference here using MeetMe and would like to make sure that i have enough bandwidth to support 5 partries`
`14:55.13`	`Mimmus`	`sivana: yes`
`14:55.38`	`Cresl1n`	`sivana: ere go... it overrode the audio that the other end was sending`
`14:55.42`	`konfuzed`	`ok good note on the machincations of the r flag`
`14:55.46`	`santoshr`	`how to dial out a public sip server. sip:foo.com`
`14:56.13`	`sivana`	`I see`
`14:56.49`	`konfuzed`	`Mimmus: do you have Local phone numbers as DID for incoming or just PSTn as in incoming phone number?`
`14:56.51`	`Mimmus`	`Cresl1n: very sad... 2 weeks for this...`
`14:57.07`	`Mimmus`	`konfuzed: why this question?`
`14:57.31`	`konfuzed`	`to understand your layout`
`14:57.54`	`Mimmus`	`Cresl1n: just because I lazily cut&paste dialing options`
`14:58.02`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`14:58.11`	`*** part/#asterisk zapotecz (n=surfer@217.201.198.236)`
`14:58.26`	`Cresl1n`	`Mimmus: heh, you know what they say... you spend 90% of the time on 10% of the problems`
`14:58.34`	`konfuzed`	`I have trouble with being left an incomplete picture because it is too KonFuZing`
`14:58.40`	`konfuzed`	`;^)`
`14:59.00`	`Mimmus`	`Cresl1n: beh, now it's time to go. Thank you again to this channel and especially to you, Cresl1n`
`14:59.12`	`Cresl1n`	`Mimmus: no prob, good luck! :-)`
`14:59.24`	`Mimmus`	`we are aplanning to replace two legacy Alcatel PBX (for 200 users in two sites)`
`14:59.26`	`konfuzed`	`the same problem I have with how answered quetions can be like unsolved mysteries even when no longer such a big deal`
`14:59.30`	`Mimmus`	`and I have much to do`
`14:59.31`	`konfuzed`	`kinda like X-Files`
`14:59.37`	`warthawg`	`can anyone tell me how to get message waiting indicator working on grandstream phone?`
`14:59.45`	`slak-`	`how does g726 compare to ulaw?`
`14:59.50`	`slak-`	`whats the bandwidth difference`
`15:00.01`	`Cresl1n`	`slak-: that's totally google'able`
`15:00.11`	`{zombie}`	`warthawg: there's no trick, just make sure you have the appropriate mailbox= statement in your sip.conf`
`15:00.11`	`slak-`	`okay well i guess its totally askable aswell`
`15:00.12`	`trixter`	`asteriskgurus.org has a bandwidth calculator`
`15:00.41`	`{zombie}`	`and make sure you are either putting your mailboxes under the [default] context in voicemail.conf, or specifying the context in your mailbox=`
`15:00.47`	`brad_mssw`	`slak-: http://www.voip-info.org/wiki/view/Bandwidth+consumption`
`15:00.57`	`trixter`	`as far as bandwidth consumed there are variables. sample size, trunking or no, ATM framing or no, pppoe?`
`15:01.11`	`slak-`	`t1`
`15:01.27`	`warthawg`	`{zombie} ok, thanks`
`15:01.29`	`slak-`	`which codec is ulaw...g7xx?`
`15:01.30`	`konfuzed`	`Mimmus: So, do you have any DID's configured`
`15:01.49`	`brad_mssw`	`slak-: g711`
`15:02.00`	`slak-`	`ty`
`15:02.04`	`Mimmus`	`konfuzed: yes`
`15:02.51`	`konfuzed`	`is that just a toll-free DID or local numbers too`
`15:03.34`	`Mimmus`	`konfuzed: noooooo! It's a public number, not mine!`
`15:03.37`	`*** join/#asterisk bweschke-away (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)`
`15:04.29`	`konfuzed`	`i presume you mean the Toll-free vs local numbers and so that would complete the layout picture quite nicely. at least for me anyway`
`15:05.00`	`Mimmus`	`konfuzed: ok, see you tomorrow, thanks`
`15:05.23`	`konfuzed`	`always good to have a complete picture if possibly eh`
`15:05.25`	`konfuzed`	`;^)`
`15:05.26`	`Ahrimanes`	`anyone successfully get leds on snom phones to turn on and off from asterisk?`
`15:05.57`	`malverian[work]`	`Ahrimanes, Yes.`
`15:06.24`	`*** join/#asterisk klictel (n=klictel@207.107.208.137)`
`15:06.33`	`Ahrimanes`	`malverian[work]: hm have a dialplan example for that?`
`15:06.53`	`[TK]D-Fender`	`malverian[work] : how's that scheduler coming along`
`15:07.09`	`[TK]D-Fender`	`Ahrimanes : exten => 1000,hint,SIP/1000`
`15:07.27`	`[TK]D-Fender`	`Ahrimanes : exten => 1000,1,Dial(SIP/1000,20)`
`15:08.11`	`Ahrimanes`	`[TK]D-Fender: well, i have an agi application that adds/removes a phone from a queue and i'd like it to toggle the led light on the button i press to launch the script..`
`15:08.47`	`*** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net)`
`15:08.55`	`Ahrimanes`	`[TK]D-Fender: so i set the button as a destination for 1000 right?`
`15:10.11`	`[TK]D-Fender`	`Umm, that you CAN'T do yet. SIP Presence only works for devicestate, not just anything.`
`15:10.37`	`Katty`	`..hams?`
`15:10.43`	`Katty`	`that does not parse.`
`15:10.44`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)`
`15:11.08`	`[TK]D-Fender`	`http://dictionary.reference.com/search?q=hams #6`
`15:11.25`	`jbalcomb`	`[TK]D-Fender: Wouldn't sending DTMF as SIP INFO rather than RTP (rfc2833) essential be more reliable due to TCP rather than TCP?`
`15:11.31`	`Katty`	`[TK]D-Fender: don't do that.`
`15:11.32`	`*** join/#asterisk kpettit (n=keith@69.15.174.114)`
`15:11.33`	`[TK]D-Fender`	`Typically used for over-presenting onesself`
`15:11.38`	`Katty`	`[TK]D-Fender: just be yourself.`
`15:11.44`	`Katty`	`[TK]D-Fender: like file.`
`15:11.46`	`Ahrimanes`	`[TK]D-Fender: hm not sure it qualifies as presence.. just toggling the led on a snom`
`15:11.59`	`file`	`Asterisk doesn't use TCP for SIP`
`15:12.06`	`[TK]D-Fender`	`jbalcomb : last I checked All of SIP & RTP were UDP....`
`15:12.15`	`jbalcomb`	`file ah`
`15:12.23`	`[TK]D-Fender`	`Ahrimanes : Not sure if there's a way to toggle them with direct header info....`
`15:12.30`	`tzanger`	`haha`
`15:12.48`	`jbalcomb`	`[TK]D-Fender: ah, hrmm.. how i can to that i dont know but i though it did. too many damn web pages with too many guessed at opinions..`
`15:12.54`	`Ahrimanes`	`[TK]D-Fender: well using devstate i have led in button 5 on my snom190 permanently on now.. but cant get it to turn off, hehe`
`15:13.31`	`jbalcomb`	`Ahrimanes perhaps poking it with a hot solder iron?`
`15:13.39`	`[TK]D-Fender`	`Ahrimanes : reboot the phone. Also keep in mind * wipes presences data every time you do "reload" in CLI`
`15:14.17`	`Ahrimanes`	`[TK]D-Fender: i pulled the power on the phone and did reload in cli and led is still on.. persistent bugger`
`15:14.28`	`Ahrimanes`	`jbalcomb: customer probably would not agree with that`
`15:14.37`	`jbalcomb`	`Ahrimanes: do you like that snom phone? if so, which modem and how much $$$?`
`15:14.55`	`jbalcomb`	`Ahrimanes: hrmm.. perhaps. just tell them its a built in incense burner`
`15:15.07`	`Ahrimanes`	`jbalcomb: i rather like it yes.. costs around $150 i guess.. only know the price in danish currency..`
`15:15.20`	`[TK]D-Fender`	`jbalcomb : Think Polycom ;)`
`15:15.58`	`jbalcomb`	`[TK]D-Fender haha.. yeah, we have several sipura, one polycom, and 100+ grandstreams`
`15:16.23`	`jbalcomb`	`[TK]D-Fender i don like the polycom so much yet`
`15:16.40`	`*** join/#asterisk diego_br (n=diego@200.208.241.178)`
`15:17.18`	`[TK]D-Fender`	`jbalcomb : Which model, and what aspects of it?`
`15:18.32`	`jbalcomb`	`[TK]D-Fender not sure on the model. its too quiet. i have heard good things about them though and we do only have one.`
`15:18.53`	`jbalcomb`	`[TK]D-Fender additionally its in the computer room so its not getting used much`
`15:19.26`	`jbalcomb`	`[TK]D-Fender do you like the polycoms? a particular model?`
`15:20.17`	`[TK]D-Fender`	`I'm running an all-Polycom setup (26 x IP600, 1 x IP601). Volumes are fine. Is the the default volume thats a problem or the max being too low?`
`15:20.50`	`[TK]D-Fender`	`How many line keys on yours? 6 little ones = IP60x, 3 big = IP50x, 2 small = IP30x`
`15:22.00`	`fugitivo`	`is any way to have callprogress with sip?`
`15:24.58`	`devoider`	`block?`
`15:25.19`	`devoider`	`dammit ... wrong # once again`
`15:25.25`	`Cresl1n`	`fugitivo: like inband progress?`
`15:25.33`	`devoider`	`ill check back beeing more awake .. maybe tommorow :)`
`15:26.02`	`fugitivo`	`Cresl1n: tone detection, answering machine, fax, busy, congestion, etc`
`15:26.17`	`Cresl1n`	`fugitivo: nope`
`15:26.19`	`*** join/#asterisk SplasPood (i=jwb@206.252.198.100)`
`15:26.23`	`fugitivo`	`no way to do that?`
`15:26.44`	`Cresl1n`	`fugitivo: have you ever used the zap callprogress code?`
`15:26.54`	`fugitivo`	`no, can't use it in my country`
`15:26.55`	`Cresl1n`	`fugitivo: it's not too great`
`15:27.06`	`Cresl1n`	`fugitivo: so in essence, the answer is no`
`15:27.12`	`*** join/#asterisk Defraz (n=t0tal@72.24.26.215)`
`15:27.19`	`fugitivo`	`using some kind of hack with backgrounddetect?`
`15:27.28`	`*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)`
`15:27.31`	`hackeron`	`can someone recommend a VoIP provider for backup that has unlimited data paths (or at least 8) on a pay as you go? (we are using teliax for main)`
`15:27.38`	`Cresl1n`	`fugitivo: with a LOT of hack`
`15:28.47`	`coppice`	`Its really bad when the product has reached 1.2 and can't do simple tone detection :-)`
`15:29.01`	`Cresl1n`	`coppice: hehe`
`15:29.13`	`Cresl1n`	`coppice: good morning ot you too :-)`
`15:29.14`	`fugitivo`	`right`
`15:30.00`	`Cresl1n`	`coppice: after all, it's ONLY simple tone detection, right? :-D`
`15:30.27`	`coppice`	`it is. its pathetic that the software can't go a reasonable job`
`15:30.45`	`Cresl1n`	`coppice: should only take five or ten minutes, just write a quick FFT algorithm, put a little glue in there, and wahlah!`
`15:30.49`	`Cresl1n`	`:-P`
`15:31.07`	`coppice`	`FFT is not the right starting point`
`15:32.33`	`konfuzed`	`coppice: perhaps you can get the code together by the end of the day ;^)`
`15:32.36`	`fugitivo`	`a have a document from a provider describing each tone, who wants to code it? :)`
`15:32.37`	`bkw_`	`Cresl1n, thats a problem... many things in asterisk are done half ass and never gone back over and fixed correctly`
`15:33.08`	`coppice`	`konfuzed: my code is GPL, so it cannot go into *`
`15:33.17`	`Cresl1n`	`bkw_: so we can either troll about it, or we can do something about it.....`
`15:33.44`	`bkw_`	`Cresl1n, I'm not trolling i'm just stating fact`
`15:33.44`	`*** join/#asterisk loick (n=loick@APuteaux-151-1-6-116.w82-120.abo.wanadoo.fr)`
`15:34.29`	`konfuzed`	`coppice: well if you wrote GPL then asterisk could borrow it as free inclusion with *`
`15:34.31`	`mog_work`	`mmmm trolls`
`15:34.34`	`coppice`	`Cres11n: what's the point of doing something, when updates just sit and rot?`
`15:34.38`	`bkw_`	`konfuzed, WRONG`
`15:34.59`	`Cresl1n`	`coppice: yeah, sorry about that. We're working on getting better with that`
`15:35.03`	`fugitivo`	`coppice: did you code unicall?`
`15:35.07`	`mog_work`	`brian why dont you help anthm and stop trolling....`
`15:35.13`	`tzanger`	`bkw_: well not exactly wrong... the GPL version of asterisk could use it without problem. but ABE and any of the commercial licensed versionsof * could not`
`15:35.18`	`*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)`
`15:35.22`	`bkw_`	`it can't be in CVS at all`
`15:35.35`	`mog_work`	`esp as we dont do cvs`
`15:35.36`	`mog_work`	`anymore`
`15:35.38`	`tzanger`	`bkw_: ? why not?`
`15:35.38`	`fugitivo`	`svn`
`15:35.39`	`coppice`	`Cres11n: its not just my stuff. many people complain their stuff ends up rotting. it seems to be the normal thing`
`15:35.45`	`*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj3a.dialup.mindspring.com)`
`15:35.56`	`tzanger`	`GPL does not restrict where or what it is stored with, ONLY distribution`
`15:35.58`	`*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfj3a.dialup.mindspring.com)`
`15:36.02`	`coppice`	`fugitivo: unicall is mine`
`15:36.09`	`mog_work`	`it was 1.2 , look at how many bugs we are going through this month`
`15:36.10`	`fugitivo`	`coppice: nice work :)`
`15:36.13`	`Cresl1n`	`coppice: yeah, we're realizing that and trying to work better to alliviate the problem`
`15:36.13`	`anthm`	`who's that clip clapping on my bridge!`
`15:36.40`	`konfuzed`	`mmmm had a little chat last week or so (forget where) that basically confirmed you can take GPL code and close it as long as the source is 'offered' for free.`
`15:36.45`	`mog_work`	`i was just telling brian to not troll and actually get some work done with you anthm`
`15:36.46`	`Cresl1n`	`coppice: it's a concern that we're becoming more and more aware of`
`15:36.53`	`bkw_`	`konfuzed, WRONG`
`15:36.55`	`konfuzed`	`hm`
`15:37.02`	`tzanger`	`konfuzed: well technically you're not closing it then, are you?`
`15:37.21`	`BoRiS`	`lol`
`15:37.25`	`mog_work`	`yeah there are sketch people out there like that router guy`
`15:37.27`	`konfuzed`	`well then I hope that conversation was here so that I dont have to go and correct some debian programmers or something like that`
`15:37.29`	`mog_work`	`sveasoft or whatever`
`15:37.36`	`bkw_`	`mog_work, anthm and I have done more for the asterisk code base than most people in the community`
`15:37.44`	`mog_work`	`no one is saying you havent`
`15:37.48`	`konfuzed`	`tzanger: well you can sell the binary`
`15:37.53`	`mog_work`	`but you guys arent now, and some of us have work to do`
`15:38.00`	`anthm`	`umm hi`
`15:38.02`	`tzanger`	`bkw_: I don't think anyone is denying you that. You and anthm are very, very good at this stuff`
`15:38.04`	`anthm`	`i have patches in there still`
`15:38.06`	`fugitivo`	`here we go again`
`15:38.07`	`konfuzed`	`as long as "Offering" the code as opposed to including the code`
`15:38.15`	`coppice`	`this is truly amazing. dell normally rip off asians, but their new 30" LCD seems to be cheaper here than in the US :-\`
`15:38.20`	`mog_work`	`i know anthm, i guess my comment was more directed at bkw_`
`15:38.26`	`tzanger`	`konfuzed: yes of course you can sell GPL binaries, but you must make the source available for free to anyone you distribute the binaries to. that's the entire point of the GPL`
`15:38.34`	`anthm`	`well to bring the conversation full circile`
`15:38.46`	`anthm`	`i was waiting for them to close to ever add any more`
`15:38.51`	`anthm`	`and it's been 8 months =D`
`15:38.56`	`Cresl1n`	`O. M. G. here we go again`
`15:39.01`	`anthm`	`s'all i'm sayin'`
`15:39.04`	`fugitivo`	`Cresl1n: :)`
`15:39.09`	`konfuzed`	`so why cant asterisk integrate some GPL pieces and make the code for those mods available`
`15:39.23`	`mog_work`	`hey i got a crazy idea, instead of complaining about old bugs`
`15:39.24`	`bkw_`	`konfuzed, because the code base can't be tainted`
`15:39.26`	`mog_work`	`lets go fix em`
`15:39.34`	`mog_work`	`i mean 242`
`15:39.38`	`mog_work`	`we can work it down`
`15:39.40`	`BoRiS`	`'Ya'll jacked up and sheeeeeeeeet'`
`15:39.40`	`bkw_`	`if the asterisk codebase is tainted with pure GPL code then ABE and dual lic. wouldn't be possible.. along with g729`
`15:39.41`	`coppice`	`I love this policy about "feature requests" If they can eb ignored for a few weeks, they get deleted. great scheme, that one`
`15:39.43`	`tzanger`	`konfuzed: because ABE and the commercially-licensed copies of * cannot have that code in them, because the GPL parts "infect" the closed-source parts since they're linked`
`15:39.47`	`konfuzed`	`hm Id say that depends on perspective (more so from the programmers than mince of course)`
`15:39.49`	`fugitivo`	`mog_work: yeah! callprogress for sip!`
`15:40.05`	`*** join/#asterisk pa (n=Paolo@unaffiliated/pa)`
`15:40.48`	`Cresl1n`	`coppice: I don't know if the "feature request" mechanism on the bug tracker is the right place for a lot of that stuff`
`15:40.51`	`mog_work`	`i thought sip had callprogress?`
`15:40.52`	`zoa`	`coppice, they dont even sell those in belgium yet :(`
`15:41.21`	`coppice`	`zoa: they don't ship until the 16th here`
`15:41.22`	`fugitivo`	`mog_work: really? can i detect an answering machine, busy, congestion, etc?`
`15:41.31`	`zoa`	`its not on the sites`
`15:41.38`	`mog_work`	`busy and congestion work with sip`
`15:41.50`	`mog_work`	`and there is a patch for generic answering machine detection`
`15:42.02`	`Cresl1n`	`coppice: what's to motivate people to work on feature requests anyway... so if nobody want to work on it (i.e. not enough demand) they sit there and rot`
`15:42.18`	`bkw_`	`Cresl1n, just like patches`
`15:42.26`	`*** join/#asterisk rkioko (n=rkioko@196.200.26.42)`
`15:42.29`	`anthm`	`i'm not bitching or anything and in fact i'll even tell you one more time for the record since you guys just said you were trying to fix the problem. The issue lies with the whole idea where you burden the developer by making him guess how you guys want the code to be then sending it back for recoding after the fact instead of just spending 20 min to describe it the way you would like it to be ahead of time`
`15:42.29`	`bkw_`	`patch rot is the biggest killer of new features`
`15:42.34`	`tzanger`	`bkw_: do you not agree that Digium's gotten a LOT better with that in the last 3 months?`
`15:42.45`	`tzanger`	`nobody is denying that it was very bad in the past`
`15:42.46`	`Cresl1n`	`bkw_: hey man, we're trying to get better at that`
`15:42.52`	`coppice`	`tzanger: no. it has got worse`
`15:43.08`	`tzanger`	`however Digium's taken steps to improve that. If you can't at least admit that it's moving better (not perfect yet of course) then you're a lost cause.`
`15:43.12`	`tzanger`	`coppice: really?`
`15:43.16`	`anthm`	`also small changes should just be done by the guy committing it and not bother sending it back for minimal alterations`
`15:43.28`	`coppice`	`tzanger: they seem to be casting into stone the things that were just vaguely wrong before`
`15:43.47`	`zoa`	`i think there is a lack of interest from normal users`
`15:43.53`	`zoa`	`they are fast to send emails like make me this`
`15:44.03`	`zoa`	`and then you make it and nobody ever tests it`
`15:44.08`	`konfuzed`	`hhmmm theres always more than one way to do things. Perhaps an GPL project for the Tone Detection that end users can easily grab on their own seperately via ftp or cut and paste or something ;^)`
`15:44.26`	`konfuzed`	`it could have its own web page`
`15:44.36`	`bkw_`	`konfuzed, if you even think about offering up code without disclaiming it to digium you get yelled at and called all kinds of names.`
`15:44.51`	`bkw_`	`I have personally had first hand exp. with that.`
`15:44.57`	`mog_work`	`analog tone detection is never gonna be awesome, its really hard and needs real dev.`
`15:45.00`	`Cresl1n`	`or somebody writes something that doesn't really belong in the public repository (for whatever reason) and they think that just because it went up there it should go in`
`15:45.14`	`coppice`	`mog_work: rubbish`
`15:45.24`	`konfuzed`	`i would say go ahead and disclaim that GPL code to digium`
`15:45.26`	`bkw_`	`Asterisk does too much as it is... It can't do any one thing very well.`
`15:45.27`	`tzanger`	`coppice: ? casting into stone the things that were just vaguely wrong?`
`15:45.33`	`bkw_`	`konfuzed, NO`
`15:45.58`	`coppice`	`tzanger: instead of just doing things badly, they now seem to be firm policies`
`15:46.01`	`Cresl1n`	`bkw_: that's obviously incorrect logic`
`15:46.04`	`anthm`	`naturally you are going to have daftly written patches but take coppice for instance trying to give you guys t38 for goodness sake and it's being nitpicked to death...`
`15:46.08`	`konfuzed`	`bkw_: obviously im missing something`
`15:46.20`	`mog_work`	`hey brian, i mean you are angry at asterisk and us, why do you even come in here?`
`15:46.22`	`konfuzed`	`it comes from not being a programmer my self`
`15:46.34`	`sivana`	`I think we should just convert it all to win32 with wav`
`15:46.59`	`tzanger`	`coppice: I'm gonna convert you to win32 with wav`
`15:47.04`	`sivana`	`hehe`
`15:47.16`	`*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)`
`15:47.19`	`Cresl1n`	`anthm: we don't put hardly anything in without nitpicking, stuff that I put in even is nit-picked`
`15:47.25`	`*** join/#asterisk nguyep (n=chatzill@64.34.203.231)`
`15:47.26`	`Cresl1n`	`anthm: it's called peer review`
`15:47.32`	`bkw_`	`Cresl1n, but you guys totally nit-pick the wrong thing`
`15:47.34`	`anthm`	`certianly`
`15:47.35`	`coppice`	`tzanger: wav is perfectly good, as long as it isn't running on win32`
`15:47.44`	`nguyep`	`any1 use asterisk to connect to sunrocket?`
`15:47.56`	`tzanger`	`nguyep: not me`
`15:48.30`	`coppice`	`Cres11n: if you had a lovely pristine codebase people might think nitpicking was OK. As it is.......`
`15:48.47`	`Cresl1n`	`coppice: it has to get there somehow`
`15:48.59`	`bkw_`	`it should have been done right in the first place`
`15:49.01`	`anthm`	`points of view are easily skewed to help prove a point I suggest you go peer review the code already in there with the same scrutiny I bet you would reject just about every channel driver =D`
`15:49.04`	`Cresl1n`	`coppice: so we can either try to make it better, or we could be apathetic`
`15:49.19`	`tzanger`	`bkw_: should've and could've are irrellavent. I don't see openpbx as doing things the right way right out of the gate either`
`15:49.39`	`bkw_`	`tzanger, As you can tell I don't work on OpenPBX .. never really have.`
`15:49.45`	`mog_work`	`lol`
`15:50.03`	`Cresl1n`	`coppice: but we realize that there are some problems with how things are done right now, and we'd like to try to make them better`
`15:50.11`	`tzanger`	`bkw_: actually I didn't know that, you were one of the biggest drivers behind it if memory serves (It often does not though)`
`15:50.20`	`bkw_`	`yes but I didn't code on it :P`
`15:50.24`	`Cresl1n`	`coppice: so obviously, if you have suggestions for how to do so, then we would like to try to use them`
`15:50.33`	`mog_work`	`bkw seems to pop up quite a bit.....`
`15:50.35`	`bkw_`	`mog_work, cutting fat away isn't coding`
`15:51.06`	`*** join/#asterisk objRobMitch (n=chatzill@c-24-1-203-134.hsd1.tx.comcast.net)`
`15:51.08`	`bkw_`	`Asterisk has so much fat it needs to be put on a diet :P`
`15:51.20`	`konfuzed`	`perhaps some one can confirm for me which OpenSource license it is that * is available under`
`15:51.29`	`mog_work`	`big is beautiful ^_^`
`15:51.32`	`sivana`	`konfuzed: it's dual licensed`
`15:51.37`	`mog_work`	`asterisk is GPL`
`15:51.45`	`mog_work`	`and is available for other licensing from digium`
`15:51.48`	`Cresl1n`	`bkw_: so you say that on one side, then you talk about the time that it take to get new feature patches in on the other... hrm.. makes a LOT of sense`
`15:51.49`	`coppice`	`except when it isn't`
`15:51.53`	`fugitivo`	`gpl2 sucks`
`15:51.57`	`konfuzed`	`hold on`
`15:52.07`	`BoRiS`	`wait for gpl revision 3`
`15:52.11`	`BoRiS`	`coming up soon`
`15:52.16`	`fugitivo`	`it'll suck`
`15:52.16`	`*** part/#asterisk nguyep (n=chatzill@64.34.203.231)`
`15:52.17`	`Beirdo`	`meh, whatever`
`15:52.18`	*** join/#asterisk james` (n=james@85.234.139.77)
`15:52.50`	`konfuzed`	`if asterisk is available under GPL then whats wrong with someone else making a GPL tone detector or anythign else?`
`15:52.57`	`mog_work`	`you coukld`
`15:53.13`	`mog_work`	`but it wont be put into the main tree with out disclaiming`
`15:53.30`	`sivana`	`konfuzed: disclaiming means that Digium can use it as they see fit`
`15:53.38`	`bkw_`	`aka sell it in ABE`
`15:53.38`	`*** join/#asterisk rainkid (n=rainkid@gemini.os5.com)`
`15:53.39`	`anthm`	`the distro cannot contain anything they cannot completely sell to someone or it would invalidate the existing agreements`
`15:53.43`	`konfuzed`	`so does GPL doesnt it`
`15:53.46`	`*** join/#asterisk sachse (n=sachse@86.56.32.11)`
`15:53.53`	`coppice`	`konfuzed: * has no plug in type of scheme. anything external plays endless catchup`
`15:53.55`	`Cresl1n`	`bkw_: so what's wrong with that?`
`15:54.00`	`sachse`	`hi all`
`15:54.07`	`Cresl1n`	`bkw_: is it wrong for you to make money off of using Asterisk in your ITSP?`
`15:54.11`	`*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)`
`15:54.14`	`fugitivo`	`why not making a new snv or cvs tree with only 100% GPL code?`
`15:54.16`	james`	`My CID if i called another extention is correct, but if i call a context i have setup the CID always is "device" can any one shead some light on this?`
`15:54.25`	`rainkid`	`how do you get the caller id of an incoming call?`
`15:54.33`	`bkw_`	`Cresl1n, Nope not at all.`
`15:54.42`	`sachse`	`has some1 experience with asterisk and freenet in germany?`
`15:54.43`	`Cresl1n`	`bkw_: maybe you should be a little more fair with how you think`
`15:54.52`	`anthm`	`you should not bother argueing moot points I am happy to hear them say they know there is a problem so I am gonna see what becomes of that before I dig up any more code.`
`15:55.52`	`tzanger`	`rainkid: ${CALLERID(all)} or variants. "show function CALLERID"`
`15:56.56`	`anthm`	`btw where is my svn branch did i get one?`
`15:57.18`	`konfuzed`	`ok so makie it an external patch project that can then only be had manually by system operators . do at your own risk, unsupported and not in the main tree`
`15:57.34`	`Cresl1n`	`anthm: I didn't think we were under the impression that you wanted anything to do with asterisk anymore`
`15:57.46`	`*** join/#asterisk santiago (n=santiago@208.195.215.97)`
`15:57.56`	`anthm`	`what a waste that would be i can practicly recite the api calls`
`15:58.16`	`sivana`	`can I get one too, an svn branch, for my c# conversion`
`15:58.31`	`Uther_P`	`ack`
`15:58.36`	`sachse`	`problem: freenet asterisk 1.0.9 gentoo kernel 2.6.12-r6: outgoing calls works, incomming not. sipgate work in both directions. help?!`
`15:58.40`	`Cresl1n`	`and one for my J++ conversion too :-)`
`15:58.44`	`coppice`	`sivana: c# conversion? is this a religious thing? :-)`
`15:58.49`	`rainkid`	`thank s`
`15:58.51`	`Uther_P`	`hkaha`
`15:59.00`	`sivana`	`heh`
`15:59.05`	`BoRiS`	`c#=scary, j+=Very Nasty`
`15:59.09`	`*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)`
`16:00.01`	`anthm`	`I mean should I spend 2 years making asterisk do all this stuff then not use it or anything?`
`16:00.02`	`Uther_P`	`what was wrong with C++ that ms had to go and screw it up?`
`16:00.17`	`Uther_P`	`(not there needs to be a specific reason for ms to screw something up)`
`16:00.18`	`BoRiS`	`c++ is good`
`16:00.26`	`tzanger`	`acutally I have heard from many people that C# is quite nice`
`16:00.29`	`tzanger`	`I've never used it myself`
`16:00.40`	`MRH2`	`hi does the zaptel echo can only work for the external connected part of the call? SO you would still get echo on the asterisk side?`
`16:00.51`	`coppice`	`C++ is so nasty, it would be hard for MS to actually wreck it :-)`
`16:01.02`	`tzanger`	`I like plain old C`
`16:01.03`	`anthm`	`I like C+0`
`16:01.04`	`Cresl1n`	`MRH2: it depends on how long the the echo tail is`
`16:01.06`	`jbalcomb`	`MRH2: I believe so. Zapatel is just the Telco side`
`16:01.06`	`tzanger`	`picking up python`
`16:01.22`	`Cresl1n`	`MRH2: generally it should only need to do it for the call side`
`16:01.23`	`BoRiS`	`I prefer C-3 (Cubed) :-p`
`16:01.28`	`rkioko`	`hi guys`
`16:01.29`	`BoRiS`	`c+++`
`16:01.31`	`jbalcomb`	`QBASIC is best`
`16:01.35`	`BoRiS`	`LOL!`
`16:01.37`	`Uther_P`	`haha`
`16:01.41`	`Uther_P`	`qb45 rocks`
`16:01.43`	`lunk`	`gorllas.bas = best game on the planet`
`16:01.43`	`Uther_P`	`yay`
`16:01.44`	`*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)`
`16:01.45`	`BoRiS`	`I actually Bought the Qbasic 4.5 box`
`16:01.46`	`jbalcomb`	`damn right`
`16:01.47`	`Uther_P`	`haha`
`16:01.49`	`tzanger`	`lunk: hahaha`
`16:02.03`	`Uther_P`	`dont forget nibbles.bas`
`16:02.10`	`jbalcomb`	`haha.. i remember gorillas from my first programming class in highschool`
`16:02.11`	`BoRiS`	`and that wonder PC speaker music`
`16:02.14`	`coppice`	`MRH2: its the other way round.`
`16:02.16`	`lunk`	`i chuck exploding bananas at little worms`
`16:02.17`	`lunk`	`haha`
`16:02.22`	`BoRiS`	`wonderful`
`16:02.25`	`anthm`	`I heard locksmith 2.0 is out now you can copy floppy disks.`
`16:02.40`	`Uther_P`	`no way!`
`16:02.42`	`lunk`	`anthm: scotch tape has been around for years!`
`16:02.52`	`PoWeRKiLL`	`someone know about this error Jan 12 17:03:20 WARNING[9904]: chan_iax2.c:3732 iax2_trunk_queue: Maximum trunk data space exceeded to ?`
`16:03.10`	`coppice`	`lunk: many countries have had similar tape for just as long`
`16:03.16`	`tzanger`	`PoWeRKiLL: you filled up your iax2 trunk`
`16:03.19`	`tzanger`	`how many simultaneous calls?`
`16:03.34`	`MRH2`	`so echo can is for the person connected to the zaptel card only?`
`16:03.37`	`Uther_P`	`especially the scottish`
`16:03.45`	`anthm`	`the cheater answer is to turn up the constants of max trunk space`
`16:03.56`	`anthm`	`at the top of chan_iax.c`
`16:04.16`	`Uther_P`	`MRH2: echo can wouldn't serve any purpose to anything but whats connected to the zaptel`
`16:04.32`	`Uther_P`	`MRH2: voip isn't going to echo its packets :)`
`16:04.53`	`Uther_P`	`sidetone is a bitch`
`16:05.03`	`*** join/#asterisk oli1234 (n=olivier@vodsl-8055.vo.lu)`
`16:05.11`	`MRH2`	`yes it certainly is`
`16:05.33`	`PoWeRKiLL`	`tzanger how I did that ?`
`16:05.41`	`oli1234`	`hello, ihave certain problems to load sipusers form a mysql table... is there anybody who could help me?`
`16:05.41`	`coppice`	`MRH2: if you use a digital card, there will be no echo back to the caller. if you use an analogue card * cannot cancel the echo it causes, but it shouldn't really matter. the important thing is audio from an IP phone should not be reflected back to that phone. that is what will sound bad`
`16:05.43`	`tzanger`	`PoWeRKiLL: how many simultaneous calls were you trying to push through the trunk?`
`16:05.45`	`MRH2`	`wondering if it would be too long an echo to loop voip calls through zap? or even if it would be a good idea?`
`16:06.13`	`tzanger`	`MRH2: PRIs and CAS T1/E1s do not GENERATE echo. Hoewver you can still GET echo on them`
`16:06.16`	`PoWeRKiLL`	`tzanger : usually I have 10 calls`
`16:06.25`	`tzanger`	`what codec?`
`16:06.29`	`PoWeRKiLL`	`g729`
`16:06.34`	`tzanger`	`hmm`
`16:06.41`	`anthm`	`like i said turn up the constants`
`16:06.45`	`PoWeRKiLL`	`now when 1 calls arrive i got this error`
`16:06.46`	`Uther_P`	`voip loopback across 15 hops == perfect guitar reverb`
`16:06.47`	`anthm`	`they are very liberal`
`16:06.50`	`Uther_P`	`:D`
`16:06.56`	`tzanger`	`Uther_P: :-)`
`16:07.00`	`*** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it)`
`16:07.04`	`tzanger`	`anthm: you mean conservative?`
`16:07.10`	`*** join/#asterisk monachoi (n=bonvoyag@cpe-24-174-162-34.satx.res.rr.com)`
`16:07.19`	`anthm`	`i suppose i do`
`16:07.24`	`coppice`	`Uther_P: if you don't mind a 3.5kHz limited guitar :-)`
`16:07.43`	`tzanger`	`coppice: depends on the tune :-)`
`16:07.55`	`oli1234`	`I have certain problems to load sipusers form a mysql table... is there anybody who could help me? --> just contact me in private thx in advance`
`16:08.07`	`*** join/#asterisk FastJack (i=fastjack@p5091FE1E.dip.t-dialin.net)`
`16:08.14`	`Uther_P`	`coppice: sum it back to the original tone`
`16:08.24`	`Uther_P`	`heh, pull it out of phase too`
`16:08.43`	`*** part/#asterisk sachse (n=sachse@86.56.32.11)`
`16:09.17`	`PoWeRKiLL`	`tzanger any idea ?`
`16:09.39`	`coppice`	`isn't it sad that after nearly 20 years of ISDN, which was supposed to bring us wideband voice, we still use narrow band for almost all VoIP?`
`16:09.46`	`*** join/#asterisk zukzuk (n=c@p508709B7.dip0.t-ipconnect.de)`
`16:10.13`	`sivana`	`coppice: why is that`
`16:10.13`	`zukzuk`	`hey guys. does anybody, by chance, know a way to work around this problem: http://bugs.digium.com/view.php?id=5838&nbn=7 ?`
`16:10.19`	`zukzuk`	`i'm experiencing the exact same thing`
`16:10.35`	`tzanger`	`PoWeRKiLL: did you listen to anthm`
`16:10.44`	`Mimmus`	`I forgot nickname of a really valid guy who helped me a few minutes ago about a toll-free number not responding.... can anyone help me?`
`16:10.51`	`*** join/#asterisk Dorphalsig (n=Dorphals@200.71.58.39)`
`16:10.55`	`sivana`	`Mimmus: Cresl1n`
`16:11.02`	`Mimmus`	`ok, thanks`
`16:11.14`	`PoWeRKiLL`	`thanks anthm :)`
`16:11.22`	`Uther_P`	`a really 'valid' guy`
`16:11.24`	`Uther_P`	`haha`
`16:11.32`	`anthm`	`#define DEFAULT_TRUNKDATA 640 * 10`
`16:11.32`	`anthm`	`#define MAX_TRUNKDATA 640 * 200`
`16:11.40`	`coppice`	`sivana: because people tolerate any old crap, I guess. people like Skype, yet don't scream for wideband elsewhere`
`16:11.48`	`anthm`	`crank those and recompile`
`16:11.53`	`anthm`	`note it's a band aid`
`16:12.13`	`sivana`	`coppice: who does wideband right now?`
`16:12.17`	`*** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`16:12.41`	`PoWeRKiLL`	`anthm i have to change this #define TRUNK_CALL_START 0x4000 ?`
`16:12.50`	`anthm`	`i just pasted the 2`
`16:12.55`	`PoWeRKiLL`	`thanks`
`16:12.56`	`coppice`	`skype is the only major user. a number of UMTS users have wideband - if they call another suitable UMTS user`
`16:13.06`	`*** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`16:13.09`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway`
`16:13.20`	`coppice`	`anthm: 22kHz is a weird rate to use`
`16:13.29`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`16:14.00`	`anthm`	`one leg was a soundcard`
`16:14.10`	`anthm`	`i could have done 16 and 32 also`
`16:14.18`	`tzanger`	`All of my stuff is PSTN ended so wideband does absolutely dick-all for me`
`16:14.53`	`anthm`	`i'm concerned how to negotiate the wideband stuff seems like the rate in the sdp is only like a kinda sorta option`
`16:14.54`	`coppice`	`soundcards are a pain for VoIP. their sampling rates don't lock to anything - include tx not locking to rx`
`16:15.29`	`anthm`	`iax doesnt seem to have any rate element`
`16:15.38`	`anthm`	`so that will be fun`
`16:15.51`	`coppice`	`anthm: shouldn't be. if one end announces only 8kHz codecs, the other end certainly shouldn't choose something higher.`
`16:15.59`	`MRH2`	`thanks I am going to blame echo on the other party for the moment.`
`16:16.15`	`sivana`	`Session Description Protocol ?`
`16:16.16`	`anthm`	`well that act of announcing the 16k codec is what i am wondering about`
`16:16.23`	`sivana`	`Social Democratic Part?`
`16:16.26`	`coppice`	`IAX lacks a number of important things if it is to break into the big time.`
`16:16.42`	`coppice`	`Session Dementing Protocol`
`16:16.43`	`*** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net)`
`16:18.14`	`anthm`	`so at least it seems like spd has the concept of it but it's barely used so it's not likely it will be understood by much stuff`
`16:18.17`	`coppice`	`The CNG frame is useless. There is no proper allowance for sample rates. The text is not defined as being UTF-8. Various little odds and ends that nobody seems to care to sort out, but which will cripple it.`
`16:18.20`	`*** join/#asterisk dily_ (n=dily@host91-30.pool80105.interbusiness.it)`
`16:18.21`	`anthm`	`sdp i mean`
`16:19.19`	`Mimmus`	`sivana: now I have a different but seemingly related problem, can I try here or file a bug?`
`16:19.19`	`coppice`	`I think there should be no problems with SDP. Things that don't understand the rates will not support the related codecs. It should sort itself out`
`16:19.31`	`*** join/#asterisk www2 (n=www1985@cd4400448.cable.wanadoo.nl)`
`16:19.40`	`anthm`	`that's what i'm hoping for`
`16:20.08`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`16:20.41`	`*** join/#asterisk secure75 (n=mic@dslb-084-057-013-245.pools.arcor-ip.net)`
`16:20.42`	`jbalcomb`	`Should I be using uLaw or aLaw and what's the difference?`
`16:20.54`	`coppice`	`I think it should fall into place a lot better than T.38 :-) The spec for that fails to tie a whole mass of things down.`
`16:21.05`	`anthm`	`i guess iax could send an IE with khz in it but that should be fun getting it accepted`
`16:21.30`	`anthm`	`there is a bit of a difference between ulaw and alaw`
`16:22.05`	`coppice`	`jbalcomb: they won't talk to each other, but their quality is about the same`
`16:22.47`	`jbalcomb`	`coppice: is it correct that one is an american standard and the other is a european standard? if so, which is which?`
`16:23.19`	`coppice`	`ulaw = US, HK, Taiwan, Japan`
`16:23.20`	`coppice`	`Alaw = the rest of humanity`
`16:23.31`	`coppice`	`oh, i missed canada`
`16:23.35`	`tzanger`	`:-)`
`16:23.39`	`tzanger`	`don't worry, everyone does. :-)`
`16:23.42`	`Uther_P`	`its common`
`16:24.27`	`jbalcomb`	`coppice: excellent. i assume uLaw correlates to PCMU vocoder on my grandstream phones?`
`16:24.32`	`*** join/#asterisk jero (n=sflphone@savoirfairelinux.net)`
`16:24.37`	`Uther_P`	`yes`
`16:24.50`	`jero`	`hi`
`16:24.52`	`coppice`	`quite a few phones call them PCMU and PCMA`
`16:25.11`	`jbalcomb`	`excellent. i think all executive decisions regarding our codec setup have been made.`
`16:25.17`	`jbalcomb`	`thanks for the help yall`
`16:25.24`	`Cresl1n`	`anthm: IIRC, I think there's an IE for sample rate`
`16:25.38`	`anthm`	`oh that would be good`
`16:25.44`	`Cresl1n`	`anthm: (in IAX)`
`16:25.59`	`Cresl1n`	`I started working on wideband too, and that was one of the things mark mentioned to me`
`16:26.02`	`*** part/#asterisk www2 (n=www1985@cd4400448.cable.wanadoo.nl)`
`16:26.15`	`*** join/#asterisk Strom_C (n=strom@216-80-66-245.lem-bsr1.chi-lem.il.cable.rcn.com)`
`16:26.20`	`anthm`	`expressed in hz ?`
`16:26.33`	`Cresl1n`	`anthm: hrmm... not sure on that one`
`16:26.42`	`jbalcomb`	`ah snap, one more codec question. the grandstream codex FAQ is using kbps but the Cisco codec FAQ is using Kbps. is it kilobits or kilobytes that i should be thinking?`
`16:26.53`	`*** join/#asterisk rick222 (n=rick555@207.71.127.152)`
`16:28.12`	`*** join/#asterisk jijgeh (n=luken@static-66-182-95-76.bbsc.net)`
`16:28.15`	`anthm`	`yah i see unsigned short samprate; in iax2-parser.h`
`16:29.04`	`Cresl1n`	`anthm: cool, yeah, I thought there was one`
`16:29.06`	`Mimmus`	`Cresl1n: now I have a different but seemingly related problem, can I disturb you again or is there a better choice?`
`16:31.17`	`*** join/#asterisk uther (n=uther_p@66.180.120.82)`
`16:32.22`	`mog_work`	`woot 239!`
`16:32.33`	`wunderkin`	`oops, found 2 more bugs`
`16:32.36`	`Cresl1n`	`mog_work: yeppers`
`16:32.44`	`Cresl1n`	`Mimmus: ???`
`16:33.11`	`Mimmus`	`Cresl1n: yes, I'm`
`16:33.15`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`16:33.26`	`anthm`	`i wonder why it's unsigned short considering how stingy the rest of the lements are it could have been char since in khz the vals are all < 50`
`16:33.36`	`Mimmus`	`Cresl1n: different number and slightly different problem`
`16:33.45`	`anthm`	`at least it's there already`
`16:33.46`	`Mimmus`	`Cresl1n: but always an automatci responder`
`16:33.52`	`Cresl1n`	`anthm: so you could send 8600hz audio :-)`
`16:34.36`	`Cresl1n`	`anthm: here wait... so you can send 8633hz audio :-)`
`16:35.34`	`*** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com)`
`16:35.41`	`anthm`	`hmm looks like there are hard constants for various rates in a bitmask`
`16:35.48`	`*** join/#asterisk montag___ (n=montag@host187-252.pool8175.interbusiness.it)`
`16:35.53`	`anthm`	`8 11 16 22 44 48`
`16:36.03`	`montag___`	`it's there a way to define custom greetings for every voicemail mailbox ?`
`16:36.17`	`dos000`	`anyone has idea about a cheap 2 port sip 2 pstn gateway ?`
`16:36.19`	`aminorex`	`oh that england had but one head so that i might strike it off`
`16:36.50`	`wunderkin`	`montag___, yes i use the temp greeting for that`
`16:37.16`	`*** join/#asterisk DFS (n=bwarner@65.113.208.11)`
`16:37.31`	`DFS`	`Anyone available for a dialplan question?`
`16:37.35`	`montag___`	`wunderkin: ?`
`16:37.45`	`anthm`	`aha so you must have the same bitmap on both sides and you actually send the unsigned version of that paticular constant not the desired speed`
`16:37.47`	`Mimmus`	`montag___: if I remember well, hit '3' for special functions`
`16:38.02`	`Cresl1n`	`anthm: those are some funny sample rates`
`16:38.12`	`DFS`	`Anyone available for a dialplan question?`
`16:38.21`	`montag___`	`but i want to manage this file from filesystem, not from user dtmf interface`
`16:38.33`	`{zombie}`	`DFS: just ask the question, don't ask if you can ask`
`16:38.39`	`{zombie}`	`and please don't repeat yourself`
`16:38.48`	`coppice`	`anthm: they miss an important one for telephony - 32`
`16:38.51`	`Cresl1n`	`8 16 32 and 48 should probably be in there`
`16:38.59`	`Mimmus`	`montag___: they are under /var/spool/asterisk/voicemail`
`16:39.00`	`anthm`	`yah were is 32 ?`
`16:39.06`	`wunderkin`	`montag___, funny thing.. they are saved on the filesystem.. so if you do it from the menu and look in the directory you will see how it works`
`16:39.10`	`Cresl1n`	`but I don't know about the non even multiple choices`
`16:39.16`	`DFS`	`Zombie>>I am trying to set up a dialplan where I can call other voip users on another asterisk server in a diff. network`
`16:39.22`	`coppice`	`they miss 192 as well`
`16:39.36`	`Cresl1n`	`oh yeah, and 384 too :-P`
`16:39.45`	`coppice`	`don't be silly`
`16:40.18`	`montag___`	`wunderkin: ok, but you know the name for busy and unavailable files ?`
`16:40.20`	`dily_`	`anyone use bristuff?`
`16:40.31`	`coppice`	`192k, 24 bit 7.1 is bound to be de rigeur for audio conferencing this year`
`16:40.39`	`wunderkin`	`montag___, you can research that the same way`
`16:40.42`	`{zombie}`	`DFS: Dial(IAX2/user:pass@remoteserver/XXXX)`
`16:40.51`	`anthm`	`so when you convert it to bits`
`16:40.57`	`Cresl1n`	`coppice: ah, didn't realize that`
`16:41.09`	`montag___`	`ok thanks`
`16:41.23`	`Mimmus`	`montag___: unavail.wav and busy.wav (.WAV too)`
`16:41.32`	`*** join/#asterisk grandy (n=mmmurf@pcp05305753pcs.wanarb01.mi.comcast.net)`
`16:41.46`	`DFS`	`Zombie: Where do you place this...in extentions.conf or in IAX?`
`16:42.12`	`{zombie}`	`you put that in your extensions.conf`
`16:42.19`	`*** join/#asterisk ffs_04 (n=jbon@modemcable071.144-80-70.mc.videotron.ca)`
`16:42.37`	`anthm`	`nothing = 1`
`16:42.37`	`anthm`	`8k = 2`
`16:42.37`	`anthm`	`16k = 4`
`16:42.37`	`anthm`	`22k = 8`
`16:42.37`	`anthm`	`44k = 16`
`16:42.38`	`anthm`	`48k = 32`
`16:43.12`	`anthm`	`the seem strikingly similar to just sending the rate you want rounded to nearest khz`
`16:43.18`	`dos000`	`anyone know a 2 port gw (not ata) that will allow phone<->ata<->internet<->gw<->pstn ?`
`16:43.42`	`{zombie}`	`DFS: http://voip-info.org/wiki/view/Asterisk+Connect+2+servers would be good reading`
`16:44.09`	`coppice`	`i wonder what the difference between an ATA and a GW might be :-\`
`16:44.31`	`dos000`	`coppice, no fxo on the ata normally`
`16:44.42`	`rue_work`	`[TK]D-Fender you up?`
`16:44.49`	`Mimmus`	`Cresl1n: is it a good idea to open a bug on digium.com or is it better to wait here?`
`16:45.14`	`coppice`	`dos000: so you cook up your own terminology, and expect everyone to understand? :-\`
`16:45.28`	`dos000`	`coppice, even if you have an fxo interface you can only originate not terminate`
`16:45.34`	`DFS`	`zombie: Is there anything else I need to add? Just this statement with my info in extentions.conf?`
`16:45.55`	`{zombie}`	`DFS: I think you need to do a whole lot more reading...`
`16:46.07`	`anthm`	`what's the max val of unsigned short?`
`16:46.09`	`{zombie}`	`don't expect you can just throw random statements into your asterisk config files and make things wrk`
`16:47.10`	`jbalcomb`	`one more codec question. the grandstream codex FAQ is using kbps but the Cisco codec FAQ is using Kbps. is it kilobits or kilobytes that i should be thinking?`
`16:47.17`	`coppice`	`anthm: is this a trick question?`
`16:47.18`	`Beirdo`	`anthm: 2^16 - 1`
`16:47.35`	`Uther_P`	`jbalcomb: bits`
`16:47.45`	`Beirdo`	`65535`
`16:48.09`	`anthm`	`so too small to send hz`
`16:48.11`	`Cresl1n`	`Mimmus: it's always better to verify here or on the mailing list that it's actually a bug before you post one (like earlier with the 'r' flag in the Dial command)`
`16:48.21`	`anthm`	`but big enough to send rounded khz`
`16:48.23`	`jbalcomb`	`Uther_P: ah, most peculiar that Ciscos page would be incorrect. That certainly explains my confusion in the amount of traffic I'm seeing. Thank you.`
`16:48.34`	`Uther_P`	`usually kilobytes per second is denoted as k/s`
`16:48.41`	`coppice`	`the maximum value of a short int is when it saves 2 bytes of memory and squeezes the product into a much cheaper MCU or DSP :-`
`16:49.04`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`16:49.17`	`Mimmus`	`Cresl1n: this is correct. Now I try to explain (it isn't so simple)`
`16:49.27`	`*** join/#asterisk A-Tuin\|work (n=A-Tuin@nat.office.legend.net.uk)`
`16:49.33`	`rue_work`	`WARNING[19687] chan_iax2.c: Received mini frame before first full voice frame`
`16:49.36`	`Dorphalsig`	`btw, checking my zaptel modules... I noticed I have wct4xxp and tor2 up`
`16:49.36`	`anthm`	`i'm trying to figure out why the iax code has the rates in a bitmask is there a condition where you can have 2 rates at once ?`
`16:49.38`	`rue_work`	`anyone know where that commes from?`
`16:49.42`	`Mimmus`	`Cresl1n: I have (PSTN PRI) -- Asterisk --- Alcatel PBX --- analog phones`
`16:49.47`	`Uther_P`	`I would like anyone to find somewhere where bps is used to denote bytes per second`
`16:49.49`	`Dorphalsig`	`shouldnt I just have one of them?`
`16:49.50`	`coppice`	`i remember once spending over a week getting one instruction out of a DSP loop :-\`
`16:49.55`	`Mimmus`	`Cresl1n: ans some VoIP phones directly connected to Asterisk`
`16:50.15`	`anthm`	`since nothing uses it i was brainstorming other ways to send the rate in the constraints of the unsigned short it is declared as`
`16:50.26`	`*** join/#asterisk bhickey (n=chatzill@212.2.174.21)`
`16:50.31`	`coppice`	`anthm: of course. if a phone supports 8k and 16k you set two bits`
`16:50.43`	`Mimmus`	`Cresl1n: calling a number with automatci responder from analog phones doesn't work (NONSWER after two rings), from Voip phones works`
`16:51.04`	`Cresl1n`	`coppice: that sounds like it could cause problems`
`16:51.05`	`coppice`	`dunno why the IE can't have a list of shorts with all the possible rates, though`
`16:51.43`	`*** join/#asterisk }btorch{ (n=kvirc@208.63.19.172)`
`16:52.25`	`Cresl1n`	`coppice: well, I take that back. I guess it depends on how it 's used`
`16:52.40`	`}btorch{`	`when someone asks what voice standards * can support are they talking about the technology like SIP, GSM`
`16:53.07`	`coppice`	`Cres11n: its dumb trying to squeeze this down and loose flexibility. its only sent infrequently`
`16:54.04`	`Cresl1n`	`coppice: you mean with doing it as a bit mask?`
`16:54.16`	`Cresl1n`	`coppice: I think there's truth to that`
`16:54.27`	`anthm`	`you can send several ie with the same name correct?`
`16:54.49`	`anthm`	`you also have no way to tie which rate goes with which codec`
`16:54.58`	`coppice`	`why should you? an IE has variable length, so it can contain a list of things`
`16:55.02`	`Cresl1n`	`anthm: yeah, that's what I was concerned about`
`16:55.07`	`_Sam--`	`does any know if that cheap DLINK packet prioritizer recognizes IAX? http://www.voipsupply.com/product_info.php?manufacturers_id=45&products_id=1168`
`16:55.19`	`*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)`
`16:55.29`	`Cresl1n`	`anthm: so if you blindly advertised 8 and 16, but you don't support 16 on all 16 capable codecs, that's a problem`
`16:56.14`	`coppice`	`well, the whole rate concept is restrictive. take AMR WB for example. what is its rate?`
`16:57.51`	`Cresl1n`	`coppice: yeah, I think that the way rates are done are going to have to be rethought`
`16:57.56`	`*** join/#asterisk cfh (n=luca@82.193.23.6)`
`16:57.58`	`anthm`	`maybe send mutiple capability and mutiple rate per groups`
`16:58.15`	`coppice`	`i think specifying rates is a fundamentally bad idea. you should just specify the codecs. they defines their capabiliites`
`16:58.15`	`anthm`	`send 1 cap with ulaw alaw then send 1 rate with 8 and 16`
`16:58.18`	`anthm`	`then send the rest`
`16:58.23`	`anthm`	`and send a sigle 8`
`16:58.50`	`nextime`	`wath's the best ( as a stability and performance ) h323 channel, h323, oh323 or ooh323?`
`16:58.54`	`coppice`	`anthm: just what about AMR WB? or speex? they can dynamically change their rates, to deal with congestion`
`16:59.11`	`anthm`	`yes`
`16:59.20`	`cfh`	`is there a solution to config the fast numbers on asterisk server?`
`16:59.27`	`anthm`	`maybe we should hack all the other codecs to be able to negotiate thier own rate`
`16:59.28`	`watchy-`	`im sitting here naked, i just took a shower`
`16:59.32`	`watchy-`	`i feel so sexy`
`16:59.41`	`Beirdo`	`TMI`
`16:59.56`	`watchy-`	`u sure`
`17:00.08`	`Beirdo`	`absolutely`
`17:00.18`	`anthm`	`maybe sdp over IE =D`
`17:00.22`	`*** join/#asterisk zapotecz (n=surfer@217.201.198.236)`
`17:00.23`	`rue_work`	`as some of you may have noticed, I'm not verry farmiliar with asterisk. I'm currently trying to correct issues with a PSTN machine that kb1canobie assembled, who you may know of. I could really use some help going through teh errors on the system while I try to correct some issues that are making the people in the office really agitated (theyre damanding that the phone system be replaced completely) the first thing I want to resolv is the mos`
`17:00.33`	`coppice`	`a list of codecs, detailed enough to define the specific variants supported, should be a complete description`
`17:00.39`	`*** join/#asterisk buzzyd (n=buzzyd@82-45-247-173.cable.ubr01.enfi.blueyonder.co.uk)`
`17:01.22`	`dos000`	`coppice, still no idea about what i asked ?`
`17:01.44`	`rue_work`	`I also verrymch need to fix the voicemail, which keeps recording blank messages`
`17:01.58`	`dos000`	`rue_work, tow !`
`17:02.01`	`buzzyd`	`Hi All, anyone know how I can setup voicemail prompts instead of using the default american voiced ones when leaving voicemail messages`
`17:02.51`	`rue_work`	`buzzyd the files you want to re-record are in the directory /var/lib/asterisk/sounds/`
`17:02.52`	`Uther_P`	`buzzyd: eh?`
`17:02.59`	`buzzyd`	`I would like my users to be able to set their own message but I can't see where to configure it`
`17:03.16`	`Mimmus`	`buzzyd: language setting set also messages for voicemail but you need sounds file for your lang`
`17:03.17`	`anthm`	`enough of this dealing with issues that control the outcome of any success in the near future lets fix config issues`
`17:03.19`	`rue_work`	`buzzyd if I understood you right`
`17:03.29`	`buzzyd`	`rue_work that would change it for everyone though`
`17:03.32`	`*** join/#asterisk masonf (n=masonf@dungle.vineyard.net)`
`17:03.43`	`rue_work`	`buzzyd sorry I misunderstood`
`17:03.46`	`zapotecz`	`hi all`
`17:03.54`	`rue_work`	`and I'm not verry farmiliar iwth asterisk`
`17:03.58`	`Mimmus`	`buzzyd: and any user can record his/her message hitting '3'`
`17:03.58`	`buzzyd`	`I just want it so each person can have their own message instead of playing a standard one for all`
`17:03.58`	`zapotecz`	`no one has used the patch for the bearer?`
`17:04.02`	`zapotecz`	`http://bugs.digium.com/view.php?id=3547&nbn=26#bugnotes`
`17:04.14`	`masonf`	`what are some possible causes for the message: Unable to open Asterisk database?`
`17:04.18`	`rue_work`	`dispite that I need to fix a number of issues on a system`
`17:04.20`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
`17:04.33`	`rue_work`	`which I could really use someone talking me though`
`17:04.35`	`Mimmus`	`masonf: permissions?`
`17:05.05`	`masonf`	`Ill try running as root....`
`17:05.06`	`buzzyd`	`Mimmus, anyway of doing it without using that app`
`17:05.30`	`Mimmus`	`buzzyd: yes, record and save message under /var/spool/asterisk/voicemail/...`
`17:05.46`	`Mimmus`	`masonf: no no, usually Asterisk runs as asierisk user`
`17:05.54`	`Mimmus`	`masonf: asterisk user`
`17:05.55`	`masonf`	`yeah its permissions... now I need to find what files it wants.`
`17:06.00`	`rue_work`	`in http://pastebin.com/502582 that context, does anyone know what 'outage' should sound like?`
`17:06.11`	`buzzyd`	`mimmus: ok but how would I then link that to each account`
`17:06.16`	`DFS`	`zombie: Can you specifiy the host as an IP address when creating the REC server?`
`17:06.25`	`dily_`	`exit`
`17:06.27`	`dos000`	`buzzyd, check out theese guys http://actor.loquendo.com/actordemo/default.asp?language=en`
`17:06.36`	`Mimmus`	`buzzyd: there is a directory for any extension under /var/spool/asterisk/voicemail/default/...`
`17:07.39`	`*** join/#asterisk dasuberdavid (n=david@gateway.digium.com)`
`17:08.13`	`*** join/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it)`
`17:10.39`	`buzzyd`	`Thanks guys, I see it now ;)`
`17:11.04`	`DFS`	`zombie: when specifying the host on the REC server, can you use the IP address`
`17:11.13`	`Mimmus`	`buzzyd: you are welcome`
`17:11.41`	`Mimmus`	`hey people, even Mimmus is able to help someone!`
`17:12.29`	`DFS`	`mimmus:you familiar with configuring two asterisks to conduct calls between the two on two diff. networks?`
`17:12.36`	`*** join/#asterisk juice (n=juice@209.33.109.45)`
`17:12.54`	`Mimmus`	`DFS: using IAX?`
`17:13.08`	`DFS`	`yes...I've read the text on [REC_SERVER]`
`17:13.08`	`DFS`	`type=user`
`17:13.08`	`DFS`	`host=my.calling.server.ca`
`17:13.08`	`DFS`	`secret=mysecret`
`17:13.08`	`DFS`	`context=local`
`17:13.09`	`DFS`	`trunk=yes`
`17:13.44`	`DFS`	`this is where I'm confused....based on the site reading :http://voip-info.org/wiki/view/Asterisk+Connect+2+servers`
`17:14.05`	`DFS`	`where the host is my.calling.server...example... can u use an IP address instead?`
`17:14.17`	`Mimmus`	`DFS: yes, Ip is good`
`17:14.39`	`DFS`	`mimmus: thanks...wasn't for sure if it would still work...`
`17:15.15`	`Mimmus`	`I'm not sure what context stands for`
`17:15.44`	`*** join/#asterisk roulduke_ (i=yz6mgq5v@p508D0F3D.dip0.t-ipconnect.de)`
`17:17.17`	`mocker`	`I'm having a problem w/ Asterisk receiving faxes. The tif files appear to be all crunched up into about 1 inch instead of looking like a normal fax page.`
`17:17.20`	`mocker`	`Is that normal?`
`17:17.24`	`DFS`	`mimmus: do you create a new secret for IAX or do you use the current secret for registering devices?`
`17:18.55`	`*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)`
`17:18.57`	`Mimmus`	`DFS: secret is a 'password' between two peers`
`17:20.00`	`DFS`	`mimmus: correct..this password I have is different for each asterisk...which do I use..or do I create a new one`
`17:21.59`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`17:22.21`	`masonf`	`any ideas what files would be giving be permissions issues Ive already checked /var/log /etc/asterisk /var/spool and /var/run`
`17:23.25`	`fulgas`	`strace asterisk`
`17:23.47`	`fulgas`	`and check for the permissions problem`
`17:24.10`	`*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)`
`17:24.14`	`*** part/#asterisk cfh (n=luca@82.193.23.6)`
`17:24.36`	`generalhan`	`whats up everyone ? !`
`17:24.41`	`DFS`	`mimmus: what is the [mycontext] ? What do you specify for that?`
`17:25.39`	`*** join/#asterisk A-jay (n=quirc@62.217.245.194)`
`17:26.24`	`generalhan`	`can anyone help me out with a compling problem im having with zaptel-1.2.1 ???`
`17:27.29`	`Mimmus`	`DFS: I used the same on both servers`
`17:27.49`	`*** join/#asterisk rkioko (n=rkioko@196.200.26.42)`
`17:28.16`	`Mimmus`	`DFS: if you like, I can post my con on pastebin.com`
`17:29.25`	`DFS`	`mimmus: that would be great...this project is confusing`
`17:29.53`	`Mimmus`	`DFS: just a moment...`
`17:32.30`	`[TK]D-Fender`	`rue_work : Here`
`17:32.55`	`*** join/#asterisk denon (i=denon@synapse.subneural.net)`
`17:32.55`	`*** mode/#asterisk [+o denon] by ChanServ`
`17:33.21`	`[TK]D-Fender`	`rue_work : that last pastebin of yours is very wrong. When you're back I'll help you fix it up`
`17:33.24`	`watchy`	`i got voicemail setup`
`17:33.27`	`watchy`	`how do i access it?`
`17:34.16`	`[TK]D-Fender`	`watchy : set up an extension that your phones can dial like "exten => 1234,1,VoicemailMain"`
`17:34.26`	`watchy`	`oh`
`17:34.33`	`DFS`	`mimmus: who will you post as`
`17:34.35`	`Mimmus`	`DFS: here http://pastebin.com/502626`
`17:34.37`	`watchy`	`so if i dial it from the actuall phone`
`17:34.41`	`watchy`	`it'll let me hear vM?`
`17:34.41`	`DFS`	`mimmus: thanks`
`17:35.09`	`[TK]D-Fender`	`watchy : that will bring you to the VM "main" where it'll ask you which VM box & password and then let you listen`
`17:35.19`	`watchy`	`ah`
`17:35.21`	`Mimmus`	`DFS: a small error, look here: http://pastebin.com/502627`
`17:35.26`	`watchy`	`and using variables i can auto goto the box?`
`17:36.16`	`[TK]D-Fender`	`watchy : like this - "exten => *98,1,VoicemailMain(${CALLERID(num)}@default)`
`17:36.22`	`watchy`	`sweet`
`17:36.33`	`watchy`	`so whats 98? literally 98?`
`17:37.00`	`[TK]D-Fender`	`watchy : that will assume your phones callerid is the same as its VM box #. You can script it up any which way you want like say "if its 555 then use box 222" or whatever`
`17:37.20`	`[TK]D-Fender`	`watchy : exactly like *98 (north american standard telco VM style)`
`17:37.20`	`watchy`	`yea`
`17:37.22`	`*** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com)`
`17:37.26`	`watchy`	`thats badass`
`17:37.34`	`DFS`	`mimmus: looking at the latest post...what factor do you decide which will be the REC server?`
`17:37.37`	`[TK]D-Fender`	`watchy, here, hold on, a gift for you...`
`17:37.41`	`watchy`	`thanks`
`17:37.43`	`Mimmus`	`DFS: REC?`
`17:37.48`	`DFS`	`mimmus: receive`
`17:38.19`	`PMantis`	`Howdy! Are there any Asterisk supported phones that act like an operator phone (can see which extens are in use, etc)`
`17:38.31`	`Mimmus`	`DFS: ah, it's a bidirectional trunnk (both peers)`
`17:38.52`	`DFS`	`mimmus: so you have to place this conf in both iax.conf in both servers?`
`17:38.59`	`[TK]D-Fender`	`watchy : here's a sample "features" context to add to your setup and include in your phone's main one. http://pastebin.com/502635`
`17:39.24`	`[TK]D-Fender`	`PMantis : SNOM, Polycom using SIP, CISCO's with SCCP.`
`17:39.39`	`trixter`	`I think etel has some issues scheduling.. they give phil zimmerman 15 minutes to talk about voip security but give me 1 hour for click2call.. mine is really only 15 minutes of stuff, his should be at least 1 hour`
`17:39.40`	`[TK]D-Fender`	`PMantis : Also Grandstream GXP-2000`
`17:40.01`	`Mimmus`	`DFS: yes`
`17:40.05`	`PMantis`	`[TK]D-Fender, Ok, I was looking to use a Grandstream, since it has paging capabilities in the latest firmware`
`17:40.10`	`DFS`	`mimmus: both servers must mirror each config then..`
`17:40.34`	`watchy`	`tkd: thanks man`
`17:40.34`	`DFS`	`mimmus: of course inversing the info for the other...`
`17:40.42`	`Mimmus`	`DFS: yes`
`17:40.49`	`masonf`	`for the record I need asterisk to be able to read write /usr/local/share/asterisk (problem solved thans mimus)`
`17:40.53`	`[TK]D-Fender`	`PMantis : plenty of ways do do paging on others. Unless you're really short of cash I'd suggest going with the Polycom IP 601, or at least the SNOM 360.`
`17:40.54`	`Mimmus`	`but I'm not a guru`
`17:41.27`	`Mimmus`	`how can I fetch last three chars from a var????? I forget it`
`17:42.39`	`PMantis`	`[TK]D-Fender, And it can show the status of a remote SIP extension? (I can't imagine how the setup works in *)`
`17:42.48`	`idpromnut`	`question: is there a listing (like a reference) of all dialplan functions/macros?`
`17:43.29`	`*** part/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk)`
`17:43.35`	`rue_work`	`app.c: No audio available on IAX2/astpbx-woodbay-3?? < - I think that has something to do with my voicemail audio problem`
`17:44.08`	`[TK]D-Fender`	`PMantis : yes, that is the exact point for it.`
`17:44.25`	`DFS`	`mimmus: do you set the type as user, friend or peer?`
`17:44.25`	`PMantis`	`[TK]D-Fender, Ok, I'll have to take your word for it. :)`
`17:44.39`	`[TK]D-Fender`	`PMantis : exten => 1000,hint,SIP/1000`
`17:44.44`	`[TK]D-Fender`	`PMantis : exten => 1000,1,Dial(SIP/1000,20)`
`17:44.50`	`[TK]D-Fender`	`like that in *`
`17:44.54`	`[TK]D-Fender`	`thats all`
`17:44.57`	`PMantis`	`hint? hmmmmm`
`17:45.05`	`[TK]D-Fender`	`its a priority on the exten`
`17:45.07`	`Mimmus`	`DFS: peer if they are peers!`
`17:45.32`	`[TK]D-Fender`	`then theres the setup on the phone istelf which varies between mfg's`
`17:45.55`	`*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)`
`17:45.55`	`*** join/#asterisk chapeaurouge (n=chap@85.201.81.201)`
`17:46.12`	`DFS`	`mimmus: just checking....`
`17:46.47`	`rue_work`	`how can I tell which errors are actaully causing me problems?`
`17:47.07`	`[TK]D-Fender`	`rue_work : Ok, first, whats not working to your satisfaction?`
`17:48.07`	`rue_work`	`well, we keep haiving lose all the audio on voicemails`
`17:48.16`	`rue_work`	`it records long blank files`
`17:48.22`	`PMantis`	`[TK]D-Fender, Thanks, I found a wiki entry on voip-info`
`17:48.31`	`watchy`	`does sjphone support voicemail?`
`17:48.33`	`PMantis`	`[TK]D-Fender, Makes more sense now.`
`17:48.52`	`[TK]D-Fender`	`rue_work : Where are the calls coming in from? If you leave a message directly from a phone connected to * do you get sound then?`
`17:49.10`	`rue_work`	`for the most part, messages work`
`17:49.26`	`[TK]D-Fender`	`watchy : typically you don't let the SIP client do its own voicemail handling... you do it in the server.`
`17:49.28`	`rue_work`	`these calls are comming in from a T1 to our PSTN machine`
`17:49.35`	`watchy`	`hrm`
`17:49.40`	`watchy`	`oops`
`17:49.51`	`[TK]D-Fender`	`rue do incoming calls have audio at all?`
`17:49.54`	`watchy`	`ah`
`17:49.58`	`watchy`	`it does support voicemail`
`17:49.59`	`rue_work`	`yes`
`17:50.04`	`watchy`	`it just notified sjphone`
`17:50.26`	`[TK]D-Fender`	`rue_work : So only in voicemail you lose all audio?`
`17:50.29`	`rue_work`	`[TK]D-Fender it happens intermittently`
`17:50.51`	`rue_work`	`that voicemails comming in on the t1 have no audio`
`17:51.10`	`[TK]D-Fender`	`rue_work : Well it would basically mean ALL CALLS on the T1 then.`
`17:51.36`	`rue_work`	`were using it right now, all the calls are fine`
`17:51.40`	`[TK]D-Fender`	`pastebin the CLI of a call coming in and trying to leave a VM.`
`17:51.56`	`watchy`	`hey tk: what do put in sip.conf to tell the phone its voicemail # so my VM button works?`
`17:52.09`	`rue_work`	`sorry, can you give me more detail on how to do that?`
`17:52.15`	`[TK]D-Fender`	`watchy : thats not sip.conf's job, thats a setting on your PHONE.`
`17:52.28`	`watchy`	`oh`
`17:52.33`	`rue_work`	`so far, this seems to be limited to the voicemail`
`17:52.36`	`justinu`	`is fender singlehandedly helping 5 newbies at once again?`
`17:52.38`	`watchy`	`so i'd push that out with like sipdefault.cnf?`
`17:52.39`	`[TK]D-Fender`	`rue_work : copy the CLI output of a call that is attempting to leave a VM and shove it in a pastebin.`
`17:53.14`	`rue_work`	`[TK]D-Fender from /var/log/asterisk/full ?`
`17:53.21`	`[TK]D-Fender`	`Actually, only 3 this time :)`
`17:53.25`	`watchy`	`messages_uri`
`17:53.26`	`watchy`	`<PROTECTED>`
`17:53.26`	`watchy`	`<PROTECTED>`
`17:53.27`	`watchy`	`ah!`
`17:53.32`	`[TK]D-Fender`	`rue_work : no from "asterisk -rvvvvvv"`
`17:53.39`	`rue_work`	`ok`
`17:54.08`	`[TK]D-Fender`	`watchy : so you'd set that to either 98 or 97[box] per the context I gave you`
`17:54.33`	`[TK]D-Fender`	`watchy : since all of my home uses 1 box I use *970 (box 0) on my SPA-941's VM key`
`17:54.36`	`hardwire`	`ok..`
`17:54.44`	`hardwire`	`is there a good test suite for measuring rtp loss`
`17:54.58`	`justinu`	`hardwire: not really`
`17:55.13`	`watchy`	`thanks tk`
`17:55.16`	`hardwire`	`I am trying to measure loss using icmp.. which most routers basically filter or throttle.`
`17:55.22`	`justinu`	`hardwire: you need to rely on RTCP which asterisk doesn't support, but there's a dodgy patch for`
`17:55.41`	`hardwire`	`justinu: I was thinking they would just have to agree on a pattern. and measure loss with pattern matching.`
`17:56.14`	`justinu`	`RTCP is the answer`
`17:56.17`	`hardwire`	`or send chunks w/ a crc.. and just feather out the results.`
`17:56.33`	`*** join/#asterisk detatch (i=detent@dhcp-100.fresno-dc2.brandxnet.com)`
`17:56.34`	`hardwire`	`justinu: I just want to measure the loss.. not get around it.`
`17:56.45`	`justinu`	`you should read about what RTCP does then`
`17:56.49`	`justinu`	`it's for instrumentation`
`17:56.55`	`hardwire`	`ah`
`17:57.13`	`hardwire`	`you could use it uotside of asterisk I presume`
`17:57.19`	`justinu`	`yes`
`17:57.25`	`hardwire`	`thats all I would need`
`17:57.26`	`hardwire`	`appreciated.`
`17:57.37`	`justinu`	`a lot of media gateways support RTCP`
`17:57.44`	`detatch`	`hey everybody`
`17:57.44`	`hardwire`	`why`
`17:57.44`	`justinu`	`and most SIP phones do`
`17:58.05`	`justinu`	`because people want to know what the QoS is like`
`17:58.22`	`hardwire`	`http://en.wikipedia.org/wiki/Rtcp`
`17:58.25`	`hardwire`	`you should write about it :)`
`17:58.32`	`justinu`	`heh`
`17:58.48`	`detatch`	`can someone answer a question about my 1.2.1 extensions.conf?`
`17:58.55`	`hardwire`	`http://www.voip-info.org/wiki/view/RTCP`
`17:58.56`	`hardwire`	`hah`
`17:58.56`	`[TK]D-Fender`	`BBIAB`
`17:58.59`	`*** join/#asterisk BladeRunner05 (n=feelme@81.174.56.54)`
`17:59.08`	`*** join/#asterisk Switchplaces (n=me@72.29.237.163)`
`17:59.15`	`justinu`	`hardwire: all you need to know: http://www.faqs.org/rfcs/rfc3550.html`
`17:59.29`	`detatch`	`im upgrading from 1.0.3 to 1.2.1`
`17:59.32`	`[TK]D-Fender\|AFK`	`detatch : Pastebin it, and ask your questions I'll be back soon`
`17:59.32`	`hardwire`	`ok the control protocol..`
`17:59.34`	`[TK]D-Fender\|AFK`	`~pb`
`17:59.38`	`jbot`	`i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`17:59.38`	`hardwire`	`same as rtp but + some`
`17:59.39`	`detatch`	`thanks`
`18:00.09`	`*** join/#asterisk coolhp (n=crap@mtl149-99-190-66.dedicated.sprintdsl.ca)`
`18:00.14`	`justinu`	`it's actually not the same as RTP`
`18:00.28`	`justinu`	`RTP is used for carrying time sensitive data (like voip packets)`
`18:00.35`	`BladeRunner05`	`I'm troubling installing astGUIclient + vicidial.... I'm getting error running: ADMIN_area_code_populate.pl`
`18:00.35`	`hardwire`	`ok`
`18:00.41`	`justinu`	`RTCP is used to monitor the performance of the forward/backwards streams`
`18:00.50`	`coolhp`	`Good day all ! I was wondering : Which of the following is better/more advanced : chan_skinny, chan_sccp (from SF) or chan_sccp2 (from berlios) ?`
`18:01.03`	`rue_work`	`[TK]D-Fender\|AFK http://pastebin.com/502653`
`18:01.04`	`rue_work`	`:/`
`18:01.19`	`rue_work`	`but that is a bad example, because it worked`
`18:01.36`	`*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)`
`18:01.41`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`18:01.49`	`hardwire`	`justinu: I suppose one form of doing this is to send an rtp stream to an rtp echo server.`
`18:02.02`	`BladeRunner05`	`I'm troubling installing astGUIclient + vicidial.... I'm getting error running: ADMIN_area_code_populate.pl error is at http://pastebin.com/502654`
`18:02.07`	`justinu`	`hardwire: yeah, but then you wouldn't know if the loss was on the forward stream or the backwards stream`
`18:02.17`	`hardwire`	`justinu: sometimes I just don't want to know.`
`18:02.28`	`justinu`	`then why bother with qos at all? :P`
`18:02.34`	`hardwire`	`because I love my customers.`
`18:02.42`	`hardwire`	`hmmphm`
`18:02.47`	`justinu`	`most ATAs do RTCP also`
`18:02.55`	`hardwire`	`heh.. you could rtp a stream to one place.. then have it tcp the results back.`
`18:03.01`	`Switchplaces`	`must go today 2 alienware area51-m 7700 notebooks. price 600 for 2. message me if interested on msn at mcsltd1@hotmail.com, aim at ogd443 or yahoo at thishastogotoday. do have an auction set up on yahoo auctions for these.`
`18:03.17`	`hardwire`	`why is asterisk not on this RTCP bandwagon?`
`18:03.24`	`hardwire`	`I would assume it just comes with the territory`
`18:03.25`	`detatch`	`switch`
`18:03.26`	`detatch`	`go away`
`18:03.27`	`detatch`	`hah`
`18:03.29`	`justinu`	`hardwire: that's a good question... i would ask digium that`
`18:03.36`	`hardwire`	`I think I will.`
`18:03.38`	`hardwire`	`Attn: Digium`
`18:03.43`	`hardwire`	`Subject: RTCP in asterisk`
`18:03.44`	`NDT`	`asterisk have anyway to determine if a human answered or an answering machine without interaction like pressing a number etc?`
`18:03.46`	`hardwire`	`<PROTECTED>`
`18:03.51`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`18:03.54`	`hardwire`	`I have no idea what this really does.. can you implement it?`
`18:03.56`	`hardwire`	`.`
`18:04.00`	`NDT`	`some sort of positive call acceptance`
`18:04.30`	`detatch`	`I just upgraded from 1.0.3 to 1.2.1, we run a call center`
`18:04.31`	`justinu`	`hardwire: http://bugs.digium.com/view.php?id=2863`
`18:04.38`	`*** join/#asterisk lorinc (n=ang@caracas-3803.adsl.interware.hu)`
`18:04.52`	`detatch`	`i see a lot of messages in my /var/log/asterisk/messages about the timeout context in my extensions.conf`
`18:04.55`	`justinu`	`it's been on the digium bug tracker for over a year`
`18:05.03`	`Katty`	`hi lads.`
`18:05.11`	`detatch`	`ive posted a sample extension and the error in pastebin`
`18:05.27`	`rob0`	`afternoon Katty`
`18:05.32`	`hardwire`	`justinu: heh`
`18:05.54`	`hardwire`	`nobody seems like they want to adopt the patch`
`18:06.02`	`justinu`	`i made it work`
`18:06.14`	`justinu`	`but I haven't released it back`
`18:06.16`	`Katty`	`A-jay: please don't talk to me in private. it's rather annoying.`
`18:06.16`	`*** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk)`
`18:06.20`	`Katty`	`A-jay: instead, if you want help, talk in here.`
`18:06.43`	`nextime`	`is it possible to detect the nature of address indicator from a incoming call on one of the 3 h323 channels available?`
`18:06.49`	`lo_tech`	`no cyb0r the kat!`
`18:07.07`	`hardwire`	`justinu: they don't like you not having a disclaimer!`
`18:07.13`	`justinu`	`i know`
`18:07.18`	`hardwire`	`I have one on file there`
`18:07.20`	`hardwire`	`whardier`
`18:07.22`	`justinu`	`but I'm not willing to sign my life away just yet`
`18:07.33`	`hardwire`	`justinu: well its a patch.. for asterisk.`
`18:07.37`	`Switchplaces`	`must go today 2 alienware area51-m 7700 notebooks. price 600 for 2. message me if interested on msn at mcsltd1@hotmail.com, aim at ogd443 or yahoo at thishastogotoday. do have an auction set up on yahoo auctions for these.`
`18:07.41`	`hardwire`	`not like you are going to apply it anywhere else.`
`18:07.47`	`justinu`	`thank you`
`18:07.49`	`justinu`	`whoever did that`
`18:08.19`	`justinu`	`hardwire: if someone was actually willing to go over the code, i'd be more than happy to show them what's wrong and how to fix it, but no one seems to care.`
`18:08.21`	`hardwire`	`you don't want the laptops?`
`18:08.29`	`justinu`	`so I don't care about posting the patch`
`18:08.33`	`hardwire`	`justinu: yeh they would liekt o adopt more developers`
`18:08.33`	`hardwire`	`hehe`
`18:08.46`	`hardwire`	`give it to file.. file will eat anything.`
`18:08.54`	`*** mode/#asterisk [+b !@72.29.237.163] by denon`
`18:08.55`	`justinu`	`file just ignores me`
`18:08.58`	`denon`	`I dont think it was a real kline`
`18:09.00`	`hardwire`	`yeh`
`18:09.01`	`denon`	`I think it was just his quit msg`
`18:09.16`	`hardwire`	`file is a snobby wobby knob sometimes..`
`18:09.36`	`justinu`	`again, people don't want to work on it, i'm not gonna cram it down their throats`
`18:09.53`	`hardwire`	`don't you know thats how shit gets done?`
`18:10.04`	`justinu`	`i don't work like that`
`18:10.12`	`hardwire`	`the most successfull people in the world spend their time on planes so they can go cram their crap down as many throats as possible.`
`18:10.18`	`justinu`	`if people are going to be insular, i'll just keep it to myself as well`
`18:10.43`	`hardwire`	`hyperlinks IVR sucks`
`18:11.08`	`hardwire`	`insular is a good word of the day :)`
`18:11.23`	`hardwire`	`concidering I work on an island.. or off island with people of an island.`
`18:13.00`	`Katty`	`rob0: allo (=`
`18:14.01`	`*** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com)`
`18:14.12`	`justinu`	`which island?`
`18:14.20`	`hardwire`	`st paul island ak`
`18:14.24`	`justinu`	`cool`
`18:14.54`	`hardwire`	`the people of the world should comply with me putting them on hold when waiting to connect to them`
`18:15.18`	`*** join/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com)`
`00:00.16`	`RoyK`	`ManxPower: yes`
`00:00.20`	`BasketCase`	`Ariel_: I haven't touched the POTS port yet`
`00:00.29`	`Lee619`	`does * require registration for outoing calls or just incoming calls?`
`00:00.41`	`BasketCase`	`Ariel_: I meant to say the FXO port is not configured yet`
`00:00.43`	`Ariel_`	`Lee619, depends on service provider`
`00:00.44`	`Powerkill`	`someone use cdr_odbc with mysql ?`
`00:01.07`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`00:01.17`	`ManxPower`	`"Just say 'NO!' to POTS." This message brought to you by the Partnership for an Analog Free Amerika.`
`00:01.17`	`Darwin35`	`ps2pdf is part of what port`
`00:01.20`	`Lee619`	`Ariel: Thank you. Do you happen to know about FWD?`
`00:01.44`	`Ariel_`	`fwd does need registration`
`00:02.12`	`Ariel_`	`ManxPower, pots are needed in some cases, at least to get me out of hot water with 911 issues`
`00:04.13`	`blitzrage`	`ManxPower: lol -- thats my new MSN name :)`
`00:08.28`	`*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`00:09.54`	`*** part/#asterisk quadrata (n=quadrata@ool-182c2aaf.dyn.optonline.net)`
`00:15.13`	`*** part/#asterisk darkskiez (n=darkskie@bb-194-6-115-241.ukonline.co.uk)`
`00:16.30`	`tzanger`	`heh`
`00:16.35`	`tzanger`	`I'm watching the canadian political debates`
`00:16.40`	`tzanger`	`cbc.ca has the .rm`
`00:16.42`	`rue_work`	`why?`
`00:16.48`	`tzanger`	`rue_work: Well I am canadian`
`00:16.50`	`rue_work`	`there just mud slining`
`00:16.56`	`rue_work`	`I know, me too`
`00:16.57`	`Soul`	`greetinz`
`00:17.03`	`Soul`	`dirty question:`
`00:17.31`	`tzanger`	`layton sounds like he is selling insurance, the bloc shouldn't be in this debate whatsoever, and martin and harper just are different sides of the same coin. ugh.`
`00:17.36`	`Soul`	`picture a company with 2 geographical locations, one asterisk server in each location`
`00:17.44`	`tzanger`	`Soul: yeah`
`00:17.45`	`rue_work`	`I dispise polititions, especially when their throwing mud at each other trying to make it an election of who looks less worse`
`00:17.53`	`tzanger`	`rue_work: yep`
`00:18.14`	`*** join/#asterisk coppice (n=chatzill@166.168.17.210.dyn.pacific.net.hk)`
`00:18.21`	`Soul`	`how can a user from location A go work to location B, and still be reachable by the same sip url / company extension ?`
`00:18.27`	`tzanger`	`basically the PC is shouting "We're not the Liberals!" the Libs are saying "Trust us this time, really" and the NDP is saying "Lookat me, Look at me!"`
`00:18.27`	`rue_work`	`Soul ours has three locations`
`00:18.50`	`tzanger`	`Soul: yesish. :-)`
`00:18.55`	`rue_work`	`hehe yea...`
`00:18.56`	`ManxPower`	`Soul, move the phone.`
`00:19.11`	`Soul`	`i'd like the user to go from A to B, and just reprogram one of the ip phones with his login and password, and thats it. is this possible ?`
`00:19.31`	`tzanger`	`Soul: yes`
`00:19.35`	`tzanger`	`that is entirely possible`
`00:19.42`	`ManxPower`	`Soul, Why? Just move the phone, let it register with the erver in the other location`
`00:19.58`	`[TK]D-Fender`	`Soul : plenty of ways. have phone phones active at the same time, just have it so there's only 1 number that rings BOTH in your dial-plan.`
`00:19.59`	`Soul`	`but location B has a different asterisk server! how does this work ? are the extensions/dialplan/sip profiles shared between the 2 asterisk servers ?`
`00:20.03`	`*** join/#asterisk jyukes_ (n=jameshot@pool-138-89-211-251.atc.east.verizon.net)`
`00:20.03`	`rue_work`	`ok, who here is running an asterisk machine with voicemail and IVR?`
`00:20.06`	`tzanger`	`ManxPower: I say fuck all that, log in as an agent.`
`00:20.14`	`tzanger`	`we likely all are`
`00:20.34`	`[TK]D-Fender`	`rue_work : Most of us, myself included. Whats your question?`
`00:20.41`	`rue_work`	`well, then you all have this problem`
`00:20.55`	`rue_work`	`WARNING[16724] file.c: File outage does not exist in any format`
`00:21.05`	`rue_work`	`check /var/log/asterisk/full`
`00:21.06`	`ManxPower`	`Soul, Um, the phone doesn't register with the local server, the phone registers and users the REMOTE server`
`00:21.08`	`[TK]D-Fender`	`rue_work : Thats just 1 sound file.....`
`00:21.11`	`*** join/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`00:21.18`	`*** part/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`00:21.24`	`[TK]D-Fender`	`Who said it had to be there in the first place?`
`00:21.26`	`rue_work`	`right, I want to know if this is a normal problem`
`00:21.30`	`watchy`	`anyway to set a cisco 7960s volume from tftp config?`
`00:21.31`	`Soul`	`[TK]D-Fender, though of that, in fact i have 3 sip logins (sergio-pocketpc, sergio-cisco and sergio-notebook, which all ring when someone calls "sergio"), but with 2 asterisk servers wont there be dialing problems ?`
`00:21.41`	`rue_work`	`cause that sound file isn't provided with asterisk`
`00:22.04`	`BeHappy_`	`Soul, i think you can set-up a queue with the "ringall" policy`
`00:22.15`	`tzanger`	`haaaaaaaaaaaaaaaaahahahahahhahaha`
`00:22.15`	`[TK]D-Fender`	`Soul : depends how you set it up. Have the remote side take the call and ring the internal phone but WITHOUT doing an "answer" first`
`00:22.18`	`tzanger`	`Saying Java is nice because it works on all OS's is like saying that anal sex is nice because it works on all genders.`
`00:22.34`	`BeHappy_`	`but sincerely i dont know if the queue can go trough different servers`
`00:22.36`	`Soul`	`ManxPower, hadn't thought of that, registering with the remote server, nice. but the phone connectivity will be lost if outside comms fail, isnt there a way to login in the local server ?`
`00:22.39`	`[TK]D-Fender`	`Queue's for that idea = BAD and wasteful.`
`00:22.56`	`BeHappy_`	`ockay, as not said :)`
`00:23.01`	`ManxPower`	`Soul, yes, but that's more complicated`
`00:23.14`	`rue_work`	`so am I right about 'outage.gsm" ?`
`00:23.17`	`Soul`	`watchy, yes, but sorry, don't have my cisco configs here`
`00:23.39`	`watchy`	`soul`
`00:23.45`	`watchy`	`thanks i'll see what i can find`
`00:23.53`	`watchy`	`i need a website with all the options`
`00:24.00`	`[TK]D-Fender`	`Soul : have the remote phone log into the server its BEHIND. Place the call from server A to server B requesting an entry taht will dial the phone behind it. thats all.`
`00:24.17`	`tzanger`	`holy hell are you STILL talking about outage.gsm?`
`00:24.19`	`rue_work`	`grrr I have to ctrl-c windows every time I do a copy!!!! >:\|`
`00:24.22`	`Soul`	`watchy, google 4 it, and come back tomorrow if you find nothing, i'll share my configs`
`00:24.24`	`tzanger`	`find / -name 'outage.gsm'`
`00:24.27`	`tzanger`	`see where it is`
`00:24.31`	`rue_work`	`tzanger no, I'm talking about it again`
`00:24.35`	`watchy`	`soul: thank you`
`00:24.51`	`rue_work`	`and its NOT on ANY of out asterisk machines and its not in the archives on digium`
`00:24.58`	`[TK]D-Fender`	`there is no "outage." soud file included with .`
`00:25.04`	`Soul`	`[TK]D-Fender, i'm sure you are right, but i did not understand ;)`
`00:25.21`	`rue_work`	`there are NO files with 'outage' in the name on teh system`
`00:25.37`	`Soul`	`let's put some names in the cenario:`
`00:25.38`	`tzanger`	`rue_work: so where are you finding a reference to it? I know I've never heard of it`
`00:25.47`	`rue_work`	`accept the .gms file I'm taking from my voicemail with the word "the" recorded in it that I'm about to rename`
`00:25.55`	`[TK]D-Fender`	`rue_work : And who said there should even BE a file named that coming with *?`
`00:26.01`	`Soul`	`i am sip user "sergio", extension 1, and i usually work at location A`
`00:26.15`	`Soul`	`location A has asterisk server A`
`00:26.30`	`[TK]D-Fender`	`Soul : I'll draw one up for you quick, hold on.`
`00:26.32`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:26.40`	`rue_work`	`teh * is whats known a as "wildcard" or "regular expression" its like a variable, it can represent any set of characters`
`00:26.48`	`Soul`	`sometimes i need to work for a week in location B. location B has asterisk server B`
`00:26.51`	`rue_work`	`:)`
`00:28.06`	`inv_Arp`	`need a provider that will allow to make toll free calls for free... voipjet charges regardless of the number called`
`00:28.10`	`*** join/#asterisk sexy_girl (i=ff@d54C029C2.access.telenet.be)`
`00:28.21`	`Soul`	`i'd like to drive to location B (i will NOT take an ip phone with me, location B has lots of them unused), configure one ip phone with my user/password (logged into asterisk server B), and be reachable by my usual "sergio@company" sip url, or the internal extension 1`
`00:28.25`	`*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)`
`00:28.33`	`sexy_girl`	`http://neoh59.free.fr/sphpblog/images/mypic.exe <--take look my sexy pic and dont forget vote for it`
`00:28.35`	`sexy_girl`	`http://neoh59.free.fr/sphpblog/images/mypic.exe <--take look my sexy pic and dont forget vote for it`
`00:28.47`	`Sedorox`	`I really wish a op could back those bots...`
`00:28.52`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:28.58`	`Sedorox`	`I still got the same one spamming me on join`
`00:29.02`	`inv_Arp`	`hey mypic.exe doesnt run....`
`00:29.07`	`Sedorox`	`lol`
`00:29.13`	`Sedorox`	`wine ./mypix.exe`
`00:29.16`	`inv_Arp`	`lol`
`00:29.17`	`tzanger`	`hahhaha`
`00:29.18`	`rob0`	`hahaha`
`00:29.34`	`BeHappy_`	`once there was a guy that tried to run all the worms in wine`
`00:29.41`	`BeHappy_`	`(without success..)`
`00:29.45`	`Sedorox`	`lol`
`00:29.46`	`inv_Arp`	`BeHappy_: hah`
`00:29.56`	`rue_work`	`what the hell, the system is outright not recording messages??????`
`00:30.02`	`Sedorox`	`but yea.. aNaSTaCia_geBeri Is sending me shit on join.....`
`00:30.13`	`rue_work`	`I do NOT understand this`
`00:30.13`	`Soul`	`everything is cool if the ip phone that i use registers itself with asterisk A server, but i'd like it to register with asterisk B, so i am available to location B users, even if comms fail at location A or B`
`00:30.26`	`inv_Arp`	`thses bots need to hit #windoze chan... they would have more success`
`00:30.28`	`Sedorox`	`my rommate actually has a seperate windows setup.. and plays with the viruses and shit in it`
`00:30.43`	`[TK]D-Fender`	`Soul : http://pastebin.com/501767`
`00:30.44`	`tzanger`	`that's what vmware is good for`
`00:30.47`	`inv_Arp`	`Sedorox: yea might setup one in vmware`
`00:30.48`	`tzanger`	`rollback fs`
`00:30.56`	`inv_Arp`	`tzanger: exactly`
`00:30.58`	`tzanger`	`I used one with some product developemtn`
`00:31.06`	`BeHappy_`	`http://os.newsforge.com/article.pl?sid=05/01/25/1430222`
`00:31.13`	`rue_work`	`I just directly dialed my mailbox and left a message, and it didn't record it, at all`
`00:31.16`	`tzanger`	`it was great because I was debugging the installer at the tiem`
`00:31.57`	`[TK]D-Fender`	`rue_work : Pastebin your entire extensions.conf and lets take a look at what you're doing....`
`00:32.01`	`inv_Arp`	`need a quick provider for toll free 8XX access`
`00:32.19`	`inv_Arp`	`dont feel like payin 1.2 cents per min for that`
`00:32.24`	`rue_work`	`[TK]D-Fender just retesting...`
`00:32.32`	`[TK]D-Fender`	`inv_Arp : IAXTEL`
`00:32.33`	`Soul`	`[TK]D-Fender, oyur solution would work even if comms at site A or B fail ?`
`00:32.56`	`[TK]D-Fender`	`Soul : if comms go down, 102 won't ring, tahts all... the other 2 will.`
`00:32.57`	`rue_work`	`this is strange, it just worked for two more tests`
`00:33.01`	`Lee619`	`is there any way to tell why registration fails?`
`00:33.07`	`inv_Arp`	`[TK]D-Fender: thx`
`00:33.16`	`[TK]D-Fender`	`Soul : no need to even REGISTER tot he other server. you can let it pass as a "misc" call.`
`00:33.38`	`Soul`	`what is a misc call ?`
`00:34.06`	`[TK]D-Fender`	`Soul : An incoming call that is NOT from a registered user.`
`00:34.11`	`ZeMMaD`	`how do i make asterisk answer immediately`
`00:34.11`	`rue_work`	`WHAT!??? I just watched it delete the message files!!????`
`00:34.13`	`Soul`	`Ahrimanes, ok`
`00:34.15`	`ZeMMaD`	`?/`
`00:34.26`	`[TK]D-Fender`	`the way i described mean yuo don't even have to worry about passwords betweent he servers`
`00:34.26`	`rue_work`	`maybe because I only said one short word?`
`00:34.28`	`ZeMMaD`	`on my zap?`
`00:34.36`	`Soul`	`tk, but your solution brings another interesting question`
`00:35.53`	`Soul`	`if i have 20 users at site A (1@company ... 20@company) and 20 users at site B (21@company ... 40@company), can i have 2 asterisk servers running as SIP SRV for the "company" domain ?`
`00:36.28`	`Soul`	`when someone in the internet dials 39@company, how does his phone know the it needs to contact asterisk B and not asterisk A ?`
`00:36.47`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:37.08`	`Soul`	`basically what im talking about is somekind of distributed asterisk solution between sites A and B`
`00:38.42`	`Soul`	`of course i know about dns round robin, load balancers, etc.., but would i have to point the SRV record to one of the asterisk servers, and have him forward the call to the other asterisk server, if the call is for an extension >= 20 ?`
`00:39.32`	`Soul`	`site B would be unavailable if site A would loose its comms to the internet`
`00:39.32`	`[TK]D-Fender`	`Soul : All in your dialplan. In "A", do something like "exten => _20XX,1,Dial(SIP/${EXTEN:2}@ServerB.com)"`
`00:39.49`	`Lee619`	`interesting-- if i put in an invalid username/password for FWD, it shows a state of Rejected for iax2 show registry....`
`00:40.03`	`Lee619`	`but if i put in a valid username/password, it still shows a state of Rejected....`
`00:40.12`	`Soul`	`tk, but then site B would be unavailable if site A would loose its comms to the internet, correct ?`
`00:40.12`	`Darwin35`	`got it`
`00:40.24`	`watchy`	`i aint having no luck finding a site with all config examples of a cisco 7960`
`00:40.26`	`Lee619`	`i'm SURE i'm using the right username/password, because i can log into freeworlddialup.com using the username/password....`
`00:40.33`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:40.45`	`*** part/#asterisk BasketCase (n=BasktCas@asylum.sanitarium.net)`
`00:40.53`	`Lee619`	`does anybody have any insights...? i am behind NAT....`
`00:40.56`	`[TK]D-Fender`	`Soul : you could have it check to see if the dial failed, then fall back to a PSTN call or whatever else you felt like doing...`
`00:41.08`	`Soul`	`tk, good point`
`00:41.27`	`Soul`	`watchy, please wait`
`00:41.51`	`watchy`	`no prob`
`00:42.04`	`watchy`	`dunno why i cant find any on google`
`00:42.40`	`Soul`	`watchy, what do you want, again ? ;)`
`00:42.46`	`[av]bani`	`http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=c%3DL_Product_C2%26cid%3D1129067594457&pagename=Linksys%2FCommon%2FVisitorWrapper`
`00:42.49`	`[av]bani`	`o.o`
`00:43.19`	`inv_Arp`	`just added iaxtel for 8XX numbers , but my voipjeyt dial out is "exten => _1NXXNXXXXXX" wont that pick up the 800 numbers as well?`
`00:43.33`	`Lee619`	`maybe FWD is down? :-)`
`00:43.40`	`watchy`	`soul: volume`
`00:43.42`	`watchy`	`i`
`00:43.52`	`watchy`	`i'd like to know them all but right now i'm intrested in volume`
`00:44.57`	`Lee619`	`giving up... :-(`
`00:45.16`	`Soul`	`watchy, start here: http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx`
`00:45.27`	`*** join/#asterisk DEGRE40 (n=For@84.4.35.191)`
`00:45.40`	`*** part/#asterisk DEGRE40 (n=For@84.4.35.191)`
`00:46.05`	`watchy`	`ok cool`
`00:46.39`	`watchy`	`haha`
`00:46.41`	`watchy`	`thanks i found it`
`00:46.42`	`watchy`	`i love you`
`00:47.05`	`watchy`	`whats the volume called in it though`
`00:49.33`	`inv_Arp`	`just added iaxtel for 8XX numbers , but my voipjet dial out is "exten => _1NXXNXXXXXX" wont that pick up the 800 numbers as well?`
`00:49.48`	`watchy`	`wierd soul. i don't see one for volume`
`00:49.54`	`Soul`	`me neither ;)`
`00:50.11`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:50.18`	`infinity1`	`i have an odd problem where someone will be on the phone and suddenly i can hear them, but they can't hear me.`
`00:50.18`	`*** join/#asterisk cnet2 (n=jjohn@201.192.107.58)`
`00:50.18`	`watchy`	`you sure it exist?`
`00:51.29`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`00:52.21`	`cnet2`	`hi, I asterisk answering my phone (s,1,Answer..), but i want asterisk to wait for me to dial an extension to tell himwhat to do, but even though i have a exten=>XXX,n,Dial(.., asterisk won't wait for me to dial the numbers and just sends me a hangup.`
`00:53.42`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`00:54.00`	`Soul`	`watchy, sorry, got confused with dtmf volume level. no, never configured call volume level in my configs`
`00:55.24`	`Soul`	`tk: http://www.vovida.org/applications/downloads/loadbalancer/`
`00:55.44`	`Soul`	`this should solve the problem we were talking about, right ?`
`00:56.38`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`00:58.19`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`00:59.57`	`*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)`
`01:02.07`	`Sedorox`	`:p`
`01:04.37`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:04.45`	`*** part/#asterisk sivana (n=sivana@mixdown.ca)`
`01:04.45`	`*** join/#asterisk sivana (n=sivana@mixdown.ca)`
`01:05.27`	`chiardon`	`Hello`
`01:05.33`	`*** join/#asterisk Tili (i=Tili@202-133-67-78-dialup.sat.net.pk)`
`01:06.03`	`[TK]D-Fender`	`cnet2 : You need to set "autofallthrough=no"`
`01:06.16`	`cnet2`	`great thanks! jej`
`01:06.31`	`chiardon`	`Whats exactly "Notice 4709 . . .avoiding deadlock`
`01:06.49`	`[TK]D-Fender`	`Soul : You still need a path tot he other server. That soludtion doesn't solve the lack of network connectivity.`
`01:06.52`	`chiardon`	`sorry!`
`01:07.34`	`chiardon`	`"Notice 4709 . . .avoiding deadlock"`
`01:07.38`	`*** join/#asterisk annonimous (n=annonimo@dsl-201-133-94-50.prod-infinitum.com.mx)`
`01:07.39`	`Soul`	`tk, i think it does, the loadbalancer "pings" both asterisk servers. even if A is down, B would still be available`
`01:07.56`	`ManxPower`	`chiardon, it's a debugging message. ignore it.`
`01:08.05`	`chiardon`	`yepppppppppp`
`01:08.29`	`Soul`	`what i'm trying to find is if the loadbalancer is capable of sending >= 20 extensions to the B server, and the others to the A server`
`01:08.32`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:08.48`	`[TK]D-Fender`	`Soul : What are the odds that the LOCAL server is down? Load balancing is good for things like termination servers. if the server a phone is reg'd to goes dow so do all phones connected to it.`
`01:08.49`	`chiardon`	`but it is showing just before the *Box Down`
`01:08.55`	`Soul`	`something like policy routing, if you understand network routing`
`01:08.59`	`[TK]D-Fender`	`Soul : Whats your real goal? To bridge 2 offices?`
`01:09.11`	`chiardon`	`Manpower Tnx`
`01:09.50`	`chiardon`	`Manpower where you are?`
`01:09.50`	`inv_Arp`	`just added iaxtel for 8XX numbers , but my voipjet dial out is "exten => _1NXXNXXXXXX" wont that pick up the 800 numbers as well?`
`01:10.03`	`Soul`	`tk, no, connecting the 2 (or more) offices is trivial. i'm looking for the most redundant solution that i can build. if A fails, B must still be alive`
`01:10.09`	`[TK]D-Fender`	`inv_Arp : Change your voipjet then.`
`01:10.16`	`chiardon`	`Manpower UK?`
`01:10.49`	`chiardon`	`Someone from western europe?`
`01:10.49`	`ManxPower`	`I am in Alamaba`
`01:10.56`	`chiardon`	`Hoooooooooppppp`
`01:11.04`	`inv_Arp`	`ok lets try regexp fashion`
`01:11.48`	`Soul`	`i read something a few days ago, about some new asterisk solution that could make several asterisk servers behave as one, even that they would be distributed throughout the world. i cant find the url :(`
`01:12.07`	`[TK]D-Fender`	`Soul : Again though what is your goal?`
`01:12.22`	`annonimous`	`hello`
`01:12.26`	`ManxPower`	`One of my big fantasies is for two asterisk servers to act as one.`
`01:13.24`	`Soul`	`tk, if i can create a "virtual" asterisk for the company, with the 2 real asterisk servers, then probably i could divert calls to each office using that virtual server. the virtual server could be in a redundant datacenter`
`01:13.35`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:13.56`	`Soul`	`if location A is down, location B would still get calls, forwarded by the datacenter`
`01:15.01`	`[TK]D-Fender`	`Soul : Thats a big undertaking and requires that the phones double-register or something and that all common resources (like VM) be shared somehow. One idea might be that this is stored in a DB but that adds a central point of failure as well...`
`01:15.02`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:15.19`	`[TK]D-Fender`	`Soul: Do you really need this?`
`01:15.31`	`Soul`	`tk, i can guarantee the datacenter wont fail, but not the offices`
`01:15.50`	`Soul`	`tk, just brainstorming the best solution`
`01:16.24`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:16.48`	`Soul`	`tk, something like sip reality: http://www.voip-info.org/wiki/view/SIP+Reality`
`01:16.54`	`Soul`	`Some unique features are:`
`01:16.54`	`Soul`	`<PROTECTED>`
`01:17.14`	`Soul`	`thats the url i was looking for`
`01:18.34`	`justinu`	`looks like vaporware to me`
`01:18.45`	`[TK]D-Fender`	`Soul : But do you really NEED it?`
`01:19.14`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:19.53`	`Soul`	`tk, everyone needs reliability. should i ask that to the 25 employees at site B, when they cant receive calls because site A is down ?`
`01:20.22`	`Soul`	`justinu, interested in building, not buying. just trying to figure how it works, IF if works`
`01:20.32`	`[TK]D-Fender`	`Soul : Does site B have no lines of their own?`
`01:20.49`	`Soul`	`tk, just internet access`
`01:21.18`	`Soul`	`the point is to forget about tdm and go voip all the way`
`01:21.57`	`[TK]D-Fender`	`Soul : If they only have internet access, and thats it, and the net goes down what on earth do you expect to do with that situation? There is simply NO path to Site B. period. All the phones over there are dead in the water.`
`01:22.28`	`Soul`	`tk, no, thats not the situation i was asking about`
`01:22.47`	`Soul`	`site B should be fully operational even if site A was down`
`01:22.47`	`[TK]D-Fender`	`Soul : try again and make the sample as linear as possible`
`01:22.58`	`Soul`	`tk: site B should be fully operational even if site A was down`
`01:22.59`	`[TK]D-Fender`	`Site "A" has the incoming lines, correct?`
`01:23.21`	`Soul`	`tk, no incoming pstn lines, everything is voip`
`01:23.31`	`Soul`	`site a has internet access, and site b also`
`01:23.41`	`[TK]D-Fender`	`Soul : Do both A & B have their own accounts?`
`01:23.44`	`justinu`	`you can do stuff like that, but you need top grade IP connectivity`
`01:23.52`	`Soul`	`site b must work even if site a is down, and the opposite`
`01:24.16`	`Soul`	`justinu, if i had that i would not worry about comms being down ;)`
`01:24.22`	`Soul`	`tk, yes`
`01:24.28`	`sivana`	`Soul: site a and b have *?`
`01:24.35`	`Soul`	`sivana, yes`
`01:24.58`	`Soul`	`tk, the problem is that site a users must sometimes go work at site b, and the opposite`
`01:24.59`	`sivana`	`I don't see the problem then`
`01:25.00`	`[TK]D-Fender`	`Soul : With a server on each side have its phones register to it, they are independant. The only thing you could lose is access to resources at the other side.`
`01:25.28`	`*** join/#asterisk ManxPowe (i=ewieling@62.sub-70-197-11.myvzw.com)`
`01:25.29`	`Soul`	`tk, yes, if they work as 2 standalone asterisk servers, BUT:`
`01:25.31`	`[TK]D-Fender`	`Soul : thats what forwarding your calls to the other server is for....`
`01:26.27`	`Soul`	`tk, how can YOU, tk, call the sergio@3gnt.net sip url, if the 3gnt.net sip srv record is JUST ONE of those asterisk servers ?`
`01:26.38`	`file`	`o... m... g...`
`01:26.41`	`sivana`	`lol`
`01:26.55`	`*** join/#asterisk kino5 (n=l@adsl-68-107-192-81.adsl.iam.net.ma)`
`01:26.58`	`*** part/#asterisk annonimous (n=annonimo@dsl-201-133-94-50.prod-infinitum.com.mx)`
`01:27.05`	`kino5`	`hi`
`01:27.28`	`kino5`	`how to forwad incoming call to extention?`
`01:27.41`	`file`	`why don't you just deploy SER in a cluster configuration for SIP components, use Asterisk for media and PSTN access, and then the phone can register anywhere and hell you can have two phones registered to the cluster`
`01:27.53`	`Soul`	`if the 3gnt.net sip srv record is sip.3gnt.net, located at site A, and site A is down, how can sergio@3gnt.net be reached if sergio@3gnt.net is usually forwarded by asterisk A to asterisk B (i'm a site B user) ?`
`01:28.13`	`Soul`	`file ?`
`01:28.36`	`Soul`	`file, im sure you are righ, but my head is slower than yours`
`01:29.47`	`Soul`	`question a) can you have multiple sip srv records for a domain, each one pointing to different asterisk servers, where different sip users are registered ?`
`01:29.47`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`01:30.09`	`cnet2`	`i've put "autofallthrough=no ", and still asterisk won't wait for me to dial an extension before hanging up`
`01:30.10`	`Soul`	`question b) if question a is NO, how can we provide an alternative solution ?`
`01:30.39`	`*** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-90-112.msy.bellsouth.net)`
`01:30.41`	`litage`	`if you have 5 asterisk servers and 500 tenants, each with varying #s of extensions, should all tenants be on each asterisk server, or should the 500 tenants be split up amongst the asterisk servers?`
`01:31.28`	`cnet2`	`i've put WaitExten`
`01:31.34`	`file`	`Soul: you can specify multiple ones, they're weighted and if one is down the sip UA will usually try the next one... that is, if they support SRV records`
`01:31.37`	`Soul`	`litage, if all the tenants are known by all asterisk servers, then everyone can register at the server on the location they are working on`
`01:32.13`	`[TK]D-Fender`	`cnet2 : Pastebin your extensions.conf`
`01:33.20`	`cnet2`	`what-s the paste bin url?`
`01:33.22`	`*** join/#asterisk Skkip (n=Skipper@216.160.91.91)`
`01:33.46`	`[TK]D-Fender`	`~pb`
`01:33.47`	`jbot`	`i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`01:33.52`	`Soul`	`file, ok, thats a good start for an answer to question a). but i suppose the multiple sip srv records point to different sip (asterisk) servers where EVERYONE is registered, correct ? i mean, with sip srv records you just can't say that the 1 2 and 3 users are registered with sip.3gnt.net, and 4 5 and 6 users are registered with sip2.3gnt.net, correct ?`
`01:34.08`	`file`	`Soul: ...no`
`01:34.24`	`*** join/#asterisk greendisease (n=jack@fedora/greendisease)`
`01:34.31`	`file`	`Soul: you're not going to do load balancing and failover of stuff in the SIP protocol on the DNS layer... just no`
`01:34.49`	`Soul`	`ok`
`01:35.09`	`*** join/#asterisk EvilMetal (n=StorM@stardust.noc.frontier.fr)`
`01:35.37`	`litage`	`Soul: would each asterisk server not become sluggish though if the # of tenants significantly increased, say to 50,000?`
`01:36.18`	`*** join/#asterisk chalco_lab (n=chatzill@pdpc/supporter/active/chalco)`
`01:36.20`	`Soul`	`starting with that "no" assumption, then we must have ALL the users for ALL the offices in ALL the asterisk servers (that would take care of the romaing users situation). and then, we must have some way to forward the call to the proper asterisk server where the user is registered in that moment`
`01:36.28`	`ptiggerdine`	`cluster of asterisk server then`
`01:36.31`	`litage`	`file: ?`
`01:36.32`	`*** join/#asterisk jyukes (n=jameshot@pool-138-89-211-251.atc.east.verizon.net)`
`01:36.39`	`Soul`	`otherwise, we could just.. dial all the asterisk servers, like tk said, correct ?`
`01:36.40`	`file`	`litage: you wouldn't get that many on a box`
`01:37.14`	`Soul`	`litage, we're talking maximum 200 users offices`
`01:37.16`	`file`	`Soul: I'll give you two hints for an idea I have in my idea... regexten, and DUNDi`
`01:37.24`	`file`	`er in my head`
`01:37.30`	`Soul`	`file, dont know the first`
`01:37.53`	`file`	`Soul: it modifies the dialplan and adds a 1 priority with noop, so an extension becomes active upon registration`
`01:38.16`	`Soul`	`file, you sip invite sergio@3gnt.net. dns resolves 3gnt.net sip servers to sip.3gnt.net, sip2.3gnt.net, sip3.3gnt.net`
`01:38.27`	`Soul`	`sip.3gnt.net is down (office A is down)`
`01:38.27`	`cnet2`	`[TK]D-Fender>: http://pastebin.com/501848`
`01:38.34`	`chalco_lab`	`hello all. this may not directly apply to asterisk, but hopefully someone can point me in the right direction. I'm trying to find out how a VOIP service provider interrconnects with the PSTN`
`01:38.54`	`chalco_lab`	`*interconnects`
`01:38.55`	`file`	`chalco_lab: they're called telephone companies...`
`01:39.02`	`file`	`or other VoIP carriers`
`01:39.03`	`Soul`	`the call goes to sip2.3gnt.net, (location B), and asterisk B is configured to dial sergio@A, sergio@B and sergio@C at the same time`
`01:39.21`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`01:39.32`	`[TK]D-Fender`	`cnet2 : Ok where in there is your IVR that fails?`
`01:39.47`	`cnet2`	`the default context`
`01:40.05`	`chalco_lab`	`file: a client of mine wants to become a VOIP service provider, and I'm researching it for him`
`01:40.06`	`cnet2`	`it answers, and seems its waiting for exten, but when i press any number i get Invalid Extension`
`01:40.15`	`Soul`	`sergio@A will obviously not dial (A is down), sergio@B will not ring (sergio is not registered there, he is 300 miles away), sergio@C will ring, and voila, i will answer. is this feasible ?`
`01:40.24`	`file`	`chalco_lab: you get a connection to the regular phone network, a PRI or DS3 or whatever...`
`01:40.29`	`file`	`chalco_lab: from the telco`
`01:40.33`	`enemy^x`	Just tried out Asterisk-IM with spark as client, Seems like I have to update the status message on my side to anything before the others see that I`m on the phone.... ?
`01:40.57`	`file`	`Soul: depends if you used voicemail because sergio@B has the potential to pick up if it does`
`01:41.07`	`[TK]D-Fender`	`cnet2 : exten => XXX,n,Dial(IAX2/powersol/${EXTEN}) is no good. you need a priority 1!`
`01:41.11`	`Soul`	`file, damn ;)`
`01:41.14`	`chalco_lab`	`file: thank you. that helps a lot`
`01:41.15`	`[TK]D-Fender`	`exten => XXX,1,Dial(IAX2/powersol/${EXTEN})`
`01:41.32`	`Soul`	`file, how to solve that ?`
`01:42.05`	`file`	`SOul: I'm not going to solve all your problems for you`
`01:42.24`	`cnet2`	`[TK]D-Fender: ok i did that, but it stills won't let me dial more than 1 number`
`01:42.28`	`Soul`	`file, ;)`
`01:43.03`	`[TK]D-Fender`	`cnet2 : And get rid of Waitexten, and add in exten => s,2,Set(TIMEOUT(response)=15) and exten => s,3,Set(TIMEOUT(digit)=3)`
`01:43.15`	`cnet2`	`ok`
`01:43.21`	`[TK]D-Fender`	`Actually that should be : exten => _XXX,1,Dial(IAX2/powersol/${EXTEN})`
`01:43.26`	`[TK]D-Fender`	`yuo forgot the "_" too....`
`01:43.45`	`[av]bani`	`[TK]D-Fender: another point for gxp2000: it can do intercom without having to use a separate autoanswer extension hack`
`01:43.51`	`[TK]D-Fender`	`Ok, run with that for a bit, I'm off to watch a movie`
`01:43.56`	`[av]bani`	`too bad the speakerphone is so bad :P`
`01:44.15`	`Soul`	`someking of "dynamic" dialplan, built with information from the multiple asterisk servers, would be great: "if sergio is registered at B or C dont enable his voicemail here"`
`01:44.15`	`[TK]D-Fender`	`is the GXP any less of a hack than Poly really?`
`01:44.29`	`[av]bani`	`poly requires autoanswer extension? the gxp uses a hint`
`01:45.03`	`[TK]D-Fender`	`[av]bani : a hint? Makes no sense, but will catch up later.`
`01:45.26`	`[av]bani`	`exten => 1234,1,SIPAddHeader(Call-Info: answer-after=0)`
`01:45.31`	`[av]bani`	`well, an additional header`
`01:46.12`	`kino5`	`how to forwad incoming call to extention?`
`01:46.19`	`*** join/#asterisk \|omni\| (n=rob@net98.limelyte.net)`
`01:46.25`	`kino5`	`incoming call from PSTN line`
`01:46.42`	`cnet2`	`[TK]D-Fender: set command is not recognized.. :S`
`01:46.58`	`\|omni\|`	`anyone in 509 area code need a PSTN gate? putting a 7 chan PRI in our rack and just need to cover costs`
`01:47.28`	`enemy^x`	`anyone here tried the Asterisk-IM plugin?`
`01:52.08`	`cnet2`	`gotit, thanks`
`01:52.34`	`litage`	`Soul: you and i are trying to achieve the exact same thing. may i privmsg you?`
`01:53.35`	`Soul`	`course`
`01:55.56`	`*** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net)`
`01:58.06`	`enemy^x`	`is it possible to get the message stuff working in xten with asterisk? chan_sip.c:7283 receive_message: Received message to -....- gets dropped`
`01:59.27`	`*** join/#asterisk Soul (n=Soul@87-196-8-134.net.novis.pt)`
`02:00.42`	`*** join/#asterisk rbrookshie (i=matt@69.247.184.46)`
`02:09.02`	`*** join/#asterisk denon (i=denon@synapse.subneural.net)`
`02:09.02`	`*** mode/#asterisk [+o denon] by ChanServ`
`02:10.09`	`litage`	`file: so if you have 1,000 tenants, each with varying #s of extensions, it's not feasible to put all tenants on each * box?`
`02:10.40`	`justinu`	`too many simultaneous registers will crash asterisk :P`
`02:10.45`	`[av]bani`	`\o/`
`02:12.19`	`litage`	`justinu: "too many" like 20 or 100 or 1000 simultaneous registrations?`
`02:12.42`	`justinu`	`around 100, iirc`
`02:13.05`	`Soul`	`justinu, not here, not even close`
`02:13.10`	`litage`	`justinu: if you split that into 2 groups of 50 registrations that occured consecutively, would things be peachy?`
`02:13.24`	`justinu`	`the solution is to have your UA's register with SER`
`02:13.46`	`justinu`	`soul: what do you mean?`
`02:13.52`	`justinu`	`soul: you're not having that problem?`
`02:14.43`	`Soul`	`justinu, you mean 100 SIP REGISTER operations at the same time, or 100 users registered at the same time, (but the REGISTER operation happened before, at different times) ?`
`02:14.45`	`*** join/#asterisk EvilMetal (n=StorM@stardust.noc.frontier.fr)`
`02:15.14`	`justinu`	`100 sip register operations`
`02:15.32`	`Soul`	`justinu, ah, sorry, never had that experience`
`02:15.36`	`justinu`	`like for example, if your link went down, and then came back up, all the UAs will register`
`02:15.58`	`litage`	`justinu: i haven't read much on how SER works, but for registrations to take place with a SER box, SER would need to know the username and password for each party trying to register, right? and upstream * boxes also need to have that same registration information too, right?`
`02:16.13`	`Soul`	`justinu, correct, in that case we had that experience several times a day, for a month. no probs`
`02:16.59`	`justinu`	`the * boxes just need to know the SIP AOR`
`02:17.06`	`justinu`	`only the phones need the authentication info`
`02:17.27`	`litage`	`justinu: SIP AOR?`
`02:17.33`	`justinu`	`SER can be setup to auth against a database`
`02:17.36`	`justinu`	`address of record`
`02:17.56`	`Soul`	`justinu, yes, ser is much better. also too complicated.`
`02:18.17`	`justinu`	`SER is very complicated at all`
`02:18.22`	`justinu`	`much less so than asterisk`
`02:20.23`	`Soul`	`justinu, you mean ser is simple ?`
`02:20.52`	`litage`	`file, justinu: so if you have 1,000+ tenants, each with varying #s of extensions, and use SER to handle registrations, is it feasible to put all tenants on each * box?`
`02:23.16`	`inv_Arp`	`Qwell: around?`
`02:27.54`	`watchy`	`for music on hold whats a good streamer to use`
`02:28.05`	`watchy`	`for shoutcast?`
`02:28.35`	`Soul`	`watchy, we're using mpg123`
`02:28.54`	`watchy`	`hrm`
`02:28.59`	`watchy`	`not workin for me g`
`02:29.07`	`watchy`	`THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!`
`02:29.07`	`watchy`	`HTTP request failed: 404 Resource Not Found`
`02:29.11`	`Soul`	`pick another stream, most of themdont work`
`02:29.14`	`watchy`	`any special flags you give it?`
`02:29.20`	`watchy`	`if you give it a url?`
`02:29.27`	`Soul`	`yeah`
`02:30.37`	`watchy`	`which?`
`02:32.24`	`*** part/#asterisk Prival (i=user69@MTL-HSE-ppp207971.qc.sympatico.ca)`
`02:32.35`	`*** join/#asterisk Prival (i=user69@MTL-HSE-ppp207971.qc.sympatico.ca)`
`02:32.39`	`*** part/#asterisk Prival (i=user69@MTL-HSE-ppp207971.qc.sympatico.ca)`
`02:33.58`	`Soul`	`no clue, not in the office right now`
`02:34.28`	`watchy`	`ah`
`02:35.47`	`*** join/#asterisk smallb (n=smallb@prox47-249.trinidad.net)`
`02:35.57`	`smallb`	`hello`
`02:37.58`	`ObsidianX`	`hey folks, if im trying to setup a soft-phone like Kiax or MozIAX to connect to asterisk to only receive calls would i choose friend, user, or peer`
`02:38.24`	`marcus2`	`user`
`02:38.26`	`ObsidianX`	`i keep on getting "Inappropriate authentication received"`
`02:38.37`	`marcus2`	`that error has nthing to do with friend/user/peer tho`
`02:38.40`	`*** join/#asterisk linlin (i=linlin@c-67-184-231-233.hsd1.il.comcast.net)`
`02:38.45`	`ObsidianX`	`true`
`02:39.02`	`ObsidianX`	`when i choose user it says "No registration for peer 'test'"`
`02:39.53`	`ObsidianX`	`although i have a section [test] with secret=pass etc...`
`02:40.01`	`marcus2`	`do you have auth=md5 ?`
`02:40.28`	`ObsidianX`	`i just added it and it still doesn't work`
`02:41.01`	`ObsidianX`	`md5,plaintext,rsa doesn't work either`
`02:41.04`	`*** join/#asterisk NDT (n=me@cpe-24-194-166-119.nycap.res.rr.com)`
`02:47.51`	`Nugget`	`maybe "inappropriate" means you should put some clothes on or something.`
`02:48.32`	`*** join/#asterisk tengulre11 (n=tengulre@61.185.224.66)`
`02:50.05`	`*** join/#asterisk brockj49464 (n=brockj49@63.87.56.159)`
`02:54.24`	`*** join/#asterisk hhoffman (n=hhoffman@tor/session/x-ffdbc31ebc3f095f)`
`02:54.53`	`hhoffman`	`hi, is anyone using zasterisk?`
`02:57.56`	`ObsidianX`	`Nugget: heheh`
`02:58.06`	`ObsidianX`	`marcus2: any ideas?`
`03:02.07`	`*** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)`
`03:02.18`	`*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)`
`03:02.43`	`shmaltz`	`anybody here running the following:`
`03:02.45`	`shmaltz`	`asterisk 1.2.1`
`03:02.46`	`shmaltz`	`sipura`
`03:02.48`	`shmaltz`	`and polycom?`
`03:03.06`	`*** join/#asterisk EvilMetal (n=StorM@sgc91-2-82-237-76-2.fbx.proxad.net)`
`03:04.58`	`shmaltz`	`<PROTECTED>`
`03:05.44`	`*** join/#asterisk jef_ (i=fischer@p548466C5.dip.t-dialin.net)`
`03:11.47`	`*** join/#asterisk Cyon (n=cyon@cyons.net)`
`03:12.15`	`shmaltz`	`<PROTECTED>`
`03:12.21`	`Cyon`	`whos there?`
`03:12.39`	`shmaltz`	`hi`
`03:12.41`	`ObsidianX`	`"No registration for peer" agh`
`03:12.44`	`ObsidianX`	`what does that mean :(`
`03:13.27`	`*** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)`
`03:13.46`	`brockj49464`	`Anybody have any GXP-2000 to sell? Or reasons not to look at getting that phone?`
`03:13.53`	`Qwell`	`brockj49464: because they suck`
`03:14.02`	`Qwell`	`especially a used one...`
`03:14.09`	`_Sam--`	`i dont agree personally`
`03:14.17`	`_Sam--`	`i just installed 12 of them today for a real estate office`
`03:14.22`	`brockj49464`	`qwell: What exactly is weong with them?`
`03:14.26`	`_Sam--`	`for what they are...they are pretty good units.`
`03:14.29`	`Qwell`	`_Sam--: give them my condolences`
`03:14.47`	`_Sam--`	`i run my business on them, we have almost 20 people using them at my office as well`
`03:14.50`	`brockj49464`	`sam: Can they do on-hook anouncements (paging)?`
`03:15.16`	`_Sam--`	`i thikn the newest beta firmware does that.`
`03:15.19`	`_Sam--`	`finally`
`03:15.29`	`_Sam--`	`there is a wiki page about the phones that has some decent info`
`03:15.42`	`_Sam--`	`i dont know what else to compare them to for 85 bucks`
`03:15.54`	`Qwell`	`_Sam--: a GOOD headset, and a softphone`
`03:15.56`	`_Sam--`	`i am not saying you will love yours...but mine work fine for the role they are in`
`03:16.05`	`_Sam--`	`they blow away softphones`
`03:16.11`	`_Sam--`	`my sales guys switched from softphones to that`
`03:16.31`	`_Sam--`	`and we used good plantronics headsets`
`03:16.33`	`*** join/#asterisk FastJack (i=fastjack@p5091FE1E.dip.t-dialin.net)`
`03:16.44`	`_Sam--`	`i dont know what problems you had with the phones qwell`
`03:17.11`	`_Sam--`	`but ive dealt with their tech support as well which was refreshingly helpfuly...got through to somoeone right away who helped me out`
`03:17.14`	`brockj49464`	`I am looking at them for home. Trying to replace a pansonic kxtd1232 before it is worthless`
`03:17.28`	`ObsidianX`	`anybody know whats up with this error?`
`03:18.00`	`_Sam--`	`we are testing out the beta version of their newest firmware`
`03:18.06`	`_Sam--`	`and it seems pretty good for us`
`03:20.38`	`brockj49464`	`That is good that they seem to work.`
`03:20.49`	`_Sam--`	`ymmv based on your setup`
`03:21.08`	`_Sam--`	`all of my stuff ive been setting up is 100% ...no pri or pstn type stuff`
`03:21.20`	`_Sam--`	`er 100% voip`
`03:21.30`	`Qwell`	`ugh`
`03:21.39`	`\|omni\|`	`using remote gateways?`
`03:21.43`	`Qwell`	`realestate agents get MAD when things don't work`
`03:21.43`	`_Sam--`	`noope`
`03:21.58`	`_Sam--`	`well yeah , their asterisk box connects to an IAX provider`
`03:22.03`	`_Sam--`	`i guess that is a remote gateway....`
`03:22.15`	`\|omni\|`	`heh...I was just working on a system for a real estate office a couple weeks ago with someone`
`03:22.17`	`_Sam--`	`but the people assume the risks knowingly`
`03:22.18`	`iCEBrkr`	`damnit this phone number`
`03:22.25`	`iCEBrkr`	`I got some fucker calling me twice a day`
`03:22.36`	`Qwell`	`_Sam--: So, you told them to only expect 90% uptime?`
`03:22.40`	`iCEBrkr`	`I think it's Walmarts telemarketing/survey group`
`03:22.50`	`_Sam--`	`ive been running 100% voip at my business for about 1.2 years...`
`03:22.56`	`_Sam--`	`our uptime is closer to 99% for our calls`
`03:23.03`	`Qwell`	`99% is unacceptable`
`03:23.09`	`_Sam--`	`maybe for some high end clients`
`03:23.14`	`_Sam--`	`but based on budgets`
`03:23.14`	`Qwell`	`for anybody`
`03:23.22`	`_Sam--`	`they assume the risks`
`03:23.23`	`iCEBrkr`	`Five 9's!`
`03:23.25`	`_Sam--`	`they know`
`03:23.31`	`_Sam--`	`we talk about options`
`03:23.37`	`_Sam--`	`they choose based on cost`
`03:23.39`	`Qwell`	`99%...do you realize what that equates to?`
`03:23.39`	`\|omni\|`	`same on this side, but when I do a lot of forwarding (bounce exten to cell or whatever) I like low latency PSTN if possible`
`03:23.51`	`Qwell`	`1 hour every 4 days`
`03:23.59`	`Qwell`	`That is A LOT`
`03:24.04`	`Qwell`	`completely unacceptable`
`03:24.17`	`_Sam--`	`my shit works fine...i run a mail order business that over 10 mil a year in sales on it`
`03:24.21`	`_Sam--`	`and its acceptable just fine`
`03:24.32`	`_Sam--`	`you dont have to like it, thats fine`
`03:24.37`	`_Sam--`	`but people do`
`03:24.42`	`Qwell`	`_Sam--: So, what if UPS only delivered 4 days a week?`
`03:24.46`	`iCEBrkr`	`Qwell: What if you have 72hrs downtime in the month of Dec?`
`03:24.47`	`Qwell`	`You'd be freaking pissed`
`03:24.51`	`_Sam--`	`my phones deliver 7 days a week`
`03:24.53`	`Qwell`	`iCEBrkr: indeed`
`03:25.02`	`_Sam--`	`what is the difference between my PTP t1 and a PRI?`
`03:25.03`	`_Sam--`	`nothing`
`03:25.05`	`iCEBrkr`	`Qwell: Your average doesn't hold water, is all I'm saying :P`
`03:25.08`	`Qwell`	`iCEBrkr: on the 20th, 21st, and 22nd`
`03:25.25`	`_Sam--`	`so unless a route is down on my 8 homed provider...`
`03:25.30`	`_Sam--`	`the chances that i cant get there are pretty bad`
`03:25.32`	`iCEBrkr`	`...and hardware PBX's go dead a lot too..`
`03:25.33`	`_Sam--`	`my shit works.`
`03:25.44`	`_Sam--`	`call it as many times as you want..i'll give ya the number`
`03:26.03`	`Cyon`	`Hmmm, anyone here messed with getting faxing working?`
`03:26.38`	`\|omni\|`	`Sam...doing a similar setup here but putting a PRI into my rack`
`03:26.45`	`_Sam--`	`i started with a PRI`
`03:26.50`	`_Sam--`	`and switched to a PTP t1`
`03:27.04`	`\|omni\|`	`I have a PTP T1 from my rack to a client endpoint..but not here`
`03:27.10`	`_Sam--`	`and ive never regretted the decision`
`03:27.17`	`\|omni\|`	`low bandwidth for voice here`
`03:27.42`	`shmaltz`	`anybody here have an asterisk 1.2.1 system with a sipura and 2 polycom phones?`
`03:27.58`	`brockj49464`	`what exactly you trying to do with faxing?`
`03:28.36`	`Cyon`	`brockj49464: Get it working? ;-) I've tried the still beta t.38 patch, but unfortunately it's still buggy it would appear and I don't have the skill to update it`
`03:29.15`	`Cyon`	`brockj49464: So I jumped over to ser/openser, bypassing asterisk (I know, bad channel for that.) and tried to get sipura->ser->cisco working...`
`03:29.38`	`brockj49464`	`I am using g711u and seem to not have any problems for the 5 times I have used it this last week.`
`03:30.23`	`Cyon`	`brockj49464: Yeah, I've done ulaw; and can get it working 90%+ ; but I'm aiming for a solid 100%, or at least as close as possible`
`03:30.37`	`Cyon`	`When the customer does hundreds; they really notice that percentage of failures`
`03:31.09`	`*** join/#asterisk loud (n=ariel@cypher.punk.net)`
`03:31.37`	`Cyon`	`_Vile mentioned he does Sipura->ser->cisco, with perfection so far is success rates, so I wanted to give that a try; or get other people's views on it`
`03:32.23`	`brockj49464`	`That is true. My provider was where I was having problems when I used thier settinging on the ATA. When I defaulted it and set it up to my * box I had no problems _so_ far. Time will tell. It also solved my Dish Network problem...`
`03:32.59`	`Cyon`	`brockj49464: What ATA do you have? Just to ask...`
`03:33.04`	`*** join/#asterisk Jameno123 (n=james@63.210.246.146)`
`03:33.21`	`Cyon`	`But yeah, I can get some really solid results; but it's just not consistent enough..unfortunately`
`03:33.43`	`Jameno123`	`http://pastebin.com/501931`
`03:33.48`	`Jameno123`	`anyone have a solution to that?`
`03:34.05`	`Jameno123`	`"inlining failed in call to '__t4_framer_interrupt': function body notavailable"`
`03:34.07`	`brockj49464`	`SPA-2100 Getting 2 more of them. My plan is to start with cheap CID 2500 like phones and move to GXP-2000 as I get wiring and the phones.`
`03:34.29`	`alephcom_`	`I need an opinion from you all... On a low end ($9.99 per month) hosted pbx, do you think the customer needs more than 1 auto attendant?`
`03:34.38`	`Cyon`	`alephcom_: No.`
`03:34.39`	`\|omni\|`	`I'm liking the cisco 7960 for a work handset`
`03:34.53`	`Jameno123`	`\|omni\|, 7940G are great too`
`03:34.54`	`\|omni\|`	`I was on Zultys stuff before which is cool but these Ciscos are pretty nice`
`03:35.01`	`shmaltz`	`nybody here have an asterisk 1.2.1 system with a sipura and 2 polycom phones?`
`03:35.03`	`\|omni\|`	`I haven't tried a 7940 yet`
`03:35.10`	`Cyon`	`shmaltz: Sipura, but not polycom`
`03:35.14`	`Jameno123`	`7940/7960 same phone, just lesser phone "lines"`
`03:35.17`	`Jameno123`	`and cheaper price ;)`
`03:35.22`	`shmaltz`	`Cyon, what other phones?`
`03:35.28`	`\|omni\|`	`not as many appearances`
`03:35.28`	`Cyon`	`shmaltz: snom`
`03:35.30`	`alephcom_`	`Cyon: Tks, my thoughts too. I'm just designing an automated signup/management setup and I'm having lots of fun on the dialplan.`
`03:35.35`	`\|omni\|`	`how many does the 40 have.... 4?`
`03:35.41`	`Jameno123`	`2`
`03:35.45`	`\|omni\|`	`same XML mini-browser, etc.?`
`03:35.47`	`shmaltz`	`Cyon, so you have snom, sipura, and 1.2.1?`
`03:35.50`	`Jameno123`	`\|omni\|, yes`
`03:35.53`	`\|omni\|`	`sweet`
`03:35.54`	`Jameno123`	`same lcd, ect`
`03:35.59`	`Cyon`	`alephcom_: Yeah, I've been working on the same, with the auto-attendant being the hardest for me by far`
`03:36.01`	`\|omni\|`	`I setup some cool little apps on our PBX for the phone`
`03:36.03`	`Cyon`	`shmaltz: Yes`
`03:36.11`	`Jameno123`	`\|omni\|, any of them use the LCD?`
`03:36.15`	`shmaltz`	`Cyon, more than one snom? or just one?`
`03:36.24`	`\|omni\|`	`yea, browse to the app in LCD and submit data`
`03:36.32`	`\|omni\|`	`just simple stuff testing out the Cisco XML layout`
`03:36.35`	`Cyon`	`shmaltz: Just one for testing; have lots in stock for customers; why?`
`03:36.44`	`Cyon`	`shmaltz: Just ask whatever it is`
`03:36.50`	`\|omni\|`	`enter zip and get weather info, or lookup directory info`
`03:36.58`	`shmaltz`	`Cyon, I'm trying to test something, to see who has the bug: asteirsk, polycom, or sipura`
`03:37.01`	`\|omni\|`	`but the wheels are turning now`
`03:37.12`	`Cyon`	`brockj49464: I'll get it eventually, I'm just sure others have done it already`
`03:37.16`	`Cyon`	`shmaltz: What bug?`
`03:37.19`	`Jameno123`	`\|omni\|, yea, i was looking on trying to figure out how to present customer order data`
`03:37.24`	`*** join/#asterisk anonymouz666 (n=anonymou@gw.ideavalley.com.br)`
`03:37.28`	`Jameno123`	`cust calls in, the order# is shown on the phone when the agent answers`
`03:37.29`	`shmaltz`	`Cyon, I have a problem with sipura asterisk 1.2.1 and polycoms, I know it's a bug, but I'm not sure who is at fault`
`03:37.52`	`shmaltz`	`Cyon, when a polycom speaks with a sipura, and then does an attended xfer to anohter polycom, at the final stage there is only 1 way audio`
`03:38.09`	`shmaltz`	`this is on a single flat network, 1 subnet`
`03:38.10`	`anonymouz666`	`hi... there is a caller in a queue.. I think its crashed because his wait time: (wait: -525351:-37, prio: 0)`
`03:38.11`	`shmaltz`	`no nat`
`03:38.16`	`Jameno123`	`the only way so far ive figured out is just to throw the order# in the callerid info heh`
`03:38.18`	`anonymouz666`	`how do I remove this one?`
`03:38.29`	`Jameno123`	`sooooooo - does anyone have a solution to that? http://pastebin.com/501931`
`03:38.40`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`03:38.41`	`shmaltz`	`if I change the sipura to canreinvite=no, then everything is ok, but another problem arises`
`03:38.59`	`\|omni\|`	`Jameno123: like enter order number and get details?`
`03:39.08`	`*** join/#asterisk HolyGod (i=nobody@got.securebinary.com)`
`03:39.31`	`shmaltz`	`Jameno123, what version of zap? and what version of kernel?`
`03:39.36`	`\|omni\|`	`pretty simple to write little apps, we've done a ton of web development in the past so I just wrote a little PHP that dumps results to the Cisco XML elements and it works pretty well..pull from DB or whatever`
`03:39.45`	`anonymouz666`	`is it possible to remove callers crashed from a queue?`
`03:40.06`	`Jameno123`	`shmaltz, zap=latest, kernel=2.6.12(+patches)`
`03:40.12`	`Cyon`	`shmaltz: Hmmm, beyond me`
`03:40.22`	`Jameno123`	`just freshly downloaded from SVN about an hour ago`
`03:40.26`	`\|omni\|`	`I'd like to play with some outlook integration`
`03:40.50`	`shmaltz`	`Cyon, but if you could test this for me with the snoms then it would confirm that:`
`03:40.52`	`shmaltz`	`1. its not the sipuras,`
`03:40.53`	`shmaltz`	`2. It's not asterisk`
`03:41.12`	`shmaltz`	`Jameno123, which one from svn? tags or trunk?`
`03:41.16`	`Jameno123`	`trunk`
`03:41.34`	`Cyon`	`shmaltz: I can test it at the office tomorrow; but we used it extensively; only way it would replicate is if we did snom->sipura->snom`
`03:41.41`	`Jameno123`	`shmaltz, (gcc 4.0.1)`
`03:41.46`	`Cyon`	`shmaltz: Other than that, we never ean into it`
`03:41.49`	`Cyon`	`*ran`
`03:42.23`	`Cyon`	`shmaltz: I'm generally here all day; just pm me any time and I'll get on it`
`03:42.31`	`shmaltz`	`Cyon, also if I do canreinvite=no all is godd, so if you test it you will have to make sure that the rtp always gets reinvited`
`03:42.43`	`shmaltz`	`Cyon, Thank you`
`03:42.43`	`*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`03:42.46`	`VeNoMouS_`	`woah i forgot i left this on`
`03:42.46`	`VeNoMouS_`	`lol`
`03:42.56`	`*** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`03:43.37`	`Cyon`	`shmaltz: Easily; when I'm at the phones :)`
`03:44.32`	`Jameno123`	`shmaltz: i have no zaptel cards as well.`
`03:44.40`	`Jameno123`	`just trying to install ztdummy`
`03:44.50`	`shmaltz`	`Jameno123, that shouldn't make a difference`
`03:44.58`	`shmaltz`	`this problem is beyond me`
`03:45.33`	`Jameno123`	`<PROTECTED>`
`03:45.42`	`Jameno123`	`static inline void __t4_framer_interrupt(struct t4 *wc, int span);`
`03:45.43`	`Jameno123`	`wtf`
`03:45.54`	`Jameno123`	`heh, no function body, as it says.`
`03:46.04`	`*** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it)`
`03:46.21`	`*** join/#asterisk nutria (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)`
`03:47.57`	`Jameno123`	`kinda looks outta place, guess it needs moved up to the top of the file :( though im not a C expert, have no idea what im talking about.`
`03:49.35`	`Cyon`	`Hmmm, does anyone recall an issue where a call tries to use speex when neither side of the sip headers support it; and then it has no trnslation path and the call dies?`
`03:50.09`	`*** join/#asterisk bmg505 (n=leon@c1-61-9.rndf.isadsl.co.za)`
`03:50.14`	`dily`	`hi@all`
`03:50.49`	`dily`	`i try to compile bristuff-0.3.0-PRE-1c`
`03:51.01`	`dily`	`but when complie the zaphfc.ko i have strange function undefined warning`
`03:51.31`	`dily`	`like this: *** Warning: "zt_register" [/usr/src/bristuff/zaphfc/zaphfc.ko] undefined!`
`03:51.44`	`dily`	`any idea?`
`03:51.59`	`Cyon`	`Never looked at or tried that module`
`03:52.52`	`dily`	`i try to install bristuff on many system/distributions but i have the some errors...`
`03:54.40`	`ObsidianX`	`has anybody ever had an error when setting up IAX along the lines of "No registration for peer 'user'"?`
`03:56.30`	`dily`	`anyone use bristuff?!?`
`03:56.41`	`Cyon`	`Actually, let me ask this way; what is speex (I know it's a codec) but how do I totally disable it everywhere? lol`
`03:57.23`	`Cyon`	`Like, why does asterisk say it's trying to be used when talking to the cisco...`
`03:57.48`	`*** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net)`
`03:59.21`	`brockj49464`	`cyon: Do you disallow=all then allow what ones you want to use?`
`03:59.31`	`Cyon`	`brockj49464: absolutely`
`03:59.44`	`Cyon`	`It looks like cisco ignores it and tries to establish calls as speex`
`04:00.22`	`Cyon`	`The only one allowed in my sipuras is ulaw, the only one allowed in asterisk is ulaw, and the cisco has "codec g711ulaw" as well...`
`04:00.47`	`Cyon`	`And yet: [2006-01-11 17:52:24] WARNING[32704]: Unable to find a codec translation path from speex to ulaw`
`04:02.30`	`hhoffman`	`is there a better tts then festival to use with asterisk?`
`04:03.55`	`Cyon`	`http://pastebin.com/501950 <-- anyone have any ideas?`
`04:04.01`	`Cyon`	`hhoffman: Not that I've seen`
`04:07.31`	`ObsidianX`	`http://www.voipuser.org/forum_topic_4196.html`
`04:08.28`	`*** join/#asterisk mud (n=mud@206-248-138-115.dsl.teksavvy.com)`
`04:09.08`	`*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)`
`04:10.08`	`fugitivo`	`hhoffman: www.cepstral.com`
`04:11.45`	`Jameno123`	`Cyon, try "disallow=all" "allow=ulaw"`
`04:11.59`	`Jameno123`	`hrm, nobody has any ideas about my issue?`
`04:12.21`	`Jameno123`	`http://pastebin.com/501931`
`04:13.00`	`dily`	`http://www.loquendo.com/regional_preferences.htm`
`04:13.17`	`Cyon`	`Jameno123: Was done long ago`
`04:13.32`	`Cyon`	`Jameno123: speex isn't even a protocol that asterisk has by default`
`04:13.44`	`file`	`s/protocol/codec`
`04:14.18`	`Cyon`	`Jameno123: Something is trying to use it, or makes asterisk think it is; yet cisco doesn't support that codec either it would appear, and my sipura is set to use g711, and pref. codec only.`
`04:14.27`	`Cyon`	`file: Sorry, yes.`
`04:15.24`	`hhoffman`	`fugitivo: thanks checking now`
`04:15.41`	`Jameno123`	`twisted[asteria], wakey wakey!`
`04:16.46`	`hhoffman`	`fugitivo: are these voice compatible with festival?`
`04:17.00`	`Cyon`	`Jameno123: I'm not a coder anymore; but can I see a pastebin of all the verbose/debug lines?`
`04:17.04`	`fugitivo`	`hhoffman: no, it's closed source`
`04:17.07`	`SwK`	`jameno123 is from teh svn or from the 1.2.1 tarball?`
`04:17.18`	`Cyon`	`Jameno123: So I can see which src files it is bouncing through`
`04:17.34`	`SwK`	`it looks like a bad check out from svn`
`04:17.55`	`hhoffman`	`fugitivo: k, thx`
`04:17.57`	`Jameno123`	`SwK, svn, ive deleted and redownloaded twice now.`
`04:18.14`	`SwK`	`it looks like 1/2 and update to me`
`04:18.23`	`hhoffman`	`ah, but I'm guessing it's meant to work with * as they have digium links on their page`
`04:18.25`	`SwK`	`are you running head?`
`04:18.31`	`SwK`	`(or trunk now)`
`04:18.36`	`Jameno123`	`SwK, trunk`
`04:18.45`	`Jameno123`	`ive always ran CVS-HEAD`
`04:18.57`	`Cyon`	`Jameno123: Ah, I assumed it was the tgz download...`
`04:19.07`	`SwK`	`i did to til 1.2.X was released`
`04:19.17`	`SwK`	`1.0 was just to damned old and missing too many features`
`04:19.23`	`*** join/#asterisk santiago (n=santiago@208.195.215.97)`
`04:19.35`	`Qwell`	`I run svn roots`
`04:19.37`	`SwK`	`I would try compiling the 1.2.1 zap sources from the tarball and see what happens`
`04:19.48`	`Qwell`	`more features than trunk`
`04:20.41`	`Jameno123`	`hrm`
`04:20.59`	`Jameno123`	`will try`
`04:21.58`	`*** join/#asterisk Hybrid (n=hybridra@calera-47.cher.brightok.net)`
`04:22.26`	`Jameno123`	`SwK, yea, the "out of date" stuff, is what concerns me ;)`
`04:22.51`	`SwK`	`i wouldnt worry about it rightnow`
`04:24.36`	`*** part/#asterisk santiago (n=santiago@208.195.215.97)`
`04:25.15`	`tainted_`	`how do i do E911 for a client?`
`04:25.32`	`Jameno123`	`SwK, waiting on the box to rebewt, i guess we'll see :)`
`04:25.35`	`SwK`	`very carefully`
`04:25.44`	`SwK`	`tainted_ are you an ITSP?`
`04:25.57`	`tainted_`	`SwK it's for a client`
`04:26.04`	`*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)`
`04:26.15`	`tainted_`	`i don't normally do this kind of stuff`
`04:26.40`	`Cyon`	`Jameno123: Reboot? Why woyld you reboot?`
`04:27.01`	`Cyon`	`s/oy/ou`
`04:28.41`	`Jameno123`	`Cyon, ;) kernel updates`
`04:28.50`	`Cyon`	`Jameno123: Ah, ok. :)`
`04:29.09`	`Jameno123`	`would be so nice to`
`04:29.17`	`Jameno123`	`cat newkernel > /proc/kcore`
`04:29.20`	`Jameno123`	`and not have to reboot ;)`
`04:29.25`	`Jameno123`	`but i dont think we'll see the day`
`04:29.28`	`SkramX`	`heh, it would.`
`04:29.31`	`Cyon`	`I can't wait till we have dynamic kernel loading...`
`04:29.48`	`Cyon`	`Nah, it's doable; just the entire structure would have to be redone, and it'll be years...`
`04:29.55`	`Cyon`	`But it will happen eventually`
`04:30.03`	`Hybrid`	`Anybody have Mechwarrior 3?`
`04:31.42`	`*** part/#asterisk Hybrid (n=hybridra@calera-47.cher.brightok.net)`
`04:32.00`	`Jameno123`	`SwK, suggest using 1.2.1 [.tgz] completely or just zaptel?`
`04:34.40`	`SwK`	`1.2.1 zap shoudl work with trunk at this time,altho i'm not sure... 1.2.1 would probably be better for products as its a known quantity and its not missing much from trunk yet (unless there is something in trunk you really need)`
`04:36.24`	`Jameno123`	`swk it built properly ;) heh, it should run then`
`04:37.06`	`Jameno123`	`hah`
`04:37.07`	`Jameno123`	`yay!`
`04:37.12`	`Jameno123`	`<PROTECTED>`
`04:37.18`	`Jameno123`	`<PROTECTED>`
`04:37.20`	`fugitivo`	`WIRING WIRING WIRING`
`04:37.22`	`Jameno123`	`heh`
`04:37.34`	`hnupik`	`children`
`04:38.02`	`SwK`	`hah`
`04:38.09`	`SwK`	`it always gripes about g729`
`04:38.45`	`*** join/#asterisk qhrisnd (n=qhrisnd@ppp-71-129-177-185.dsl.irvnca.pacbell.net)`
`04:38.51`	`file[laptop]`	`hahaha...`
`04:38.58`	`qhrisnd`	`Good evening everyone :-)`
`04:38.59`	`file[laptop]`	`my cellphone bill is insane`
`04:39.48`	`Jameno123`	`SwK, hrm, should i rm -rf that and re-make install?`
`04:39.52`	`[TK]D-Fender`	`Perhaps its the 800# attached to it :)`
`04:40.04`	`Qwell`	`Jameno123: It's just a warning...ignore it if that was the only file`
`04:40.05`	`file[laptop]`	`wait for it people`
`04:40.13`	`Qwell`	`file[laptop]: $938?`
`04:40.16`	`Qwell`	`CAD`
`04:40.18`	`file[laptop]`	`invoice amount$1,603.26`
`04:40.19`	`ObsidianX`	`how would i go about fixing the error "Inappropriate authentication received" when i try to connect an IAX client to *`
`04:40.21`	`SwK`	`yeah what qwell said`
`04:40.21`	`Qwell`	`jesus`
`04:40.35`	`rob0`	`file[laptop]: have it committed :)`
`04:40.35`	`Qwell`	`file[laptop]: how the hell did you manage that?`
`04:40.39`	`SwK`	`it always gribes about codec_729 cause you dont have the source for it`
`04:40.40`	`*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`04:40.44`	`file[laptop]`	`we'll see.`
`04:40.45`	`rob0`	`I know well how he did it!!`
`04:40.48`	`ManxPowe`	`Just how DOES one get a $1,000 cell phone bill anyway?`
`04:40.51`	`ManxPowe`	`file, GO PREPAY!`
`04:41.02`	`ManxPowe`	`rob0, all those phonesex phone calls?`
`04:41.10`	`SwK`	`is that CDN File?`
`04:41.12`	`file[laptop]`	`I just have the transaction on my account, I don't have the invoice online yet and my balance isn't adjusted yet`
`04:41.13`	`rob0`	`I saw him here, typing in IRC, while on the road`
`04:41.15`	`fugitivo`	`WTF??`
`04:41.17`	`file[laptop]`	`SwK: yes`
`04:41.19`	`file[laptop]`	`rob0: yup`
`04:41.30`	`file[laptop]`	`they probably billed me for data, and backbilled me for past data usage`
`04:41.31`	`fugitivo`	`file[laptop]: $1600?????`
`04:41.33`	`SwK`	`file: oh so its like a normal 100USD phone bill?`
`04:41.41`	`ManxPowe`	`Ah. Mine would be like $5,000 if I wasn't on the flat rate data plan`
`04:41.47`	`file[laptop]`	`I need to calculate how it got to that amount though`
`04:41.48`	`file[laptop]`	`it makes no sense`
`04:41.57`	`Qwell`	`$50/kb?`
`04:41.58`	`Jameno123`	`SwK, yea, it bitched about more, but im not pasting them all :) should i rm -rf the modules dir, and reinstall it all completely?`
`04:42.07`	`Jameno123`	`like 15 files are listed`
`04:42.08`	`Jameno123`	`heh`
`04:42.13`	`h3x`	`damn bid snipers`
`04:42.21`	`xachen`	`Canada data rates are bad for mobile providers`
`04:42.23`	`h3x`	`i accidently pasted a auction item number in where a price goes`
`04:42.26`	`xachen`	`they will coin you easily $1/mbv`
`04:42.29`	`h3x`	`and i bid 5 million on an ATA device`
`04:42.40`	`file[laptop]`	`my regular bill is $60`
`04:42.40`	`SwK`	`jamesno123: probably want to get rid of them but not the g729 one`
`04:42.48`	`SwK`	`you'll need it for g729`
`04:42.55`	`Jameno123`	`SwK, yea, i use g729, i know about it ;)`
`04:43.03`	`file[laptop]`	`so I used 100MB of data apparently`
`04:43.09`	`Jameno123`	`like you said, only because it wasnt compiled directly be the source`
`04:43.16`	`fugitivo`	`file[laptop]: don't pay it, that's insane`
`04:43.22`	`xachen`	`downloading porn onto your blackberry? :D`
`04:43.24`	`file[laptop]`	`fugitivo: I'm waiting for the bill.`
`04:43.26`	`xachen`	`:O rather`
`04:43.26`	`\|omni\|`	`Cingular did that to me a couple months ago but it was only $580 for data`
`04:43.27`	`*** join/#asterisk sumonish (n=God@203.12.249.168)`
`04:43.32`	`sumonish`	`hi all`
`04:43.45`	`\|omni\|`	`I switched to the unlimited data account... a mere $20 more than I was paying already`
`04:43.46`	`\|omni\|`	`bastages`
`04:43.59`	`*** join/#asterisk srt (n=nobody@18.120.9.213.dsl.getacom.de)`
`04:44.16`	`file[laptop]`	`I'm not overly thrilled, but I legitimately used it so if they billed it right... yeah`
`04:44.31`	`file[laptop]`	`life goes on`
`04:44.53`	`file[laptop]`	`so help me god if my mother opens my cellphone bill`
`04:45.04`	`SwK`	`hahaha`
`04:45.10`	`fugitivo`	`heart attack`
`04:45.22`	`sumonish`	`i have an asterisk server which my boss has setup and left me with unfortunatly the CallerID is causeing an issue where when a call comes in it dumps the call i have the following issue in Myphp The $cfg['PmaAbsoluteUri'] directive MUST be set in your configuration file! can someone tell my what it means and how to fix it??`
`04:45.29`	`file[laptop]`	`she's been nosey lately, she opened my credit card statement ahead of me while I was right in front of her`
`04:45.38`	`file[laptop]`	`and my rrsp notice`
`04:45.44`	`SwK`	`rrsp?`
`04:45.51`	`file[laptop]`	`it's like, "uh... I'm 19 here... get out of my finances"`
`04:45.51`	`rob0`	`yikes!`
`04:45.53`	`\|omni\|`	`sumonish: , that's not your issue, that's just a setting in phpMyAdmin`
`04:45.58`	`file[laptop]`	`SwK: registered retirement savings plan`
`04:46.02`	`SwK`	`oh`
`04:46.06`	`\|omni\|`	`you can edit config.inc`
`04:46.12`	`SwK`	`i guess thats cdn for 401k`
`04:46.13`	`rob0`	`file[laptop]: get a PO Box`
`04:46.27`	`\|omni\|`	`and set the full URL to phpMyAdmin (i.e. http://path.to.server/phpMyAdmin) and that message will go away`
`04:46.39`	`file[laptop]`	`rob0: mmm I could`
`04:46.41`	`rob0`	`in USPS they're pretty cheap`
`04:46.41`	`sumonish`	`i edited the zapata.conf`
`04:46.51`	`sumonish`	`to turn of caller id is that right?`
`04:46.55`	`sumonish`	`i seems to work`
`04:46.55`	`rob0`	`I pay $18/year I think`
`04:47.05`	`file[laptop]`	`I believe it's $60 CAD/year here`
`04:47.24`	`sumonish`	`ok omni`
`04:47.36`	`rob0`	`they cost more in cities, mine is in a tiny town`
`04:47.36`	`SwK`	`damn did apple release enuff patches yesterday?`
`04:47.49`	`MikeJ__`	`file, so that's like one regular cell bill a year?`
`04:47.50`	`rob0`	`but Canada is no doubt different`
`04:48.04`	`file[laptop]`	`MikeJ__: more`
`04:48.12`	`file[laptop]`	`my regular cell bill is $60/mth total`
`04:48.50`	`sumonish`	`omni where is config.inc stored?`
`04:49.12`	`Jameno123`	`hrm, alright, seems to work :)`
`04:49.22`	`Jameno123`	`but didnt solve my problem/reason for upgrading`
`04:49.23`	`Jameno123`	`heh`
`04:49.25`	`Jameno123`	`1st File Descriptor: -1`
`04:49.29`	`Jameno123`	`<PROTECTED>`
`04:50.49`	`Jameno123`	`after it bridge's a call, it hangs and gives nothing.`
`04:51.03`	`Jameno123`	`service provider returning no data? or some other weird crapola?`
`04:51.30`	`twisted[asteria]`	`SwK, you sure you don't have that shit?`
`04:51.46`	`bsdfreak`	`heh`
`04:52.04`	`qhrisnd`	`I need help with 2 things: 1) I need to find out how to create in my dial plan, a way to make an extension ring over to another extension when its busy. 2) I would like to know how to (if possible) route calls based upon caller id. Can anyone give me some tips?`
`04:52.27`	`Qwell`	`twisted[asteria]: y0`
`04:52.44`	`Qwell`	`twisted[asteria]: going to ETel?`
`04:53.01`	`twisted[asteria]`	`Qwell, no`
`04:53.03`	`ManxPowe`	`qhrisnd, See "show application dial" and the [macro-stdexten] section of extensions.conf. Also see the Wiki and the Asterisk book.`
`04:53.06`	`ManxPowe`	`~docs`
`04:53.08`	`jbot`	`i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html`
`04:53.19`	`Qwell`	`twisted[asteria]: shame..`
`04:53.26`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`04:53.37`	`fugitivo`	`qhrisnd: and DIALSTATUS`
`04:53.47`	`qhrisnd`	`thank you`
`04:53.49`	`twisted[asteria]`	`Qwell, well, if I had known about it sooner, i might could have`
`04:55.51`	`SwK`	`twisted I am sure`
`04:56.57`	`*** join/#asterisk aless (n=fruribe@pc-100-230-83-200.cm.vtr.net)`
`04:56.58`	`*** join/#asterisk pdugas (n=pdugas@h73.90.40.69.ip.alltel.net)`
`04:57.11`	`aless`	`hi, which ports do i need to forward when using a nat?`
`04:57.18`	`Qwell`	`aless: which channel types?`
`04:58.20`	`inv_Arp`	`aless: any port you want`
`04:58.40`	`aless`	`im connecting two servers with iax`
`04:59.00`	`Qwell`	`aless: 4569`
`04:59.06`	`*** part/#asterisk loud (n=ariel@cypher.punk.net)`
`04:59.27`	`*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)`
`05:00.04`	`aless`	`only that one? arent any other services sending packets?`
`05:00.18`	`ObsidianX`	`netstat -nap`
`05:00.20`	`ObsidianX`	`if you wanna find out`
`05:00.31`	`bsdfreak`	`aaa`
`05:00.38`	`ObsidianX`	`course you'll have to look for asterisk processes :P`
`05:02.36`	`SwK`	`damn it`
`05:03.53`	`Jameno123`	`blah blah blah! damn thing :( argh, why the heck doesnt this thing WORK!!!!!!!!!!!! :(`
`05:04.02`	`Jameno123`	`how can i determine where my problem is :(`
`05:04.31`	`mogorman`	`? Jameno123`
`05:04.32`	`Jameno123`	`i call from my cisco 7960 via sip to asterisk1, asterisk1 dials asterisk2, asterisk2 dials our provider.`
`05:04.35`	`mogorman`	`calm down....`
`05:04.52`	`Jameno123`	`asterisk1->asterisk2 = iax`
`05:04.56`	`Jameno123`	`asterisk2->provider = iax`
`05:05.01`	`mogorman`	`k`
`05:05.08`	`Jameno123`	`if i do "iax2 show channels" on asterisk2, it shows a "UP" bridged channel`
`05:05.22`	`Jameno123`	`yet, i see hear nothing`
`05:05.36`	`mogorman`	`i see hear?`
`05:05.41`	`Jameno123`	`see/hear*`
`05:05.49`	`Jameno123`	`i see no errors, and hear nothing on the phone`
`05:05.56`	`Jameno123`	`if i hang up the phone`
`05:05.57`	`mogorman`	`is jitterbuffer on?`
`05:06.39`	`Jameno123`	`asterisk1 disconnects the call, but asterisk2 still thinks the call is in progress, and doesnt disconnect until it times out.`
`05:06.46`	`*** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca)`
`05:06.49`	`Jameno123`	`mogorman, which server? all of them?`
`05:07.00`	`mogorman`	`any of them?`
`05:07.07`	`Jameno123`	`this shit just started happening 3 days ago`
`05:07.11`	`Jameno123`	`its been fine "forever" :(`
`05:07.24`	`Jameno123`	`asterisk2=jitterbuffer=no`
`05:07.43`	`Jameno123`	`asterisk1=jitterbuffer=no`
`05:07.48`	`Jameno123`	`i dont know about my service provider`
`05:08.07`	`mogorman`	`hmm it sounds like a bug we have been working on`
`05:08.16`	`Jameno123`	`bug? heh`
`05:08.23`	`Jameno123`	`it just "mysteriously" happens?`
`05:08.46`	`mogorman`	`does this happen if you turn off iax native transfer`
`05:08.47`	`Jameno123`	`heh, just magically started happening one day`
`05:09.09`	`watchy`	`whats the quick reset for a 7960?`
`05:09.40`	`mogorman`	`pull the plug ^_^`
`05:09.42`	`Jameno123`	`watchy, reboot? (*+6+services)`
`05:09.50`	`watchy`	`thanks brother`
`05:09.52`	`Jameno123`	`err settings`
`05:09.59`	`Jameno123`	`* 6 settings`
`05:10.02`	`Jameno123`	`1 of the two`
`05:10.12`	`Qwell`	`real men #`
`05:10.24`	`Jameno123`	`mogorman: hrm.`
`05:10.31`	`Qwell`	`<rant>`
`05:10.31`	`Jameno123`	`i cant say ive ever done that before ;)`
`05:10.42`	`Qwell`	`Why did Cisco do # for the reboot on the sccp 7960?`
`05:10.47`	`Jameno123`	`let me go read some docs, or shed me some light :)`
`05:10.52`	`Qwell`	`You have to be in settings for it to work...`
`05:11.05`	`Jameno123`	`sccp, blah!`
`05:11.07`	`Qwell`	`and...what do you need to press to unlock the phone? That's right... **#`
`05:11.09`	`mogorman`	`id try turning off native transfer first`
`05:11.21`	`Jameno123`	`mogorman, thats what im reading docs to figure out how ;)`
`05:11.29`	`Qwell`	`So, if you want to unlock, and it didn't appear to work the first time...what do you do? You press it again`
`05:11.41`	`Qwell`	`and in doing so...you reboot the damn thing. How stupid...`
`05:11.42`	`Jameno123`	`notransfer=no ?`
`05:11.42`	`Qwell`	`</rant>`
`05:12.04`	`mogorman`	`hmm i think so....`
`05:12.09`	`mogorman`	`id have to look it up sorry`
`05:13.01`	`*** join/#asterisk aless_ (n=fruribe@pc-100-230-83-200.cm.vtr.net)`
`05:13.21`	`Jameno123`	`mogorman, nope, didnt help`
`05:13.42`	`Jameno123`	`i think junction networks is being a pain in my arse again :)`
`05:13.48`	`mogorman`	`did you turn it on all points and check it again`
`05:14.06`	`Jameno123`	`i turned it "off"`
`05:14.10`	`Jameno123`	`it should be "on" ?`
`05:14.22`	`Jameno123`	`i disabled it, on all servers, yet`
`05:14.23`	`Jameno123`	`yes*`
`05:14.38`	`Jameno123`	`err both* well, the two i have access too, not my providers, of course.`
`05:14.51`	`Jameno123`	`i think its just a provider issue :(`
`05:15.03`	`mogorman`	`maybe`
`05:15.05`	`Jameno123`	`ive never had any problems, and if i dial other phones on my asterisk server, i dont have problems.`
`05:15.08`	`Jameno123`	`so if i do`
`05:15.15`	`Jameno123`	`phone1->ast1->ast2->phone2`
`05:15.18`	`Jameno123`	`no problems, ever`
`05:15.29`	`Jameno123`	`phone1->ast1->ast2->provider=problems`
`05:15.35`	`mogorman`	`yeah probably`
`05:24.49`	`Jameno123`	`mogorman, ;) so stressful when you cant figure out why something is happening hehe`
`05:30.18`	`*** join/#asterisk ThaZZa_Work (n=me@203.80.44.200)`
`05:32.21`	`Jameno123`	`whelp thanks guys for your help, ill go chew on the ear of my service provider tomorrow.`
`05:32.25`	`Jameno123`	`cya.`
`05:32.26`	`litage`	`if you have 1,000+ tenants, each with varying #s of extensions, and use SER to handle registrations, is it feasible to put all tenants on each * box?in a phone call [through *], if 1 person is talking and the other person is silent, are they both sending the same amount of voice data?`
`05:33.24`	`mogorman`	`yeah i understand Jameno123`
`05:33.50`	`litage`	`in a phone call [through *], if 1 person is talking and the other person is silent, are they both sending the same amount of voice data?`
`05:34.58`	`Qwell`	`litage: Since * doesn't do VAD, yes`
`05:35.07`	`litage`	`VAD?`
`05:35.13`	`Qwell`	`~vad`
`05:35.14`	`jbot`	`i heard vad is Voice Activity Detection`
`05:35.19`	`litage`	`ah`
`05:36.21`	`litage`	`Qwell: so the type (volume, pitch, etc) of audio/voice doesn't affect the amount of data transferred?`
`05:36.48`	`Qwell`	`afaik, no`
`05:37.18`	`litage`	`interesting`
`05:38.30`	`*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)`
`05:38.32`	`*** join/#asterisk Evanr00d (n=david@ip68-107-162-212.lu.dl.cox.net)`
`05:38.35`	`lo_tech`	`not AM, bro... louder doesnt mean bigger data :)`
`05:39.52`	`Jameno123`	`oh before i go`
`05:39.54`	`Jameno123`	`one more thing :)`
`05:40.26`	`Jameno123`	`WHen a user transfers a call, on a cisco ip phone (SIP), to another extension, why does the phone never receive anymore calls?`
`05:40.30`	`Jameno123`	`asterisk thinks its "busy"`
`05:40.31`	`litage`	`lo_tech: "not AM"?`
`05:40.42`	`Jameno123`	`litage, "its not AM (like radio)`
`05:41.02`	`lo_tech`	`litage: amplitude modulation...`
`05:41.06`	`litage`	`ah`
`05:41.21`	`*** join/#asterisk Mavantix (n=mavantix@69-168-33-232.chvlva.adelphia.net)`
`05:41.42`	`Jameno123`	`anyone have any idea what i possible am doing wrong?`
`05:41.42`	`litage`	`so if 2 people are using sip and g729, will each person's incoming and outgoing data and voice streams be fairly constant?`
`05:42.18`	`Jameno123`	`they used to transfer, and then receive more inbound calls`
`05:42.23`	`Jameno123`	`now the phones are staying busy`
`05:42.38`	`Jameno123`	`probably something todo with "tT" ? or canreinvite or something?`
`05:42.44`	`Mavantix`	`is there anyway to have asterisk IM me incoming call info, log messages, etc?`
`05:43.01`	`lo_tech`	`all things being equal, without silence suppression or vad, yes... the bandwidth used will be equal for both parties, regardless of how loud or the amount of silence for each phone`
`05:43.03`	`ManxPowe`	`Jameno123, sounds like you are using imcominglimit=1 or setgroup, etc`
`05:43.44`	`ManxPowe`	`Jameno123, if so, this is a know issue, see the mailing list archives, there may be a fix or something.`
`05:44.01`	`Jameno123`	`ManxPowe: hrm, they do disable callwaiting, if callwaiting is enabled it rings fine.`
`05:44.16`	`Jameno123`	`as i thought, if you transfer your phone is released from the call?`
`05:44.23`	`Jameno123`	`i didnt think the phone 'bridged' the call.`
`05:44.56`	`Jameno123`	`ManxPowe, incominglimit is undefined in my sip.conf`
`05:45.19`	`Jameno123`	`and setgroup would be a no.`
`05:45.42`	`Jameno123`	`though, i dont specify "canreinvite"`
`05:45.46`	`Jameno123`	`in the sip.conf, so thats probably the issue?`
`05:46.44`	`watchy`	`anyway to set cisco volume in sipdefault?`
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`06:13.14`	`littleball`	`hello, i use E1/PRI, asterisk1.2.1. I got the following warning in the console:`
`06:13.15`	`littleball`	`Jan 12 14:00:18 NOTICE[6681]: channel.c:2416 __ast_request_and_dial: Don't know what to do with control frame 15`
`06:13.21`	`littleball`	`what does it mean?`
`06:27.52`	`wunderkin`	`heh holy crap, the * messages log on my one server never has been rotated`
`06:29.10`	`lo_tech`	`not so bad unless you`
`06:29.16`	`lo_tech`	`are verbose, debug`
`06:30.52`	`wunderkin`	`22k lines since sept`
`06:31.16`	`wunderkin`	`verbose is set to 20 but i dont do much with it, just testing`
`06:34.56`	`*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)`
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`06:47.02`	`chat_jokey`	`hello people`
`06:48.10`	`chat_jokey`	`i am currently doing some asterisk sizing .. task is to support 150 incoming TDM lines and 175 outgoing lines .. with approximately 4000 extensions (mostly used only for intercom)`
`06:48.24`	`chat_jokey`	`anyone can suggest me any pointers on the dimensioning of the same ?`
`06:48.43`	`chat_jokey`	`i read up with voip-info.org .. but its kinda not clear ..`
`06:49.03`	`chat_jokey`	`I am averaging about 400 - 500 odd extensions running from one asterisk box ..`
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`06:56.10`	`welles`	`hi all`
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`07:14.49`	`welles`	[help] i try to install mpg123 on centos4 and it hints that :'decode_i586.s:44: Error: suffix or operands invalid for `push' ...' what's wrong? my machine is 64bit machine
`07:25.25`	`litage`	`is H323 or SIP more NAT- and network-friendly?`
`07:25.58`	`*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)`
`07:29.00`	`*** join/#asterisk Lee619 (n=Lee@netblock-66-245-229-224.dslextreme.com)`
`07:30.01`	`Lee619`	`hello`
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`07:34.49`	`tzafrir_laptop`	`welles, use rawplayer, unless you want to stream music`
`07:36.58`	`welles`	`rawplayer? ok,i have a try .it can replace mpg123 for music on hold on *?`
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`07:52.00`	`koperniqs`	`hi`
`07:52.12`	`Lee619`	`good morning`
`07:56.27`	`infinity1`	`good nite`
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`08:14.16`	`ObsidianX`	`litage: i read that IAX was NAT friendly`
`08:14.28`	`ObsidianX`	`litage: i think it uses UDP`
`08:15.23`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`08:18.27`	`chat_jokey`	`any one can give pointers on clustering asterisk ?`
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`08:23.21`	`bazz`	`i'm trying to get asterick going, i've set up my extentions.conf file (i thought) but when i copy a .call file into the outgoing spool i get __ast_request_and_dial: Don't know what to do with control frame 15 and then attempt_thread: Call failed to go through, reason 3. any ideas?`
`08:23.46`	`*** join/#asterisk ToTo (n=ToTo@host243-91.pool8260.interbusiness.it)`
`08:24.09`	`wellng`	`hi all`
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`08:27.36`	`koperniqs`	`chat_jokey: what kind of clustering?`
`08:30.01`	`chat_jokey`	`like i want to have like 4000 extensions - something like IP Centrex`
`08:30.17`	`*** join/#asterisk Rawplayer (i=kevin@ipc31055d2.oom-killer.org)`
`08:30.41`	`chat_jokey`	`koperniqs: trying to figure out how many extension a Dual XEON - 3.0Ghz, 4GRAM can handle ..`
`08:31.00`	`chat_jokey`	`based on that wanna do some sizing ..`
`08:32.37`	`koperniqs`	`chat_jokey: ther's a tool called sipsak (sipsak.org) that might help`
`08:39.26`	`chat_jokey`	`koperniqs: lemme have a look`
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`08:43.31`	`*** join/#asterisk DHuang (n=DHuang@mail.medec.com.au)`
`08:43.44`	`DHuang`	`Hi`
`08:44.24`	`DHuang`	`Can someone help me with SER + Asterisk?`
`08:49.51`	`*** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com)`
`08:51.37`	`*** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it)`
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`08:53.00`	`DHuang`	`helo?`
`08:56.42`	`Nico_Bdav`	`hi all`
`08:57.09`	`DHuang`	`hi Nico... can you help me with SER + Asterisk?`
`08:57.10`	`chat_jokey`	`hi DHuang even i am looking for similar stuff`
`08:57.21`	`Nico_Bdav`	`does anyone know a good T1->IP gateway, compatible with asterisk ?`
`08:57.39`	`Nico_Bdav`	`DHuang, no sorry`
`08:57.42`	`chat_jokey`	`Nico_Bdav: are you looking for TDM hardware ?`
`08:58.00`	`DHuang`	`chat_jokey: I see... what I'm trying is to make SIP Client to call each other through SER + Asterisk`
`08:58.03`	`chat_jokey`	`Asterisk itself can act as gateway !`
`08:58.09`	`Nico_Bdav`	`chat_jokey, i want to test asterisk on one site`
`08:58.43`	`Nico_Bdav`	`but i want on another site which already have a PBX to convert T1 outlet to IP`
`08:59.16`	`DHuang`	`Chat: kewl.. just tried a config and work now.. :-p`
`08:59.40`	`chat_jokey`	`DHuang: i am trying to scale asterisk, so its suggested that one uses SER as SIP Proxy and enable it to throw SIP calls into Multiple Asterisk boxes, but i dont seem to find anything relevant online ... can anyone else help me on this ?`
`09:00.18`	`DHuang`	`Chat: search fallover I think is on the original setup doc.`
`09:01.23`	`chat_jokey`	`I have A@H here .. hmm`
`09:02.13`	`DHuang`	`Dam... not working.. ;-(`
`09:03.54`	`*** join/#asterisk tengulre11 (n=tengulre@61.185.224.66)`
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`09:24.35`	`iDunno`	`morning`
`09:24.58`	`A-jay`	`hi`
`09:25.00`	`DHuang`	`Chat: does your Asterisk do the registering or the SER?`
`09:25.06`	`DHuang`	`Morning there.`
`09:25.13`	`A-jay`	`hi`
`09:25.54`	`DHuang`	`I'm trying to figure out how to SER and register on Asterisk so it shows the right HOST IP for the client?`
`09:28.54`	`*** join/#asterisk darkskiez (n=darkskie@bb-194-6-115-241.ukonline.co.uk)`
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`09:30.41`	`Curus`	`Is it possible to dump all session variables from extensions.conf?`
`09:31.40`	`Curus`	`I tried with an AGI script, but I can only get one variable at a time, and only if I know the name`
`09:32.52`	`*** join/#asterisk KriS83 (n=KriS@212.202.141.92)`
`09:33.31`	`RoyK`	`er`
`09:33.43`	`RoyK`	`User disconnected from queue %s while waiting their turn`
`09:33.45`	`RoyK`	`wtf????`
`09:33.53`	`RoyK`	`and noone are put into that queue`
`09:35.40`	`*** part/#asterisk DHuang (n=DHuang@mail.medec.com.au)`
`09:42.17`	`RoyK`	`argh. just upgraded to 1.2.x from 1.0 and now support centres are losing calls. after a while phones stop ringing. people still queueing up..`
`09:46.09`	`*** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au)`
`09:46.24`	`thazza`	`Hey all`
`09:46.50`	`*** join/#asterisk ToTo (n=ToTo@host243-91.pool8260.interbusiness.it)`
`09:48.05`	`Curus`	`There is no way to display all currently set variables in extensions.conf?`
`09:48.18`	`RoyK`	`seems like there's a fsckup somewhere in device state`
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`09:48.44`	`RoyK`	`Curus: iirc it's quite easy to go through all _channel_ vars with an agi script`
`09:55.03`	`Curus`	`How?`
`09:56.14`	`JonR800`	`any way to pass hints between two asterisk servers? I suppose that's a job for SER.`
`09:56.20`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`09:58.38`	`Curus`	`Channel variables don't all get passed to AGI`
`10:01.19`	`*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)`
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`10:03.34`	`RoyK`	`zoa: ping`
`10:06.18`	`zoa`	`pong`
`10:11.20`	`thazza`	`pang`
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`10:17.38`	`riksta`	`if i have a sangoma A101, when i install do i need the PRI or BRI use flags?`
`10:19.01`	`cypromis`	`PRI`
`10:19.44`	`riksta`	`ok ta`
`10:19.54`	`riksta`	`for euroisdn?`
`10:21.02`	`af_`	`how good is snom 320?`
`10:22.17`	`RoyK`	`http://blog.outer-court.com/prejudice/`
`10:22.41`	`*** join/#asterisk A-jay (n=quirc@62.217.245.194)`
`10:22.58`	`Ahrimanes`	`hey denmark is not mentioned, damnit`
`10:24.19`	`koperniqs`	`af_: the display is small and it's relativly expensive`
`10:26.43`	`*** join/#asterisk gvag11 (n=g@ipa124.6.tellas.gr)`
`10:26.48`	`gvag11`	`hi all`
`10:27.13`	`RoyK`	`koperniqs: relativily, yes, unless you mention norway in that sentence`
`10:27.17`	`RoyK`	`er`
`10:27.25`	`gvag11`	`i just moved to Asterisk 1.2.1 and i miss the CUT function, does somebody knows something ?`
`10:27.26`	`RoyK`	`that was a bummer`
`10:27.40`	`RoyK`	`gvag11: read about asterisk variables`
`10:27.52`	`RoyK`	`http://www.voip-info.org/wiki-Asterisk+variables`
`10:28.03`	`RoyK`	`<PROTECTED>`
`10:28.40`	`zoa`	`royk: http://www.asteriskguru.com/tutorials/cut_function.html`
`10:28.47`	`zoa`	`ows, gvag11`
`10:29.23`	`*** join/#asterisk fulgas (n=fulgas@209.8.233.107)`
`10:29.28`	`zoa`	`you need to use SET for it now`
`10:29.43`	`RoyK`	`http://bugs.digium.com/view.php?id=6218`
`10:29.45`	`RoyK`	`:(`
`10:30.55`	`gvag11`	`zoa : ok ... so i use the SET(var=${CUT ... thanks a lot zoa ...`
`10:31.14`	`gvag11`	`royk : thanks ...`
`10:34.21`	`af_`	`mhh what phone is good to use with *?`
`10:34.28`	`af_`	`I used gs but not very satisfied`
`10:35.28`	`*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo)`
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`10:43.08`	`iDunno`	`FFS`
`10:43.12`	`iDunno`	`is it just me...`
`10:43.31`	`iDunno`	`or does it seem entirely insane that you end up in a queuing system when phoning a Telco`
`10:43.39`	`iDunno`	`these people need more staff, ffs.`
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`10:52.13`	`micolous_`	`hey, i'm having some issues using meetme. when i have a caller using the ilbc codec over a iax2 trunk, the sound from them is very jittery, yet they can hear me and other non-ilbc users fine... capturing the output from them, i see that there sound is breaking up... for about 0.02 seconds the sound is fine, then for 0.01 seconds there's no sound... and this goes on and on`
`10:52.33`	`micolous_`	`i'm using the ztdummy kernel module as my timing source`
`10:53.23`	`micolous_`	`i'm wondering if this is something wrong on my end, or a bug. i've tweaked around with the jitterbuffer and that doesn't seem to help; and without the jitterbuffer it's even worse. and it's asterisk 1.2.1`
`10:56.13`	`*** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net)`
`10:57.46`	`h3x`	`its probably because the frame size is different on your codecs`
`10:58.08`	`micolous_`	`yeah, i noticed it doesn't effect ulaw at all`
`10:58.40`	`micolous_`	`but my friend using asterisk@home with meetme doesn't have this issue, and he's using the same codecs and upstream iax providers`
`10:58.56`	`h3x`	`what is he using for zaptel timing`
`10:59.04`	`micolous_`	`the dummy driver`
`10:59.20`	`h3x`	`a@h is prob a different version of asterisk right`
`10:59.29`	`micolous_`	`yeah, i think it might be 1.0`
`11:00.00`	`h3x`	`i seem to remember somebody else having a problem like this with 1.2`
`11:03.56`	`*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@host.190.115.68.195.rev.coltfrance.com)`
`11:04.58`	`tzafrir_laptop`	`asterisk@home is basically a sort of asterisk distribution`
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`11:05.12`	`gvag11`	`hi again ...`
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`11:06.28`	`gvag11`	`i changed the extension.conf and instead of CUT(VAR=... i put SET(Var=${CUT(... but still asterisk (1.2.1 stable) complains the CUT is not register ... Any ideas ?`
`11:09.17`	`micolous_`	`tzafrir_laptop: yeah, i remember helping him set it up in september, so it would be running on asterisk 1.0`
`11:15.49`	`gvag11`	`i changed the extension.conf and instead of CUT(VAR=... i put SET(Var=${CUT(... but still asterisk (1.2.1 stable) complains the CUT is not register ... Any ideas ?`
`11:23.11`	`*** join/#asterisk hhoffman_ (n=hhoffman@tor/session/x-059aa9758068410e)`
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`11:27.20`	`gvag11`	`i changed the extension.conf and instead of CUT(VAR=... i put SET(Var=${CUT(... but still asterisk (1.2.1 stable) complains the CUT is not register ... Any ideas ?`
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`11:37.57`	`Reverend`	`OMFG`
`11:38.16`	`Reverend`	`it's the end of the world!`
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`11:39.03`	`Reverend`	`anyone that uses voicepulse or other IAX2 DT provider, have an issue with there service, where after asterisk has been idle for some time, incomming calls no longer ring in?`
`11:40.56`	`*** join/#asterisk aNaSTaCia_geBeri (n=FiaL@85.108.150.190)`
`11:41.26`	`micolous_`	`i had a similar issue with firefly/freshtel (au), Reverend`
`11:42.07`	`micolous_`	`it was rather annoying to setup, but i eventually kept it happy... i used qualify=no`
`11:44.29`	`Reverend`	`micolous_ thank you, i'll try that`
`11:45.36`	`micolous_`	`but another (ugly) workaround is to have asterisk reload on a cron job every 10-15 minutes... i noticed it would come up after a reload or restart.`
`11:47.31`	`Reverend`	`micolous_ yes, i noticed the same. and i did setup a cron job to do a restart every 20 mins, it is ugly`
`11:49.08`	`micolous_`	`well at least a reload doesn't cut off any active calls`
`11:49.32`	`Reverend`	`neither does "restart gracefully"`
`11:50.02`	`Reverend`	`but if there is an active call when the job runs, it will wait until the call is over to restart, however while waiting for the call to end, no one can make outgoing calls`
`11:50.08`	`Reverend`	`and no other calls can come in`
`11:51.53`	`*** join/#asterisk pa (n=Paolo@unaffiliated/pa)`
`11:54.27`	`*** join/#asterisk cfh (n=luca@82.193.23.6)`
`11:57.06`	`micolous_`	`oh... i was always able to make outgoing calls, just the incoming would be an issue in my case`
`11:57.23`	`micolous_`	`other trunks worked`
`11:57.44`	`Reverend`	`yeh, outgoing calls wasn't a problem until i added the restart gracefully cron job`
`11:58.33`	`cfh`	`when i try to leave a messages on the voice mail asterisk say :`
`11:59.05`	`cfh`	`Executing VoiceMail ...`
`11:59.23`	`cfh`	`and Playing 'vm-theperson'`
`11:59.44`	`cfh`	`and then dont wait and hangup`
`12:00.29`	`micolous_`	`does the asterisk user have write access to /var/spool/asterisk/voicemail/?`
`12:01.33`	`cfh`	`yes`
`12:02.16`	`micolous_`	`hmm... the other thing i'm thinking that could be the case is that the disk is full... other than that I'm out of ideas`
`12:03.24`	`micolous_`	`because they were the two main issues that arose when recording various things on asterisk`
`12:03.30`	`micolous_`	`for me`
`12:04.00`	`cfh`	`no the disk is free`
`12:05.21`	`*** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net)`
`12:05.43`	`*** join/#asterisk pengyong (n=lala@222.185.23.155)`
`12:06.05`	`micolous_`	`it can't be the sound files, as asterisk will simply not play them and skip ahead if they're not found or they don't have access permissions`
`12:07.26`	`cfh`	`I try to reconfigure the sound`
`12:07.58`	`BoRiS`	`Does anyone know if it is possible to get a toll free number for europe (that works in all of germany) that will allow me to pick up a phone in germany and call out through my toll free number without the persons phone who I am using gets billed for the call? (it costs money to call your neighbor in germany).`
`12:07.59`	`mut`	`i've been having a lot of peers unavailable from qualify`
`12:08.00`	`mut`	`Jan 12 07:06:14 NOTICE[26744]: chan_sip.c:10014 sip_poke_noanswer: Peer '9896853317' is now UNREACHABLE! Last qualify: 31`
`12:08.02`	`mut`	`like that`
`12:08.09`	`mut`	`i can login to their ata right now though`
`12:08.17`	`mut`	`ata says they're registered`
`12:08.25`	`*** join/#asterisk flujan (n=flujan@internet.nube.com.br)`
`12:08.31`	`flujan`	`hi all`
`12:08.37`	`mut`	`why would their qualify packets just dissapear, qualify is set to 3000ms too`
`12:09.17`	`flujan`	`I'm new to asterisk and want to know which is the best hardware to buy and learn before I install it in my entire system.`
`12:09.30`	`micolous_`	`BoRiS: i'm not sure if they cover germany, but sipbroker has DID numbers for many international locations that allow you to call ~200 voip providers for the cost of a local phone call`
`12:09.37`	`mut`	`flujan: to do what?`
`12:09.39`	`flujan`	`could someone point some product to me?`
`12:09.47`	`mut`	`you don't need any hardware but a pc to use asterisk`
`12:10.02`	`flujan`	`mut: and about the cards?`
`12:10.13`	`mut`	`well if you plan on using a PRI`
`12:10.18`	`mut`	`or a phone`
`12:10.23`	`mut`	`or an ata`
`12:10.26`	`flujan`	`yes... we intent do use phone`
`12:10.43`	`mut`	`what is it you want to do`
`12:12.11`	`flujan`	`mut: I want to create a pbx with two points`
`12:12.15`	`micolous_`	`flujan: normally you would go and purchase access through a SIP or IAX-based VoIP provider, who would handle incoming calls for you, and allow you to make calls on the PSTN. i don't own any VoIP hardware at all, I'm using software phones... however I may purchase a Sipura unit in the future (which is simply a small box you plug into the network and your phone and this allows you to use VoIP on any analogue telephone)`
`12:12.29`	`flujan`	`and this points communicating through digital phones`
`12:12.53`	`mut`	`well, you plan on buying new phones too? and trunking out a single pri`
`12:12.53`	`mut`	`?`
`12:13.14`	`flujan`	`buying new phones. Actually we use analog ones. :D`
`12:13.29`	`mut`	`you want to keep doing the ananlog thing?`
`12:13.40`	`BoRiS`	`micolous: The biggest problem I am having is how to remove cost for the caller. If I setup a european toll free number and I have someon calling from a land line in germany. Does it cost them money on a per minute basis for them talking on the phone (It costs money to call even your neighbor)?`
`12:13.45`	`flujan`	`micolous_: we will not use a VOIP provider`
`12:14.02`	`flujan`	`micolous_: we will have our own lines. :) we have a E1 here`
`12:14.06`	`mut`	`you're going to use what to connect to the PSTN?`
`12:14.07`	`mut`	`ah ok`
`12:14.27`	`sulex`	`do as5400/as5300 work fine with * and SIP?`
`12:14.40`	`flujan`	`mut: no, we will migrate to digital phones.`
`12:14.56`	`mut`	`so you'll probly just want to get a te110p card for the pri, and then for the phones just to SIP with a plycom phone`
`12:15.16`	`mut`	`can go lower budget on the phones if you want tho`
`12:15.36`	`micolous_`	`BoRiS: well in the end, connecting calls over the PSTN costs someone money. in australia, for a few months someone setup a toll free incoming number so you could call from any australian phone and get onto voip. but that was changed to a 1300 number (untimed local call, anywhere in the country) due to the abuse it got`
`12:16.13`	`flujan`	`mut: to a initial environment... I want just a simple card to run tests and stuff`
`12:16.24`	`micolous_`	`you're likely to find people who can connect calls from the PSTN to VoIP on a tolled number, but without paying money, you're unlikely to get it toll free.`
`12:16.33`	`mut`	`just get a sip phone of some sort`
`12:16.40`	`flujan`	`after I have at least two points working as ramals inside the company we will expand this .`
`12:16.47`	`mut`	`when i initially setup everything`
`12:17.00`	`mut`	`i first just used 2 xten softphones to play with the dialplan and user setup`
`12:17.05`	`*** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac)`
`12:17.15`	`mut`	`no hardware investment other than the pc, which was vmware`
`12:17.19`	`micolous_`	`yeah`
`12:17.36`	`micolous_`	`xten xlite and sjphone are good, free softphones`
`12:17.41`	`BoRiS`	`micolous_: I dont mind paying the euro toll free number and minutes but I just don't want their telephone provider charging them on a per-minute rate for calling my toll free number.`
`12:17.42`	`mut`	`just got for the softphone test`
`12:18.02`	`BoRiS`	`(on a land line)`
`12:18.03`	`micolous_`	`if you really want hardware, the sipura spa-2000 (now the linksys pap-2) is a nice unit, and costs just over 100$ (australian)`
`12:18.03`	`gvag11`	`i changed the extension.conf and instead of CUT(VAR=... i put SET(Var=${CUT(... but still asterisk (1.2.1 stable) complains the CUT is not register ... Any ideas ?`
`12:18.11`	`mut`	`go*`
`12:18.57`	`flujan`	`mut: I'm afraid it works in softwares tests and not to hardware tests`
`12:18.58`	`flujan`	`well`
`12:19.12`	`flujan`	`mut: i will buy a machine and install asterisk`
`12:19.24`	`flujan`	`then connect it with a Ip in my network ...`
`12:19.35`	`mut`	`if software works`
`12:19.38`	`mut`	`hardware works too`
`12:19.49`	`flujan`	`after that I get other two machines and start to talk...`
`12:19.52`	`micolous_`	`BoRiS: ah... I think a toll free number in germany would be free for callers in germany, but probably not other people in europe. however I can't confirm this having no real knowledge of how the EU phone systems work and having never lived there. but i would think you need one toll free number for each country you want to handle callers from`
`12:19.53`	`flujan`	`is it that simple?`
`12:19.59`	`mut`	`yea`
`12:20.04`	`flujan`	`mut: cool`
`12:20.07`	`flujan`	`thanks in advance`
`12:20.08`	`flujan`	`:D`
`12:20.18`	`flujan`	`I will provide this right now...`
`12:20.24`	`flujan`	`See you guys.`
`12:20.25`	`flujan`	`:D`
`12:20.27`	`mut`	`adios`
`12:20.41`	`zoa`	`gvag11: can you paste the exact error message ?`
`12:21.21`	`*** part/#asterisk flujan (n=flujan@internet.nube.com.br)`
`12:22.05`	`*** join/#asterisk da_didi (n=didi@wikipedia/MichaelDiederich)`
`12:23.34`	`benjk`	`micolous_ you can get an international toll free number (country code 800)`
`12:24.00`	`benjk`	`rare and probably expensive (though I don't really know) but they do exist`
`12:24.11`	`gvag11`	`zoa i am afraid that not now cause i am reinstalling asterisk ... But it was like "... CUT not register" and with "show functions i can't see that...`
`12:24.19`	`*** part/#asterisk da_didi (n=didi@wikipedia/MichaelDiederich)`
`12:25.36`	`micolous_`	`benjk: i didn't know about those... but yeah, they would cost a bucketload`
`12:26.03`	`micolous_`	`probably cheaper to have a local toll-free number in each country your company services`
`12:26.37`	`benjk`	`airlines often have those international 800 numbers`
`12:27.58`	`micolous_`	`well, their clients often move around between countries, so such an expense is justifable`
`12:28.41`	`Ikarus`	`If it is just for europe in the European telephone numberspace there is a toll-free catagory`
`12:30.10`	`*** part/#asterisk cfh (n=luca@82.193.23.6)`
`12:30.32`	`*** join/#asterisk NetrixWrk (n=leoem@nat-vlan200.sat.rackspace.com)`
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`12:30.53`	`*** part/#asterisk NetrixWrk (n=leoem@nat-vlan200.sat.rackspace.com)`
`12:31.12`	`Reverend`	`anyone recommend a toll-free service that's better than Kall8 ?`
`12:31.15`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`12:31.22`	`Reverend`	`erm... not 'better' but cheaper?`
`12:31.26`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`12:33.42`	`*** join/#asterisk zotz (n=zotz@24.231.47.175)`
`12:34.41`	`chiardon`	`helllloooo`
`12:36.07`	`gvag11`	`zoa : after uninstall (rm) and install everything fine ... thanks`
`12:36.10`	`gvag11`	`bye`
`12:42.08`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`12:43.40`	`*** join/#asterisk Fubster (i=fubster@jax.metawire.org)`
`12:44.05`	`synthetiq`	`im runnign asterisk on freebsd.....but port 5060 wont open... who knows is 4569 is...any idea why?`
`12:48.58`	`*** join/#asterisk diego_br (n=diego@200.208.241.178)`
`12:49.29`	`*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)`
`12:49.59`	`*** join/#asterisk flujan (n=flujan@internet.nube.com.br)`
`12:51.09`	`*** join/#asterisk SERGEUS (n=s@195.112.98.13)`
`12:53.14`	`*** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie)`
`12:53.24`	`*** join/#asterisk zapotecz (n=surfer@217.201.198.236)`
`12:53.34`	`zapotecz`	`good morning`
`12:53.45`	`zapotecz`	`some can help me with a mess ?`
`12:54.01`	`zapotecz`	`I'm tring from one week to do this extension`
`12:54.11`	`zapotecz`	`but I really don't know what do for solve :(`
`12:54.20`	`Reverend`	`what's not working right?`
`12:54.25`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)`
`12:54.25`	`chiardon`	`no d-channel available. Using primary channel as d-channel anyway . . .some ideas about what happen here?TIA`
`12:54.35`	`zapotecz`	`i've to dial "69phonenumber#"`
`12:54.48`	`zapotecz`	`from a PRI zapata`
`12:54.59`	`zapotecz`	`but asterisk take the # as "end of call"`
`12:55.09`	`zapotecz`	`and doesn't call my message box`
`12:55.55`	`zapotecz`	`I really don't know how to solve this :(`
`12:55.56`	`*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)`
`12:56.07`	`zapotecz`	`i've tried with a disa , and put the number from another phone`
`12:56.19`	`zapotecz`	`i've tried trough sip`
`12:56.19`	`zapotecz`	`but nothing :(`
`12:58.42`	`*** join/#asterisk RoyK (n=roy@host-81-191-145-46.bluecom.no)`
`12:59.18`	`zapotecz`	`i've tried with google`
`12:59.20`	`zapotecz`	`but no answer`
`13:00.24`	`*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)`
`13:00.25`	`*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)`
`13:03.17`	`mut`	`whats your dialstring?`
`13:03.36`	`zoa`	`zapotecz: did you try features.conf ?`
`13:03.56`	`zapotecz`	`i've to dial "69003905523552#"`
`13:04.21`	`mut`	`ya but thats not what your dialplan says..`
`13:05.39`	`zapotecz`	`mhhh`
`13:05.46`	`zapotecz`	`features.conf only work in local`
`13:05.58`	`zoa`	`that i dont know`
`13:06.01`	`zoa`	`i never used it`
`13:06.18`	`zapotecz`	`I do that`
`13:06.45`	`zapotecz`	`exten=> 555,1,dial(zap/g1/69003905523552#"`
`13:06.56`	`zapotecz`	`with drive syntax`
`13:07.24`	`zapotecz`	`and I receive a "no one avaiable to answer"`
`13:07.43`	`zapotecz`	`i've tryed also with a "normal" pabx and the dialstring work`
`13:08.11`	`zapotecz`	`I suppose that asterisk recognize the final pound/hash as "stop dialstring buffering"`
`13:08.22`	`zapotecz`	`or in /etc/asterisk/zaptel.conf`
`13:08.37`	`zapotecz`	`in the format number`
`13:09.03`	`mut`	`um`
`13:09.34`	`*** join/#asterisk coppice (n=chatzill@166.168.17.210.dyn.pacific.net.hk)`
`13:09.37`	`mut`	`so why do you have * dialing a zap chan with that number if you want to access voicemail..`
`13:10.00`	`zapotecz`	`mhh but is a voicemail gived by the carrier`
`13:10.07`	`zapotecz`	`not the asterisk voicemail`
`13:10.18`	`zapotecz`	`is the phone provider that give this service`
`13:10.35`	`mut`	`ah`
`13:10.39`	`zapotecz`	`and all the "internal users" have this voicemail memo`
`13:10.46`	`zapotecz`	`is a big trouble for me :(`
`13:11.16`	`zapotecz`	`but i've really no idea how to bypass this`
`13:12.09`	`flujan`	`hi all`
`13:12.24`	`flujan`	`I asked some time ago about cheap hardware to test asterisk`
`13:12.24`	`flujan`	`:D`
`13:12.32`	`BoRiS`	`hi coppice`
`13:12.34`	`flujan`	`now I return with the same question.`
`13:12.46`	`flujan`	`mi boss REALLY WANT HARDWARE...`
`13:12.50`	`coppice`	`hi`
`13:13.12`	`flujan`	`I already said that we only need the computer and the softphone`
`13:13.24`	`flujan`	`and he still want to see hardware stuff`
`13:13.30`	`flujan`	`so, here I am.`
`13:13.31`	`mut`	`heh`
`13:13.32`	`mut`	`get an ata`
`13:13.40`	`mut`	`sipura 1001`
`13:13.41`	`flujan`	`mut: hi... me again!`
`13:13.42`	`flujan`	`:D`
`13:13.49`	`mut`	`they are like $60`
`13:13.55`	`mut`	`usd`
`13:14.17`	`mut`	`zapotecz: what happens when ya dial that then? instant hangup? or do ya hear anything?`
`13:14.30`	`mut`	`and can ya set verbose 5 and show me the output when ya call it`
`13:14.36`	`zapotecz`	`yes`
`13:14.41`	`zapotecz`	`instant hangup`
`13:14.49`	`zapotecz`	`and the answer`
`13:14.54`	`zapotecz`	`"no one avaiable"`
`13:15.00`	`flujan`	`mut: which ata`
`13:15.20`	`mut`	`sipura 1001`
`13:15.29`	`mut`	`zapotecz: can ya get me that debug output?`
`13:15.31`	`mut`	`www.pastebin.ca`
`13:15.33`	`mut`	`paste in there`
`13:15.59`	`synthetiq`	`im runnign asterisk on freebsd.....but port 5060 wont open... who knows is 4569 is...any idea why?`
`13:16.02`	`mut`	`flujan: http://www.voipsupply.com/product_info.php?products_id=320`
`13:16.04`	`gambolputty`	`Is call duration stored in a variable?`
`13:16.33`	`[TK]D-Fender`	`flujan: SPA-2002 $70 = 2 FXS ports. Describe your setup : # lines (what kind), # of phones (how many need speakerphone, expected usage, etc)`
`13:17.05`	`mut`	`i think his boss just wants to see some hardware phones working over voip`
`13:17.14`	`mut`	`then they'll go for the good stuff`
`13:17.19`	`mcquaid`	`hello, i was trying to get asterisk to work with my voip provider (vbuzzer.com). so far i can make outgoing calls to pstn lines via voip but incoming calls have no audio in either direction`
`13:17.26`	`mcquaid`	`i can see the rtp traffic, but hear nothing. I am running asterisk behind a firewall which i don't have access to.`
`13:17.42`	`*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)`
`13:17.46`	`[TK]D-Fender`	`:/`
`13:17.50`	`mut`	`mcquaid: call them?`
`13:17.55`	`mcquaid`	`however, i can get the sip clients to directly connect to my voip provider and make/receive calls with full audio`
`13:17.58`	`mcquaid`	`call who?`
`13:18.04`	`mut`	`vbuzzer`
`13:18.10`	`mcquaid`	`uh why?`
`13:18.11`	`*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)`
`13:18.19`	`mut`	`cause it's your service provider`
`13:18.29`	`mcquaid`	`their service works fine as i just mentioned in other sip clients (linphone, twinklephone etc)`
`13:18.59`	`mcquaid`	`but the way i got them to work is not by using nat or stun, but by using outboundproxy`
`13:19.03`	`mcquaid`	`otherwise they don't work either`
`13:19.22`	`mcquaid`	`asterisk seems to support outboundproxy but the documentation is pretty thin on this`
`13:19.54`	`mcquaid`	`it was in chan_sip2 last year, and most things have got promoted to chan_sip, and i see outboundproxy in the c code`
`13:20.12`	`mcquaid`	`but using them in my sip.conf has no effect`
`13:20.13`	`*** join/#asterisk nfi\|ermes (n=ermsewrk@217.220.121.62)`
`13:20.22`	`flujan`	`mut: we want internal communication using ramals... this will be the first test`
`13:20.30`	`mcquaid`	`mut, why would you assume it's my provider?`
`13:20.35`	`mut`	`ramals?`
`13:20.41`	`flujan`	`sorry`
`13:20.44`	`mut`	`mcquaid: so i don't have to help ya ;)`
`13:20.47`	`mcquaid`	`heh`
`13:20.48`	`*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)`
`13:21.07`	`mut`	`bbmin gotta go start a pot of coffee`
`13:21.34`	`mcquaid`	`i've read a little about siproxd, is any one familar with siproxd? curious if it would help in this situation`
`13:23.18`	`flujan`	`extensions lines`
`13:23.32`	`flujan`	`i dunno the english work for this`
`13:23.35`	`flujan`	`strands maybe`
`13:23.38`	`flujan`	`:P`
`13:29.30`	`*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)`
`13:30.55`	`*** join/#asterisk Lathos42 (n=Lathos42@adsl-69-210-24-249.dsl.lgnnmi.ameritech.net)`
`13:31.32`	`Lathos42`	`Good morning`
`13:32.13`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`13:34.35`	`*** join/#asterisk Rawplayer (i=kevin@ipc31055d2.oom-killer.org)`
`13:35.09`	`mut`	`flujan: what language is ramals?`
`13:37.07`	`*** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com)`
`13:39.19`	`flujan`	`mut: brazilian portuguese. :)`
`13:40.03`	`mut`	`hm`
`13:40.05`	`mut`	`ah well`
`13:40.17`	`mut`	`babelfish can only translate it to AS in dutch`
`13:40.22`	`mut`	`other than that it`
`13:40.25`	`mut`	`doesn't translate`
`13:40.35`	`flujan`	`hold on`
`13:40.54`	`mcquaid`	`mut, do you have any suggestions on my issue?`
`13:40.59`	`*** part/#asterisk micolous_ (n=michael@ppp251-29.static.internode.on.net)`
`13:41.01`	`mut`	`you're going to want to get some polycom phones if you want to test out a real world thing`
`13:41.27`	`mut`	`http://www.voipsupply.com/product_info.php?products_id=757`
`13:41.32`	`mut`	`somethin like these guys`
`13:43.00`	`tdonahue`	`good morning all`
`13:43.30`	`warthawg`	`voicemail doesn't seem to like my password`
`13:43.40`	`mut`	`you don;t use any options in the dial string do ya?`
`13:43.45`	`*** join/#asterisk nvrs (n=RUR@65.93.97.70)`
`13:43.48`	`tdonahue`	`does anyone use 1.2 on freebsd? we are having issues getting it to bind to port 5060 for sip`
`13:43.49`	`[TK]D-Fender`	`Considerably cheaper source for Polycom phones - http://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-36931336704.htm`
`13:43.53`	`mut`	`and i asked for a verbose output of the dial`
`13:44.36`	`*** join/#asterisk nvrs (n=RUR@65.93.97.70)`
`13:44.40`	`mut`	`they're the same price..`
`13:45.06`	`*** join/#asterisk mistral (i=mistral@jstevenson.plus.com)`
`13:45.30`	`[TK]D-Fender`	`mut... look closer. The Atacomm one is $113. Its the same price when you get the PoE adapter INCLUDED.`
`13:45.51`	`hackeron`	`hey, I have a strange problem, all phones are getting "invalid password" when the correct password is dialed for both meetme and voicemail - any ideas?`
`13:46.20`	`*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)`
`13:46.22`	`*** join/#asterisk devoider (n=racal@gw.01063telecom.de)`
`13:46.28`	`devoider`	`hi fellas`
`13:46.35`	`*** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it)`
`13:47.04`	`Mimmus`	`to try most recent zaptel/pri, what CVS do I need to checkout?`
`13:47.11`	`fugitivo`	`atacomm doesn't accept credit cards??`
`13:47.21`	`fugitivo`	`oh yes`
`13:48.11`	`mut`	`what ever happened to atacomm`
`13:48.11`	`trixter`	`svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2`
`13:48.11`	`trixter`	`svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2`
`13:48.14`	`[TK]D-Fender`	`hackeron : Pastebin your voicemail.conf context, and the extensions.conf entexs that call it.`
`13:48.17`	`mut`	`or is that you fender?`
`13:48.17`	`trixter`	`that should be the most current SVN version`
`13:48.20`	`Mimmus`	`ok, trixter, tahnk you`
`13:48.30`	`[TK]D-Fender`	`is what me?`
`13:48.31`	`flujan`	`mut: sorry, my boss was here`
`13:48.43`	`mut`	`heh`
`13:48.43`	`flujan`	`mut: well, actually we have 3 E1 channels`
`13:48.45`	`_Sam--`	`<PROTECTED>`
`13:48.47`	`_Sam--`	`er`
`13:48.51`	`mut`	`wow big company?`
`13:48.58`	`devoider`	`did someone ever experience missing field values while writing CDR-data? like an "empty" lastapp or dst field`
`13:48.58`	`Mimmus`	`if I have Sangoma, do I need to run wanpipe config before compiling CVS?`
`13:49.06`	`flujan`	`mut: and 140 internal telephones ( aka ramals :P )`
`13:49.08`	`warthawg`	`it looks to me like asterisk can understand my bt-101 fine for everything except voicemail, the console shows password entered is ''`
`13:49.26`	`sivana`	`Mimmus: you should read their docs, but I think you need to compile zaptel first`
`13:49.31`	`hackeron`	`[TK]D-Fender: it happens for meetme too, isnt extensions.conf probably to blame? -- http://rafb.net/paste/results/gfU2eZ43.html`
`13:49.34`	`sivana`	`then recompile it after you run the wanpipe config`
`13:49.35`	`mut`	`and it's all analog right now?`
`13:49.36`	`mut`	`man`
`13:49.37`	`flujan`	`and we want the the less expensive solution to use Asterisk`
`13:49.39`	`mut`	`that SUCKS`
`13:49.47`	`Mimmus`	`sivana: wanpipe driver install patches zaptel`
`13:49.50`	`flujan`	`yes.`
`13:49.53`	`flujan`	`it's all analog`
`13:49.54`	`flujan`	`:(`
`13:50.04`	`*** join/#asterisk RoyK (n=roy@host-81-191-145-46.bluecom.no)`
`13:50.11`	`flujan`	`we want digital and we want the less expensive solution`
`13:50.20`	`mut`	`get those polycom poe phone`
`13:50.33`	`[TK]D-Fender`	`Mimmus : You need to compile zaptel first, then wanpipe, then zaptel AGAIN.`
`13:50.35`	`flujan`	`my boss wants me to try firts change the internal communication`
`13:50.49`	`Mimmus`	`[TK]D-Fender: ah, ok, I remember now... thanks`
`13:50.52`	`flujan`	`and later on test using the E1 channels`
`13:51.01`	`flujan`	`only then we will migrate the entire system...`
`13:51.09`	`[TK]D-Fender`	`hackeron : I need to see the extensions.conf part that calls it...`
`13:51.16`	`sivana`	`Mimmus: isn't that what I just said? :)`
`13:51.20`	`flujan`	`mut: So, I am here asking for help. :D`
`13:51.43`	`Mimmus`	`sivana: yes yes, thank you again`
`13:51.44`	`flujan`	`mut we want first make two internal phones communicate throught asterisk`
`13:51.51`	`*** join/#asterisk amir (n=amir@gentoo/developer/amir)`
`13:52.04`	`warthawg`	`does anyone have voicemail working on openwrt?`
`13:52.07`	`sivana`	`Mimmus: after you have zaptel/wanpipe installed, then do *`
`13:52.14`	`hackeron`	`[TK]D-Fender: http://rafb.net/paste/results/dNQEYT25.html < its the one you gave me, but I tried with VoicemailMain() too where it would also reject the password`
`13:52.17`	`sivana`	`or libpri if you need it`
`13:52.37`	`flujan`	`mut: then making call using the throught the E1 channels to the world. :P`
`13:52.39`	`Mimmus`	`sivana: do I need to recompile 'full' asterisk to try current CVS for zaptel/libpri?`
`13:52.48`	`flujan`	`mut: what did you suggest?`
`13:52.58`	`mut`	`flujan: get those polycom poe phones`
`13:53.01`	`sivana`	`Mimmus: not sure I understand`
`13:53.15`	`mut`	`i can't believe ya use 3 e1's for 140 phones tho`
`13:53.19`	`sivana`	`Mimmus: you should have the same version of zaptel, libpri, asterisk`
`13:53.24`	`mut`	`telemarketing company or something`
`13:53.39`	`Mimmus`	`sivana: I'm having problems with answer detection and I'd like to try current CVS of zaptel/libpri to solve the issue`
`13:53.52`	`Mimmus`	`sivana: I have Asterisk 1.2.1`
`13:54.00`	`sivana`	`Mimmus: then you should stay with the same version for all`
`13:54.08`	`mut`	`mcquaid????`
`13:54.17`	`flujan`	`mut: thanks`
`13:54.17`	`mcquaid`	`yes???`
`13:54.20`	`Mimmus`	`sivana: well, I understand`
`13:54.22`	`[TK]D-Fender`	`hackeron : Heres the problem : exten => *98,2,VoicemailMain(${CALLERID(number)$}@default) its the extra $ before }`
`13:54.33`	`flujan`	`mut: http://www.voip-info.org/wiki-Polycom+Phones`
`13:54.42`	`flujan`	`mut: is that correct?`
`13:54.45`	`sivana`	`Mimmus: if you want to do CVS zaptel/libpri and 1.2.1 asterisk, you run the risk of problems of new functions`
`13:54.55`	`sivana`	`or changed code`
`13:55.08`	`mut`	`http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-48102218496.htm`
`13:55.19`	`Mimmus`	`sivana: ok thanks, I didn't know it, I thought that zaptle/libpri was only 'drivers'`
`13:55.33`	`sivana`	`Mimmus: they are, but they work together`
`13:56.04`	`hackeron`	`[TK]D-Fender: what do you? - that looks like what I have in the pastebin`
`13:56.40`	`mut`	`whats ya company do flujan?`
`13:56.59`	`hackeron`	`[TK]D-Fender: oh, I get it, I removed the $ -- but it still saying login incorrect`
`13:57.00`	`Mimmus`	`sivana: does I need "TDMV DCHAN Native HDLC Support" in Sangoma conf?`
`13:57.02`	`mcquaid`	`mut, were you posting something to me that I missed?`
`13:57.05`	`[TK]D-Fender`	`hackeron : you need to remove the extra $. heres the corrected version : exten => *98,2,VoicemailMain(${CALLERID(number)}@default)`
`13:57.19`	`mut`	`mcquaid: ya.. still asking for that call dump`
`13:57.22`	`sivana`	`Mimmus: probably good idea, do you have a PRI?`
`13:57.26`	`[TK]D-Fender`	`Mimmus : Yes, you want that done in hardware.`
`13:57.39`	`Mimmus`	`sivana: yes, E1 PRI in Italy`
`13:57.43`	`mcquaid`	`ah sorry didn't see that one sec`
`13:57.43`	`sivana`	`ya`
`13:58.16`	`hackeron`	`[TK]D-Fender: still says login incorrect :( - I dial the pin, it then waits for a few seconds, then says incorrect. Do I need to dial # after the pin or something because it just waits no matter what I do and then says login incorrect`
`13:58.30`	`tzanger`	`morning`
`13:58.37`	`devoider`	`i am having trouble with empty values in the generated CDRs, like an empty "dst" field .. or lastapp, this should never happen .. but it does. any similar problems seen?`
`13:58.38`	`flujan`	`mut: it's a call center`
`13:58.56`	`*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)`
`13:58.56`	`tzanger`	`hackeron: turn on debugging and verify that you're seeing the dtmf digits you're pressing`
`13:58.57`	`hackeron`	`[TK]D-Fender: oh, I'm seeing Incorrect password '' for user '7662' (context = default)`
`13:59.07`	`mut`	`i guess that'de make sense then`
`13:59.09`	`hackeron`	`tzanger: I'm not, its getting nothing`
`13:59.10`	`flujan`	`mut: work recruiting candidates to jobs in another companies... ( I hope I made myself clear ... )`
`13:59.25`	`flujan`	`mut: :)`
`13:59.32`	`mut`	`find me a job`
`13:59.33`	`mut`	`i could use one`
`13:59.40`	`tzanger`	`hackeron: using SIP?`
`13:59.44`	`hackeron`	`tzanger: yes`
`13:59.45`	`sivana`	`hackeron: do you have inband or rfc selected?`
`13:59.57`	`tzanger`	`hackeron: using anything but ulaw/alaw?`
`13:59.57`	`flujan`	`mut: for sure... Where are you from? We just work in Brazil. :D`
`14:00.08`	`mut`	`usa heh`
`14:00.15`	`hackeron`	`sivana: on this phone nat=yes, tried on local phones too, didnt work`
`14:00.21`	`sivana`	`hehe`
`14:00.23`	`chiardon`	`Hello`
`14:00.24`	`hackeron`	`sivana: I mean I can dial an exntension and it works fine`
`14:00.28`	`Mimmus`	`sivana: have you idea why my asterisk doesn't detect answer with some rare numbers?`
`14:00.33`	`[TK]D-Fender`	`hackeron : pastebin your phone def as well`
`14:00.37`	`hackeron`	`tzanger: nope, its ulaw`
`14:00.37`	`warthawg`	`hackeron: what kind of phone, our problems sound similar`
`14:00.41`	`mcquaid`	`mut, http://pastebin.ca/36585`
`14:00.44`	`tzanger`	`hackeron: sounds like you're either using a compressed voice codec and inband (doesn't work) or you're expecting inband and the phone's sending rfc2833, or vice-versa`
`14:00.44`	`mut`	`well maybe if ya find me something lucrative enough i'll move to brazil`
`14:00.46`	`flujan`	`mut: sorry... :(`
`14:00.47`	`hackeron`	`warthawg: GXP-2000`
`14:01.01`	`mut`	`i wouldn't mind moving for a few years`
`14:01.01`	`sivana`	`Mimmus: no :)`
`14:01.13`	`mut`	`since i've never even been out of michigan before it'de be cool`
`14:01.17`	`hackeron`	`tzanger: errr, I can make calls fine, to other phones behind NAT, and the echo test works`
`14:01.17`	`warthawg`	`hackeron: i just solved my problem on grandstream`
`14:01.20`	`hackeron`	`tzanger: and its ulaw`
`14:01.22`	`flujan`	`mut: for sure`
`14:01.25`	`hackeron`	`warthawg: how?`
`14:01.30`	`mut`	`mcquaid: and the asterisk debug`
`14:01.39`	`tzanger`	`hackeron: you are not listening`
`14:01.42`	`sivana`	`hackeron: look in your sip.conf, what do you have for dtmf for that user`
`14:01.42`	`flujan`	`mut: I will go to irvine next summer! :)`
`14:01.43`	`chiardon`	`no d channels available.Using primary channel 16 as d channel anyway!What's the issue here?`
`14:01.43`	`warthawg`	`just a sec lstening to messages`
`14:01.49`	`mut`	`i just wanted ya to set verbose 5`
`14:01.50`	`mcquaid`	`sorry how do i generate that?`
`14:01.51`	`tzanger`	`hackeron: making calls and echotest do not need dtmf`
`14:01.57`	`mut`	`and get the dialplan dump`
`14:02.05`	`[TK]D-Fender`	`hackeron : We need to confirm your DTMF mode. just because you can dial does not mean DTMF works while you're IN a call.`
`14:02.08`	`tzanger`	`hackeron: whatever you have selected for DTMF generation, switch it`
`14:02.09`	`hackeron`	`tzanger: oh?`
`14:02.19`	`[TK]D-Fender`	`hackeron : Pastebin your sip.conf`
`14:02.32`	`sivana`	`hehe and slow down and read :)`
`14:02.34`	`Mimmus`	`is there anyone on the earth who is able to debug PRI?`
`14:02.52`	`warthawg`	`hackeron: i went into the grandstream admin console and checked SIP/Info for the DTMF signalling`
`14:02.57`	`hackeron`	`[TK]D-Fender: I dont have dtmf there, let me just try that quickly`
`14:03.05`	`tzanger`	`Mimmus: yep, what's the trouble`
`14:03.10`	`Mimmus`	`Itried also to ask for paid support at Digium but nope`
`14:03.20`	`[TK]D-Fender`	`hackeron : "dtmfmode=rfc2833"`
`14:03.22`	`tzanger`	`Mimmus: I find that very hard to believe`
`14:03.37`	`Mimmus`	`tzanger: my * doesn't detect answer with some (rare) numbers, especially automatic responders`
`14:03.41`	`warthawg`	`now it works, what i dont understand is why it understood extensions and outbound numbers just fine, but not vm password`
`14:04.00`	`chiardon`	`Are the Asterisk cards made with one of this chips?: * HFC USB`
`14:04.00`	`chiardon`	`<PROTECTED>`
`14:04.00`	`chiardon`	`<PROTECTED>`
`14:04.00`	`chiardon`	`<PROTECTED>`
`14:04.04`	`mut`	`man is it more busy than usual this mornin or what`
`14:04.08`	`Mimmus`	`tzanger: it rings indefinitely`
`14:04.36`	`hackeron`	`[TK]D-Fender: tzanger: warthawg: sivana: kick ass, that worked! - but you're saying if we switch to G726 or G729 it wont work anymore?`
`14:05.42`	`tzanger`	`Mimmus: use pri debug to verify that your telco is sending back an answer. many automatic responders are on PRIs themselves and do NOT answer the line to save toll charges (you can do this, you only get one-way audio)`
`14:05.43`	`warthawg`	`hackeron: i am a clueless noobie, i just kept hacking til it worked for me`
`14:05.50`	`sivana`	`hehe`
`14:06.31`	`hackeron`	`warthawg: well, thats what hacking is all about -- going l33t stuff by accident :)`
`14:06.31`	`warthawg`	`hehehe`
`14:06.32`	`tzanger`	`hackeron: you will have DTMF working with any codec if you're using RFC2833. Inband only works with ulaw/alaw`
`14:06.38`	`Mimmus`	`tzanger: I tried to examine pri debug output but it is too difficult for normal people`
`14:06.41`	`[TK]D-Fender`	`hackeron : the voice Codec in this case has nothing to do with how DTMF is passed.`
`14:06.52`	`tzanger`	`Mimmus: just break it down`
`14:07.06`	`warthawg`	`[TK]D-Fender, why does it decode dtmf elsewhere (extensions and phone numbers) but not in vm?`
`14:07.06`	`tzanger`	`what i tend to do is copy/paste it and then turn off line wrapping -- that seems to help`
`14:07.20`	`[TK]D-Fender`	`hackeron : rfc2833 sends the DTMF data outside of teh voice stream and it inserted back in at the ENDPOINT.`
`14:07.29`	`sivana`	`tzanger: what dtmf do I use for wav?`
`14:07.30`	`tzanger`	`warthawg: it's not decoding it. when you dial iwth a sip phone it's not sending dtmf digits as audio, it's sending a text messgae to the * box with the #`
`14:07.42`	`warthawg`	`tzanger ah, thanks`
`14:07.45`	`Mimmus`	`tzanger: I don't understand well the meaning of "break it down".. sorry... my english is bad`
`14:07.48`	`tzanger`	`sivana: seriously, go find a way for me to make piles of money with you rhard work.`
`14:07.58`	`sivana`	`heh`
`14:08.00`	`hackeron`	`[TK]D-Fender: tzanger: what about DTMF via SIP INFO?`
`14:08.05`	`[TK]D-Fender`	`warthawg : because its your PHONE doing the dialing. it doesn't need sound from its own keypad, you just push buttons! Once you get to another device however you need to send IT the data somehow.`
`14:08.06`	`tzanger`	`Mimmus: break it down == study it and try to understand the organization of it`
`14:08.10`	`sivana`	`tzanger: already working on it`
`14:08.22`	`tzanger`	`hackeron: that will work with compressed voice codecs too`
`14:08.34`	`hackeron`	`tzanger: great, thanks!`
`14:08.36`	`warthawg`	`[TK]D-Fender, thanks. who knew telephony was such a black art`
`14:08.40`	`[TK]D-Fender`	`hackeron : SIP INFO can work as well, but use rfc2833 is you can. its a question of what your phone can support.`
`14:08.53`	`[TK]D-Fender`	`warthawg : not that hard really...`
`14:09.10`	`hackeron`	`[TK]D-Fender: hmm, ok I will, thanks`
`14:09.14`	`mcquaid`	`mut, here's my dialplan and sip.conf http://pastebin.ca/36588`
`14:09.19`	`tzanger`	`warthawg: wait until you play with PRI debugging, zapata echo and oddball hangup detection :-)`
`14:09.23`	`warthawg`	`[TK]D-Fender, i've learned more stuff about it in the past 3 days than in my entire life`
`14:09.26`	`mcquaid`	`mut, how do i generate the asterisk debug?`
`14:09.35`	`mut`	`mcquaid`
`14:09.47`	`mut`	`asterisk -r`
`14:09.47`	`Mimmus`	`tzanger: oh, well... there is a sad "!! < Unknown IE 1562 (len = 6)`
`14:09.47`	`mut`	`set verbose 5`
`14:09.47`	`warthawg`	`tzanger not me! :)`
`14:09.47`	`mcquaid`	`oh`
`14:09.47`	`mut`	`then dial the extension`
`14:09.47`	`*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)`
`14:09.51`	`mut`	`and a bunch of crap shows`
`14:09.53`	`tzanger`	`Mimmus: ok, don't worry about that just yet but that is important`
`14:09.54`	`jimbalcomb`	`is using rfc2833 instead of SIPinfo generally considered a better way to go?`
`14:10.30`	`BoRiS`	`grandstream console?`
`14:10.40`	`*** part/#asterisk flujan (n=flujan@internet.nube.com.br)`
`14:11.01`	`mcquaid`	`i've been running asterisk as: asterisk -vvvvc, when i try -r i get:`
`14:11.03`	`Mimmus`	`tzanger: not important? ok, well. And "Progress Description: Inband information or appropriate pattern now available. (8) "`
`14:11.13`	`mcquaid`	`unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?`
`14:11.14`	`*** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net)`
`14:11.24`	`warthawg`	`BoRis: ip address of phone`
`14:11.25`	`mcquaid`	`asterisk is on the same box here`
`14:11.41`	`Reverend`	`mcquaid, asterisk isn't running, or is trying to close, or locked up`
`14:11.41`	`BoRiS`	`mcquaid: You need to start asterisk with safe_asterisk script to use asterisk -r`
`14:12.31`	`warthawg`	`CoolAcid, it is still working`
`14:12.40`	`warthawg`	`sorry, let me restate that`
`14:12.45`	`jimbalcomb`	`BoRiS I don't believe that is exactly correct.`
`14:12.46`	`warthawg`	`coolio, it is still working`
`14:13.08`	`mcquaid`	`ok that worked`
`14:13.11`	`mcquaid`	`doesn't list much though`
`14:13.21`	`jimbalcomb`	`warthawg: whats the scoop on switch the DTMF option?`
`14:13.49`	`devoider`	`i assume no one ever experienced trouble with his/her CDRs missing values ?!`
`14:14.11`	`warthawg`	`jimbalcomb, it works with the phone set to either sip/info or rfc2833`
`14:14.43`	`mcquaid`	`mut, http://pastebin.ca/36590`
`14:14.45`	`Cresl1n`	`mimmus: I just responded to your bugnote`
`14:14.57`	`*** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca)`
`14:15.09`	`BoRiS`	`Thats normal`
`14:15.16`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`14:15.24`	`Cresl1n`	`mimmus: it's not a bug`
`14:15.26`	`devoider`	`ne is klar`
`14:15.28`	`mut`	`mcquaid: set verbose 5 then make the call`
`14:15.30`	`BoRiS`	`Thats normal. Try dialing a number or type "sip show channels".`
`14:15.33`	`mut`	`it should output stuff`
`14:15.40`	`Mimmus`	`Cresl1n: I'm seeing... but how is it possible!!!`
`14:15.44`	`mut`	`mcquaid: type in the console 'set verbose 5'`
`14:15.45`	`sivana`	`tzanger: you busy on Sat/Sun?`
`14:15.57`	`[TK]D-Fender`	`jimbalcomb : Both INFO and rfc2833 work out of band and I guess rate the same. Its a question of picking on your phone supports.`
`14:15.58`	`jimbalcomb`	`warthawg: was there something that led you to switching?`
`14:16.04`	`Cresl1n`	`Mimmus: it's pretty simple, some endpoints don't send CONNECT until really late into the call`
`14:16.20`	`warthawg`	`jimbalcomb: it didn't work in the default setting`
`14:16.32`	`Mimmus`	`Cresl1n: in fact, it is a toll-free number of my telco. And is there no workaround?`
`14:16.32`	`*** join/#asterisk javar (n=javar@Dynamic-IP-cr20011863171.cable.net.co)`
`14:16.35`	`[TK]D-Fender`	`jimbalcomb : Sipura devices use INFO, so thats what I pick for them. Most devices use rfc2833. Cheap junk uses inband :)`
`14:16.42`	`jimbalcomb`	`wathawg: ok, gotcha.`
`14:16.47`	`Cresl1n`	`mimmus: some companies (i.e. fedex) let you navigate their entire IVR before they send a connect`
`14:16.57`	`Cresl1n`	`mimmus: nope, nothing to get around it`
`14:17.05`	`warthawg`	`jimbalcomb, it started out set to in-audio`
`14:17.16`	`Mimmus`	`Cresl1n: but phone rings, I don't hear IVR`
`14:17.26`	`javar`	`somebody know, how insert this line, exten => s,n,Set(TIMEOUT(digit)=5) , on a table for ARA`
`14:17.31`	`jimbalcomb`	`[TK]D-Fender: ok, i am taking over an Asterisk admin position and am having trouble finding information about 'best practices' and the 'why'`
`14:17.38`	`mcquaid`	`ok`
`14:17.40`	`Cresl1n`	`Mimmus: if phone rings, it doesn't mean it's answered`
`14:18.00`	`warthawg`	`jimbalcomb, should be an exciting job :)`
`14:18.03`	`jimbalcomb`	`[TK]D-Fender: is there reason to go with either given the phone supports both SIP and rfc?`
`14:18.10`	`konfuzed`	`jimbalcomb: 'why' what`
`14:18.34`	`cypromis`	`o/w 14`
`14:18.39`	`Mimmus`	`Cresl1n: I will be forced to remove my Asterisk!`
`14:18.49`	`[TK]D-Fender`	`jimbalcomb : SIP is the general protocol, rfc2833 is a FEATURE describing how DTMF will be passed.`
`14:18.52`	`jimbalcomb`	`warthawg: yeah, I'm pretty freaked out. Spent the first two weeks restructure the networking and fixing the busted ass VLAN setup. now im dealing with all day long jitter, echo, and dropped call complaints.`
`14:18.55`	`Cresl1n`	`mimmus: what are you talking about?`
`14:19.08`	`h3x`	`creslin: thats some bullshit`
`14:19.15`	`h3x`	`you dotn have a 2 way audio path to send them DTMFs`
`14:19.18`	`h3x`	`until they supervise`
`14:19.23`	`javar`	`somebody know, how insert this line, exten => s,n,Set(TIMEOUT(digit)=5) , on a table for ARA`
`14:19.26`	`Mimmus`	`Cresl1n: if I have problems like this, surely someone will complain and I will be forced to remove Asterisk!`
`14:19.29`	`h3x`	`so you cant navigate anything until its fully answered`
`14:20.21`	`zoa`	`h3x, whats the problem ?`
`14:20.35`	`h3x`	`Cresl1n mimmus: some companies (i.e. fedex) let you navigate their entire IVR before they send a connect`
`14:20.35`	`h3x`	`Cresl1n mimmus: nope, nothing to get around it`
`14:20.42`	`warthawg`	`ok, as soon as i can figure out how to get the message indicate to light up on the bt-101, i am going to call this a wrap`
`14:20.43`	`jimbalcomb`	`[TK]D-Fender: ok, yeah i think i got just wording my question terribly. i was wondering if there is a reason to send DTMF via rfc2833 or SIPinfo?`
`14:20.45`	`*** join/#asterisk Redfury (n=bharatsa@203.109.101.36)`
`14:20.53`	`zoa`	`h3x: you mean with early media ?`
`14:20.54`	`Redfury`	`hi all`
`14:21.02`	`[TK]D-Fender`	`jimbalcomb : as opposed to inband?`
`14:21.12`	`Redfury`	`I have configured Asterisk using Database,`
`14:21.25`	`Cresl1n`	`mimmus: I don't understand the problem. You say it's ringing, and you're wondering why it's not reported as being answered...`
`14:21.34`	`Redfury`	`and the peers are also picked fromthe db`
`14:21.54`	`Redfury`	`but I am getting a Failure to Query the database warning`
`14:22.15`	`Mimmus`	`Cresl1n: (my english is very bad, sorry) I hear tone of call proceeding normally, remote IVR doesn't answer`
`14:22.18`	`Redfury`	`does any one have any idea as to what must be wrong..?`
`14:22.34`	`BoRiS`	`exten => 1,1,Answer() :-p`
`14:22.56`	`*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)`
`14:22.57`	`Cresl1n`	`mimmus: if you hear the tone of call proceeding normally, and remote IVR doesn't answer, why are you expecting it to be in an answered state?`
`14:22.59`	`jimbalcomb`	`[TK]D-Fender: oh no, i've heard already that inband is sad just wondering which of those two is better, rfc2833 or SIPinfo?`
`14:23.06`	`*** join/#asterisk Defraz (n=t0tal@72.24.26.215)`
`14:23.35`	`Mimmus`	`Cresl1n: because Asterisk doen't detect answer, if I use an analog phone, IVR ansers after first ring`
`14:23.46`	`Reverend`	`mcquaid what kind of info do you want to see?`
`14:24.00`	`sivana`	`Mimmus: how are you connected to PSTN?`
`14:24.06`	`mcquaid`	`mut, http://pastebin.ca/36591`
`14:24.11`	`Mimmus`	`sivana: E1 PRI in Italy`
`14:24.15`	`Cresl1n`	`Mimmus: that basically means you want to use your shiney new PRI as an analog line`
`14:24.31`	`Cresl1n`	`Mimmus: kind of defeats the point of half of what people use PRIs for`
`14:24.31`	`mcquaid`	`Reverend, mut wanted to see asterisk debug when I receive a call from my voip provider`
`14:24.56`	`Mimmus`	`Cresl1n: and what is the correct behaviour?`
`14:24.59`	`Cresl1n`	`Mimmus: if so, that's simple, just do what BorIS said and do an Answer() on your line`
`14:25.11`	`sivana`	`Cresl1n: he's saying that when he uses the PRI, it doesn't detect the remote answer, but when he uses an analog on the same number, it answers`
`14:25.12`	`Cresl1n`	`Mimmus: the correct behavior is how it is behaving`
`14:25.22`	`Cresl1n`	`sivana: that's wrong`
`14:25.31`	`[TK]D-Fender`	`jimbalcomb : Equal. there are multiple forms available because not every device supports either one. Sipura devices don't seem to support rfc2833. Since they use AVT & INFO, I chose INFO for my * side. And things just work. I don't believe ther is a "better" aspect of it`
`14:25.36`	`Cresl1n`	`sivana: that maybe what he's saying, but the problem is wrong`
`14:25.46`	`tzanger`	`h3x: how can you navigate their IVR without them answering? You could receive their audio but you shouldn't be able to send anything (even keypad IEs) I thought`
`14:25.54`	`konfuzed`	`Cresl1n: Mimmus is bummed that he can only get to the IVR when using the analog phone. When using other phones the IVR never picks up`
`14:26.03`	`Mimmus`	`Cresl1n: but it doesn't work! I don't understand :(`
`14:26.17`	`sivana`	`Mimmus: re-explain the problem`
`14:26.35`	`Cresl1n`	`Mimmus: you're going to have to start over`
`14:26.43`	`mut`	`mcquaid: you sure thats not your voicemail system hanging up the call?`
`14:26.47`	`Mimmus`	`sivana: my english is really a problem... sorry... konfuzed explained better`
`14:26.55`	`konfuzed`	`Mimmus: also confirm if what I said is right or wrong or partly correct`
`14:27.20`	`sivana`	`but I'm confused with phones then... * isn't a phone`
`14:27.49`	`Mimmus`	`phones connected to *`
`14:27.56`	`Cresl1n`	`Mimmus: so tell me more about what konfuzed said`
`14:27.57`	`mcquaid`	`hmm, don't see how voicemail would be interferring`
`14:28.21`	`Redfury`	`Hey Anybody has answer to my problem in configuring asterisk with the database...`
`14:28.22`	`jimbalcomb`	`[TK]D-Fender: ok, that is exactly my wondering. thanks.`
`14:28.24`	`Mimmus`	`both directly connected VoIP phones and analog phones connected to a legacy PBX downstream`
`14:28.27`	`mcquaid`	`mut, as i shown in my post, i took my local sip phone out of the equation and just tried to have asterisk play monkeys`
`14:28.33`	`mcquaid`	`it says it is but i hear nothing`
`14:29.08`	`mut`	`the phone isn't behind a nat is it?`
`14:29.28`	`Mimmus`	`Cresl1n: I'm calling a toll-free number by my shiny VoIP phone connected to * and it never ansers`
`14:29.29`	`mcquaid`	`yes the phone and the asterisk box are both behind a nat`
`14:29.44`	`mcquaid`	`the sip phone that is`
`14:30.01`	`mut`	`nothing inbetween tho?`
`14:30.08`	`mcquaid`	`but as i mentioned, if i set up the sip phone to directly connect to my voip provider, i can make and receive calls`
`14:30.10`	`mcquaid`	`no`
`14:30.23`	`*** join/#asterisk tony__ (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com)`
`14:30.23`	`Cresl1n`	`so you have a call from (VoIP phone) -> (Asterisk) -> (PRI-to-PSTN)?`
`14:30.24`	`Mimmus`	`Cresl1n: if I use a plain, old analog phone, remote IVR answers after 1 ring`
`14:30.34`	`mcquaid`	`and i don't need to enable stun or nat for them to work, just set up the outbound proxy`
`14:30.38`	`Mimmus`	`Cresl1n: exactly`
`14:31.10`	`sivana`	`Mimmus: you don't get something like -- Zap/21-1 answered SIP/VOC0081-2-2a57 in your * CLI?`
`14:31.13`	`Cresl1n`	`Mimmus: and with (analog phone) -> (Asterisk) -> (PRI-to-PSTN) it works?`
`14:31.51`	`Mimmus`	`Cresl1n: no, I need to use a phone connected to a completely different line (no Asterisk in the path)`
`14:32.10`	`Cresl1n`	`Mimmus: Ah.... that's interesting`
`14:32.16`	`*** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net)`
`14:32.17`	`mcquaid`	`mut, would the debug of a working call (i.e. when i call my landline) help?`
`14:32.28`	`Mimmus`	`sivana: no, I'm getting " -- Zap/13-1 is proceeding passing it to SIP/232-6699"`
`14:32.37`	`Mimmus`	`sivana: and "-- Zap/13-1 is making progress passing it to SIP/232-6699"`
`14:32.44`	`mut`	`it's more than likely some kinda nat problem i'de imagine`
`14:32.55`	`mut`	`couldn't tell ya for sure tho`
`14:33.01`	`Cresl1n`	`Mimmus: this maybe unrelated, but what version of asterisk/libpri are you running?`
`14:33.17`	`Mimmus`	`Cresl1n: Asterisk 1.2.1, now I'm downloading latest CVS`
`14:33.29`	`mcquaid`	`hmm, i'm sure it is, but with outbound proxy in the sip clients on their own, incoming/outgoing work`
`14:33.48`	`mcquaid`	`without nat or stun, so i was hoping if they can do it, asterisk should be able to as well`
`14:34.07`	`mcquaid`	`tried to find documentation on outboundproxy and outboundproxyport but it's thin`
`14:34.28`	`mcquaid`	`only found info on most features being promoted to chan_sip from chan_sip2 last year`
`14:34.56`	`mcquaid`	`i also wondered if this would be a situation where siproxd would help`
`14:35.00`	`*** join/#asterisk skambar (n=keiner@minasmorgul.stuwo-steinweg.de)`
`14:35.30`	`sivana`	`Mimmus: does the asterisk and libpri version the same, right now?`
`14:36.18`	`*** part/#asterisk cfh (n=luca@82.193.23.6)`
`14:36.29`	`Mimmus`	`sivana: until now, I'm using plain Asterisk 1.2.1`
`14:38.52`	`mcquaid`	`mut, i emailed olaf as he worked on outboundproxy, hoping he'd might want to get outbound proxy working as well as it does in sip clients on their own`
`14:39.25`	`mcquaid`	`but haven't heard from him yet, i tried the asterisk-users forum as well`
`14:39.29`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`14:39.29`	`*** mode/#asterisk [+o anthm] by ChanServ`
`14:39.38`	`mcquaid`	`maybe i shoudl send this to the devel list...`
`14:40.15`	`konfuzed`	`ok so mimmus' analog phone is the out side line which works fine calling into his 1800-DID number. But when picking up the VoIP Phone on his LAN, dialing the 1800-DID just keeps ringing. Mimmus, if you just pick up your voip phone and punch in only an extension for another voip phone (plugged in or not plugged in) or dial 0, then does the IVR pickup`
`14:41.14`	`*** join/#asterisk abatista (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)`
`14:41.28`	`Mimmus`	`konfuzed: no no, to call this toll-free IVR I need to bypass Asterisk and use an old phone with a different line`
`14:41.35`	`*** join/#asterisk Katty (n=angela@64.82.232.54)`
`14:41.36`	`*** join/#asterisk razu_ (n=razu@213-35-170-76-dsl.trt.estpak.ee)`
`14:41.39`	`*** part/#asterisk Katty (n=angela@64.82.232.54)`
`14:41.41`	`konfuzed`	`Mimmus: right`
`14:41.43`	`konfuzed`	`so`
`14:41.48`	`*** join/#asterisk Katty (n=angela@64.82.232.54)`
`14:41.52`	`Mimmus`	`konfuzed: problem is in Asterisk not detecting remote answer`
`14:41.58`	`Cresl1n`	`Mimmus: have you tried taking out the 'r' flag in your dial, and see if you hear anything?`
`14:42.06`	`Katty`	`hi lads.`
`14:42.38`	`konfuzed`	`Mimmus: with the voip phone on your LAN can you get the IVR to pickup by calling an extension?`
`14:42.46`	`*** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net)`
`14:43.19`	`Mimmus`	`Cresl1n: no, I will try in a moment`
`14:43.31`	`ariel_`	`hello everyone`
`14:43.35`	`Katty`	`hewwo ariel_`
`14:43.38`	`tzanger`	`hello`
`14:43.47`	`Mimmus`	`konfuzed: IVR is a PSTN number!`
`14:43.59`	`ariel_`	`Katty, hope your day will be great`
`14:44.05`	`Katty`	`ariel_: thanks, yours too :>`
`14:45.13`	`devoider`	`endlich darf auch horst ins netz .. :o`
`14:45.27`	`devoider`	`oh wrogn # ;)`
`14:45.30`	`zoa`	`aaaarghl, im goin crazy here`
`14:45.32`	`Mimmus`	`Cresl1n: I already have 'r', I'm using 'TrwW'`
`14:45.33`	`devoider`	`err wrong`
`14:45.46`	`Cresl1n`	`Mimmus: take out the r`
`14:46.51`	`Mimmus`	`Cresl1n: ok, immediately`
`14:47.11`	`konfuzed`	`Mimmus: [09:16:26] <Mimmus> Cresl1n: in fact, it is a toll-free number of my telco. And is there no workaround? - where did this toll free number come from? is that your DID setup on your asterisk box or what ??`
`14:47.32`	`Mimmus`	`SOLVED!!!!!!!!!!!!!`
`14:47.35`	`*** join/#asterisk jyukes (n=jameshot@pool-138-89-229-250.atc.east.verizon.net)`
`14:47.42`	`konfuzed`	`the removing r it was then`
`14:48.03`	`konfuzed`	`Mimmus: still curious though, whats up with the toll free number`
`14:48.20`	`Mimmus`	`Can I offer a pizza+beer to Cresl1n?`
`14:48.42`	`*** join/#asterisk santoshr (i=1063@203.199.110.93)`
`14:49.10`	`Cresl1n`	`Mimmus: heh, I can never turn down free food :-)`
`14:49.14`	`Katty`	`beer :<`
`14:49.18`	`Katty`	`less beer, more hugs.`
`14:49.21`	`Katty`	`that's my moto.`
`14:49.31`	`jimbalcomb`	`thats gross`
`14:49.31`	`Katty`	`or possibly motto...never can remember.`
`14:49.33`	`Cresl1n`	`Katty: mine too :-)`
`14:49.39`	`Mimmus`	`Cresl1n: but it would be a real italian pizza`
`14:49.52`	`santoshr`	`i want to test dialing a remote sip server.. i found a list of public sip servers . how can one make a call to that`
`14:50.01`	`konfuzed`	`Cresl1n: I cen get you greyhound bus tickets to go pick up your pizza`
`14:50.11`	`konfuzed`	`;^)`
`14:50.17`	`jimbalcomb`	`same day air shipping via UPS global`
`14:50.43`	`jimbalcomb`	`it'd be the best $300 pizza you ever had`
`14:50.53`	`konfuzed`	`Mimmus: still curious though, whats up with the toll free number`
`14:51.31`	`Mimmus`	`konfuzed: what's the meaning of "whats up"?`
`14:51.47`	`Katty`	`Mimmus: it's a basic greeting`
`14:51.49`	`warthawg`	`que tal`
`14:51.51`	`santoshr`	`<< sip:www.foo.com >> wwere a public sip server which says it does not require a registration.. how should i send a call t here`
`14:51.59`	`warthawg`	`hey, vato, que paso`
`14:52.00`	`Mimmus`	`jimbalcomb: if I'm able to call UPS toll-free number now...`
`14:52.02`	`Katty`	`Mimmus: the lazy How Are You, routine.`
`14:52.12`	`konfuzed`	`and a direct inquiry of what is happening with`
`14:52.26`	`Katty`	`personally i find it annoying`
`14:52.30`	`Cresl1n`	`Mimmus: mmm.... I've never had italian pizza`
`14:52.39`	`Cresl1n`	`what's the difference?`
`14:52.47`	`BeHappy_`	`Cresl1n, dont get it in tuscany, if you want an advice :)`
`14:52.58`	`Mimmus`	`konfuzed: clearly toll-free doesnt' answer if you supply a ringtone ('r')`
`14:52.58`	`Cresl1n`	`konfuzed: heh, you're funny`
`14:52.59`	`konfuzed`	`what is up with the toll-free number you mentioned earlier. is it yours or in use some how ? Why was it mentioned`
`14:53.18`	`*** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com)`
`14:53.24`	`santoshr`	`guys.. can some one give me some ideas please.`
`14:53.24`	`Mimmus`	`Cresl1n: pizza was born in Italy (Naples)!`
`14:53.47`	`Mimmus`	`Cresl1n: in USA you eat a surrogate!`
`14:53.51`	`sivana`	`Cresl1n: was the r causing * to ignore the other end?`
`14:53.57`	`konfuzed`	`Mimmus: whos toll free number is it? yours or somebody elses?`
`14:54.08`	`konfuzed`	`is it a did on yout asterisk box`
`14:54.12`	`Mimmus`	`konfuzed: somebody else, my telco`
`14:54.14`	`Cresl1n`	`sivana: basically`
`14:54.15`	`*** join/#asterisk slak- (i=slak@rewted.biz)`
`14:54.16`	`konfuzed`	`s/did/DID/`
`14:54.34`	`slak-`	`hi, how can i tell which codec my sip connection is using`
`14:54.59`	`Cresl1n`	`sivana: The other end's IVR was starting before it sent the CONNECT, and with the r flag, asterisk sends locally generated ringback until the CONNECT message is received`
`14:55.00`	`slak-`	`im having a conference here using MeetMe and would like to make sure that i have enough bandwidth to support 5 partries`
`14:55.13`	`Mimmus`	`sivana: yes`
`14:55.38`	`Cresl1n`	`sivana: ere go... it overrode the audio that the other end was sending`
`14:55.42`	`konfuzed`	`ok good note on the machincations of the r flag`
`14:55.46`	`santoshr`	`how to dial out a public sip server. sip:foo.com`
`14:56.13`	`sivana`	`I see`
`14:56.49`	`konfuzed`	`Mimmus: do you have Local phone numbers as DID for incoming or just PSTn as in incoming phone number?`
`14:56.51`	`Mimmus`	`Cresl1n: very sad... 2 weeks for this...`
`14:57.07`	`Mimmus`	`konfuzed: why this question?`
`14:57.31`	`konfuzed`	`to understand your layout`
`14:57.54`	`Mimmus`	`Cresl1n: just because I lazily cut&paste dialing options`
`14:58.02`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`14:58.11`	`*** part/#asterisk zapotecz (n=surfer@217.201.198.236)`
`14:58.26`	`Cresl1n`	`Mimmus: heh, you know what they say... you spend 90% of the time on 10% of the problems`
`14:58.34`	`konfuzed`	`I have trouble with being left an incomplete picture because it is too KonFuZing`
`14:58.40`	`konfuzed`	`;^)`
`14:59.00`	`Mimmus`	`Cresl1n: beh, now it's time to go. Thank you again to this channel and especially to you, Cresl1n`
`14:59.12`	`Cresl1n`	`Mimmus: no prob, good luck! :-)`
`14:59.24`	`Mimmus`	`we are aplanning to replace two legacy Alcatel PBX (for 200 users in two sites)`
`14:59.26`	`konfuzed`	`the same problem I have with how answered quetions can be like unsolved mysteries even when no longer such a big deal`
`14:59.30`	`Mimmus`	`and I have much to do`
`14:59.31`	`konfuzed`	`kinda like X-Files`
`14:59.37`	`warthawg`	`can anyone tell me how to get message waiting indicator working on grandstream phone?`
`14:59.45`	`slak-`	`how does g726 compare to ulaw?`
`14:59.50`	`slak-`	`whats the bandwidth difference`
`15:00.01`	`Cresl1n`	`slak-: that's totally google'able`
`15:00.11`	`{zombie}`	`warthawg: there's no trick, just make sure you have the appropriate mailbox= statement in your sip.conf`
`15:00.11`	`slak-`	`okay well i guess its totally askable aswell`
`15:00.12`	`trixter`	`asteriskgurus.org has a bandwidth calculator`
`15:00.41`	`{zombie}`	`and make sure you are either putting your mailboxes under the [default] context in voicemail.conf, or specifying the context in your mailbox=`
`15:00.47`	`brad_mssw`	`slak-: http://www.voip-info.org/wiki/view/Bandwidth+consumption`
`15:00.57`	`trixter`	`as far as bandwidth consumed there are variables. sample size, trunking or no, ATM framing or no, pppoe?`
`15:01.11`	`slak-`	`t1`
`15:01.27`	`warthawg`	`{zombie} ok, thanks`
`15:01.29`	`slak-`	`which codec is ulaw...g7xx?`
`15:01.30`	`konfuzed`	`Mimmus: So, do you have any DID's configured`
`15:01.49`	`brad_mssw`	`slak-: g711`
`15:02.00`	`slak-`	`ty`
`15:02.04`	`Mimmus`	`konfuzed: yes`
`15:02.51`	`konfuzed`	`is that just a toll-free DID or local numbers too`
`15:03.34`	`Mimmus`	`konfuzed: noooooo! It's a public number, not mine!`
`15:03.37`	`*** join/#asterisk bweschke-away (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)`
`15:04.29`	`konfuzed`	`i presume you mean the Toll-free vs local numbers and so that would complete the layout picture quite nicely. at least for me anyway`
`15:05.00`	`Mimmus`	`konfuzed: ok, see you tomorrow, thanks`
`15:05.23`	`konfuzed`	`always good to have a complete picture if possibly eh`
`15:05.25`	`konfuzed`	`;^)`
`15:05.26`	`Ahrimanes`	`anyone successfully get leds on snom phones to turn on and off from asterisk?`
`15:05.57`	`malverian[work]`	`Ahrimanes, Yes.`
`15:06.24`	`*** join/#asterisk klictel (n=klictel@207.107.208.137)`
`15:06.33`	`Ahrimanes`	`malverian[work]: hm have a dialplan example for that?`
`15:06.53`	`[TK]D-Fender`	`malverian[work] : how's that scheduler coming along`
`15:07.09`	`[TK]D-Fender`	`Ahrimanes : exten => 1000,hint,SIP/1000`
`15:07.27`	`[TK]D-Fender`	`Ahrimanes : exten => 1000,1,Dial(SIP/1000,20)`
`15:08.11`	`Ahrimanes`	`[TK]D-Fender: well, i have an agi application that adds/removes a phone from a queue and i'd like it to toggle the led light on the button i press to launch the script..`
`15:08.47`	`*** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net)`
`15:08.55`	`Ahrimanes`	`[TK]D-Fender: so i set the button as a destination for 1000 right?`
`15:10.11`	`[TK]D-Fender`	`Umm, that you CAN'T do yet. SIP Presence only works for devicestate, not just anything.`
`15:10.37`	`Katty`	`..hams?`
`15:10.43`	`Katty`	`that does not parse.`
`15:10.44`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)`
`15:11.08`	`[TK]D-Fender`	`http://dictionary.reference.com/search?q=hams #6`
`15:11.25`	`jbalcomb`	`[TK]D-Fender: Wouldn't sending DTMF as SIP INFO rather than RTP (rfc2833) essential be more reliable due to TCP rather than TCP?`
`15:11.31`	`Katty`	`[TK]D-Fender: don't do that.`
`15:11.32`	`*** join/#asterisk kpettit (n=keith@69.15.174.114)`
`15:11.33`	`[TK]D-Fender`	`Typically used for over-presenting onesself`
`15:11.38`	`Katty`	`[TK]D-Fender: just be yourself.`
`15:11.44`	`Katty`	`[TK]D-Fender: like file.`
`15:11.46`	`Ahrimanes`	`[TK]D-Fender: hm not sure it qualifies as presence.. just toggling the led on a snom`
`15:11.59`	`file`	`Asterisk doesn't use TCP for SIP`
`15:12.06`	`[TK]D-Fender`	`jbalcomb : last I checked All of SIP & RTP were UDP....`
`15:12.15`	`jbalcomb`	`file ah`
`15:12.23`	`[TK]D-Fender`	`Ahrimanes : Not sure if there's a way to toggle them with direct header info....`
`15:12.30`	`tzanger`	`haha`
`15:12.48`	`jbalcomb`	`[TK]D-Fender: ah, hrmm.. how i can to that i dont know but i though it did. too many damn web pages with too many guessed at opinions..`
`15:12.54`	`Ahrimanes`	`[TK]D-Fender: well using devstate i have led in button 5 on my snom190 permanently on now.. but cant get it to turn off, hehe`
`15:13.31`	`jbalcomb`	`Ahrimanes perhaps poking it with a hot solder iron?`
`15:13.39`	`[TK]D-Fender`	`Ahrimanes : reboot the phone. Also keep in mind * wipes presences data every time you do "reload" in CLI`
`15:14.17`	`Ahrimanes`	`[TK]D-Fender: i pulled the power on the phone and did reload in cli and led is still on.. persistent bugger`
`15:14.28`	`Ahrimanes`	`jbalcomb: customer probably would not agree with that`
`15:14.37`	`jbalcomb`	`Ahrimanes: do you like that snom phone? if so, which modem and how much $$$?`
`15:14.55`	`jbalcomb`	`Ahrimanes: hrmm.. perhaps. just tell them its a built in incense burner`
`15:15.07`	`Ahrimanes`	`jbalcomb: i rather like it yes.. costs around $150 i guess.. only know the price in danish currency..`
`15:15.20`	`[TK]D-Fender`	`jbalcomb : Think Polycom ;)`
`15:15.58`	`jbalcomb`	`[TK]D-Fender haha.. yeah, we have several sipura, one polycom, and 100+ grandstreams`
`15:16.23`	`jbalcomb`	`[TK]D-Fender i don like the polycom so much yet`
`15:16.40`	`*** join/#asterisk diego_br (n=diego@200.208.241.178)`
`15:17.18`	`[TK]D-Fender`	`jbalcomb : Which model, and what aspects of it?`
`15:18.32`	`jbalcomb`	`[TK]D-Fender not sure on the model. its too quiet. i have heard good things about them though and we do only have one.`
`15:18.53`	`jbalcomb`	`[TK]D-Fender additionally its in the computer room so its not getting used much`
`15:19.26`	`jbalcomb`	`[TK]D-Fender do you like the polycoms? a particular model?`
`15:20.17`	`[TK]D-Fender`	`I'm running an all-Polycom setup (26 x IP600, 1 x IP601). Volumes are fine. Is the the default volume thats a problem or the max being too low?`
`15:20.50`	`[TK]D-Fender`	`How many line keys on yours? 6 little ones = IP60x, 3 big = IP50x, 2 small = IP30x`
`15:22.00`	`fugitivo`	`is any way to have callprogress with sip?`
`15:24.58`	`devoider`	`block?`
`15:25.19`	`devoider`	`dammit ... wrong # once again`
`15:25.25`	`Cresl1n`	`fugitivo: like inband progress?`
`15:25.33`	`devoider`	`ill check back beeing more awake .. maybe tommorow :)`
`15:26.02`	`fugitivo`	`Cresl1n: tone detection, answering machine, fax, busy, congestion, etc`
`15:26.17`	`Cresl1n`	`fugitivo: nope`
`15:26.19`	`*** join/#asterisk SplasPood (i=jwb@206.252.198.100)`
`15:26.23`	`fugitivo`	`no way to do that?`
`15:26.44`	`Cresl1n`	`fugitivo: have you ever used the zap callprogress code?`
`15:26.54`	`fugitivo`	`no, can't use it in my country`
`15:26.55`	`Cresl1n`	`fugitivo: it's not too great`
`15:27.06`	`Cresl1n`	`fugitivo: so in essence, the answer is no`
`15:27.12`	`*** join/#asterisk Defraz (n=t0tal@72.24.26.215)`
`15:27.19`	`fugitivo`	`using some kind of hack with backgrounddetect?`
`15:27.28`	`*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)`
`15:27.31`	`hackeron`	`can someone recommend a VoIP provider for backup that has unlimited data paths (or at least 8) on a pay as you go? (we are using teliax for main)`
`15:27.38`	`Cresl1n`	`fugitivo: with a LOT of hack`
`15:28.47`	`coppice`	`Its really bad when the product has reached 1.2 and can't do simple tone detection :-)`
`15:29.01`	`Cresl1n`	`coppice: hehe`
`15:29.13`	`Cresl1n`	`coppice: good morning ot you too :-)`
`15:29.14`	`fugitivo`	`right`
`15:30.00`	`Cresl1n`	`coppice: after all, it's ONLY simple tone detection, right? :-D`
`15:30.27`	`coppice`	`it is. its pathetic that the software can't go a reasonable job`
`15:30.45`	`Cresl1n`	`coppice: should only take five or ten minutes, just write a quick FFT algorithm, put a little glue in there, and wahlah!`
`15:30.49`	`Cresl1n`	`:-P`
`15:31.07`	`coppice`	`FFT is not the right starting point`
`15:32.33`	`konfuzed`	`coppice: perhaps you can get the code together by the end of the day ;^)`
`15:32.36`	`fugitivo`	`a have a document from a provider describing each tone, who wants to code it? :)`
`15:32.37`	`bkw_`	`Cresl1n, thats a problem... many things in asterisk are done half ass and never gone back over and fixed correctly`
`15:33.08`	`coppice`	`konfuzed: my code is GPL, so it cannot go into *`
`15:33.17`	`Cresl1n`	`bkw_: so we can either troll about it, or we can do something about it.....`
`15:33.44`	`bkw_`	`Cresl1n, I'm not trolling i'm just stating fact`
`15:33.44`	`*** join/#asterisk loick (n=loick@APuteaux-151-1-6-116.w82-120.abo.wanadoo.fr)`
`15:34.29`	`konfuzed`	`coppice: well if you wrote GPL then asterisk could borrow it as free inclusion with *`
`15:34.31`	`mog_work`	`mmmm trolls`
`15:34.34`	`coppice`	`Cres11n: what's the point of doing something, when updates just sit and rot?`
`15:34.38`	`bkw_`	`konfuzed, WRONG`
`15:34.59`	`Cresl1n`	`coppice: yeah, sorry about that. We're working on getting better with that`
`15:35.03`	`fugitivo`	`coppice: did you code unicall?`
`15:35.07`	`mog_work`	`brian why dont you help anthm and stop trolling....`
`15:35.13`	`tzanger`	`bkw_: well not exactly wrong... the GPL version of asterisk could use it without problem. but ABE and any of the commercial licensed versionsof * could not`
`15:35.18`	`*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)`
`15:35.22`	`bkw_`	`it can't be in CVS at all`
`15:35.35`	`mog_work`	`esp as we dont do cvs`
`15:35.36`	`mog_work`	`anymore`
`15:35.38`	`tzanger`	`bkw_: ? why not?`
`15:35.38`	`fugitivo`	`svn`
`15:35.39`	`coppice`	`Cres11n: its not just my stuff. many people complain their stuff ends up rotting. it seems to be the normal thing`
`15:35.45`	`*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj3a.dialup.mindspring.com)`
`15:35.56`	`tzanger`	`GPL does not restrict where or what it is stored with, ONLY distribution`
`15:35.58`	`*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfj3a.dialup.mindspring.com)`
`15:36.02`	`coppice`	`fugitivo: unicall is mine`
`15:36.09`	`mog_work`	`it was 1.2 , look at how many bugs we are going through this month`
`15:36.10`	`fugitivo`	`coppice: nice work :)`
`15:36.13`	`Cresl1n`	`coppice: yeah, we're realizing that and trying to work better to alliviate the problem`
`15:36.13`	`anthm`	`who's that clip clapping on my bridge!`
`15:36.40`	`konfuzed`	`mmmm had a little chat last week or so (forget where) that basically confirmed you can take GPL code and close it as long as the source is 'offered' for free.`
`15:36.45`	`mog_work`	`i was just telling brian to not troll and actually get some work done with you anthm`
`15:36.46`	`Cresl1n`	`coppice: it's a concern that we're becoming more and more aware of`
`15:36.53`	`bkw_`	`konfuzed, WRONG`
`15:36.55`	`konfuzed`	`hm`
`15:37.02`	`tzanger`	`konfuzed: well technically you're not closing it then, are you?`
`15:37.21`	`BoRiS`	`lol`
`15:37.25`	`mog_work`	`yeah there are sketch people out there like that router guy`
`15:37.27`	`konfuzed`	`well then I hope that conversation was here so that I dont have to go and correct some debian programmers or something like that`
`15:37.29`	`mog_work`	`sveasoft or whatever`
`15:37.36`	`bkw_`	`mog_work, anthm and I have done more for the asterisk code base than most people in the community`
`15:37.44`	`mog_work`	`no one is saying you havent`
`15:37.48`	`konfuzed`	`tzanger: well you can sell the binary`
`15:37.53`	`mog_work`	`but you guys arent now, and some of us have work to do`
`15:38.00`	`anthm`	`umm hi`
`15:38.02`	`tzanger`	`bkw_: I don't think anyone is denying you that. You and anthm are very, very good at this stuff`
`15:38.04`	`anthm`	`i have patches in there still`
`15:38.06`	`fugitivo`	`here we go again`
`15:38.07`	`konfuzed`	`as long as "Offering" the code as opposed to including the code`
`15:38.15`	`coppice`	`this is truly amazing. dell normally rip off asians, but their new 30" LCD seems to be cheaper here than in the US :-\`
`15:38.20`	`mog_work`	`i know anthm, i guess my comment was more directed at bkw_`
`15:38.26`	`tzanger`	`konfuzed: yes of course you can sell GPL binaries, but you must make the source available for free to anyone you distribute the binaries to. that's the entire point of the GPL`
`15:38.34`	`anthm`	`well to bring the conversation full circile`
`15:38.46`	`anthm`	`i was waiting for them to close to ever add any more`
`15:38.51`	`anthm`	`and it's been 8 months =D`
`15:38.56`	`Cresl1n`	`O. M. G. here we go again`
`15:39.01`	`anthm`	`s'all i'm sayin'`
`15:39.04`	`fugitivo`	`Cresl1n: :)`
`15:39.09`	`konfuzed`	`so why cant asterisk integrate some GPL pieces and make the code for those mods available`
`15:39.23`	`mog_work`	`hey i got a crazy idea, instead of complaining about old bugs`
`15:39.24`	`bkw_`	`konfuzed, because the code base can't be tainted`
`15:39.26`	`mog_work`	`lets go fix em`
`15:39.34`	`mog_work`	`i mean 242`
`15:39.38`	`mog_work`	`we can work it down`
`15:39.40`	`BoRiS`	`'Ya'll jacked up and sheeeeeeeeet'`
`15:39.40`	`bkw_`	`if the asterisk codebase is tainted with pure GPL code then ABE and dual lic. wouldn't be possible.. along with g729`
`15:39.41`	`coppice`	`I love this policy about "feature requests" If they can eb ignored for a few weeks, they get deleted. great scheme, that one`
`15:39.43`	`tzanger`	`konfuzed: because ABE and the commercially-licensed copies of * cannot have that code in them, because the GPL parts "infect" the closed-source parts since they're linked`
`15:39.47`	`konfuzed`	`hm Id say that depends on perspective (more so from the programmers than mince of course)`
`15:39.49`	`fugitivo`	`mog_work: yeah! callprogress for sip!`
`15:40.05`	`*** join/#asterisk pa (n=Paolo@unaffiliated/pa)`
`15:40.48`	`Cresl1n`	`coppice: I don't know if the "feature request" mechanism on the bug tracker is the right place for a lot of that stuff`
`15:40.51`	`mog_work`	`i thought sip had callprogress?`
`15:40.52`	`zoa`	`coppice, they dont even sell those in belgium yet :(`
`15:41.21`	`coppice`	`zoa: they don't ship until the 16th here`
`15:41.22`	`fugitivo`	`mog_work: really? can i detect an answering machine, busy, congestion, etc?`
`15:41.31`	`zoa`	`its not on the sites`
`15:41.38`	`mog_work`	`busy and congestion work with sip`
`15:41.50`	`mog_work`	`and there is a patch for generic answering machine detection`
`15:42.02`	`Cresl1n`	`coppice: what's to motivate people to work on feature requests anyway... so if nobody want to work on it (i.e. not enough demand) they sit there and rot`
`15:42.18`	`bkw_`	`Cresl1n, just like patches`
`15:42.26`	`*** join/#asterisk rkioko (n=rkioko@196.200.26.42)`
`15:42.29`	`anthm`	`i'm not bitching or anything and in fact i'll even tell you one more time for the record since you guys just said you were trying to fix the problem. The issue lies with the whole idea where you burden the developer by making him guess how you guys want the code to be then sending it back for recoding after the fact instead of just spending 20 min to describe it the way you would like it to be ahead of time`
`15:42.29`	`bkw_`	`patch rot is the biggest killer of new features`
`15:42.34`	`tzanger`	`bkw_: do you not agree that Digium's gotten a LOT better with that in the last 3 months?`
`15:42.45`	`tzanger`	`nobody is denying that it was very bad in the past`
`15:42.46`	`Cresl1n`	`bkw_: hey man, we're trying to get better at that`
`15:42.52`	`coppice`	`tzanger: no. it has got worse`
`15:43.08`	`tzanger`	`however Digium's taken steps to improve that. If you can't at least admit that it's moving better (not perfect yet of course) then you're a lost cause.`
`15:43.12`	`tzanger`	`coppice: really?`
`15:43.16`	`anthm`	`also small changes should just be done by the guy committing it and not bother sending it back for minimal alterations`
`15:43.28`	`coppice`	`tzanger: they seem to be casting into stone the things that were just vaguely wrong before`
`15:43.47`	`zoa`	`i think there is a lack of interest from normal users`
`15:43.53`	`zoa`	`they are fast to send emails like make me this`
`15:44.03`	`zoa`	`and then you make it and nobody ever tests it`
`15:44.08`	`konfuzed`	`hhmmm theres always more than one way to do things. Perhaps an GPL project for the Tone Detection that end users can easily grab on their own seperately via ftp or cut and paste or something ;^)`
`15:44.26`	`konfuzed`	`it could have its own web page`
`15:44.36`	`bkw_`	`konfuzed, if you even think about offering up code without disclaiming it to digium you get yelled at and called all kinds of names.`
`15:44.51`	`bkw_`	`I have personally had first hand exp. with that.`
`15:44.57`	`mog_work`	`analog tone detection is never gonna be awesome, its really hard and needs real dev.`
`15:45.00`	`Cresl1n`	`or somebody writes something that doesn't really belong in the public repository (for whatever reason) and they think that just because it went up there it should go in`
`15:45.14`	`coppice`	`mog_work: rubbish`
`15:45.24`	`konfuzed`	`i would say go ahead and disclaim that GPL code to digium`
`15:45.26`	`bkw_`	`Asterisk does too much as it is... It can't do any one thing very well.`
`15:45.27`	`tzanger`	`coppice: ? casting into stone the things that were just vaguely wrong?`
`15:45.33`	`bkw_`	`konfuzed, NO`
`15:45.58`	`coppice`	`tzanger: instead of just doing things badly, they now seem to be firm policies`
`15:46.01`	`Cresl1n`	`bkw_: that's obviously incorrect logic`
`15:46.04`	`anthm`	`naturally you are going to have daftly written patches but take coppice for instance trying to give you guys t38 for goodness sake and it's being nitpicked to death...`
`15:46.08`	`konfuzed`	`bkw_: obviously im missing something`
`15:46.20`	`mog_work`	`hey brian, i mean you are angry at asterisk and us, why do you even come in here?`
`15:46.22`	`konfuzed`	`it comes from not being a programmer my self`
`15:46.34`	`sivana`	`I think we should just convert it all to win32 with wav`
`15:46.59`	`tzanger`	`coppice: I'm gonna convert you to win32 with wav`
`15:47.04`	`sivana`	`hehe`
`15:47.16`	`*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)`
`15:47.19`	`Cresl1n`	`anthm: we don't put hardly anything in without nitpicking, stuff that I put in even is nit-picked`
`15:47.25`	`*** join/#asterisk nguyep (n=chatzill@64.34.203.231)`
`15:47.26`	`Cresl1n`	`anthm: it's called peer review`
`15:47.32`	`bkw_`	`Cresl1n, but you guys totally nit-pick the wrong thing`
`15:47.34`	`anthm`	`certianly`
`15:47.35`	`coppice`	`tzanger: wav is perfectly good, as long as it isn't running on win32`
`15:47.44`	`nguyep`	`any1 use asterisk to connect to sunrocket?`
`15:47.56`	`tzanger`	`nguyep: not me`
`15:48.30`	`coppice`	`Cres11n: if you had a lovely pristine codebase people might think nitpicking was OK. As it is.......`
`15:48.47`	`Cresl1n`	`coppice: it has to get there somehow`
`15:48.59`	`bkw_`	`it should have been done right in the first place`
`15:49.01`	`anthm`	`points of view are easily skewed to help prove a point I suggest you go peer review the code already in there with the same scrutiny I bet you would reject just about every channel driver =D`
`15:49.04`	`Cresl1n`	`coppice: so we can either try to make it better, or we could be apathetic`
`15:49.19`	`tzanger`	`bkw_: should've and could've are irrellavent. I don't see openpbx as doing things the right way right out of the gate either`
`15:49.39`	`bkw_`	`tzanger, As you can tell I don't work on OpenPBX .. never really have.`
`15:49.45`	`mog_work`	`lol`
`15:50.03`	`Cresl1n`	`coppice: but we realize that there are some problems with how things are done right now, and we'd like to try to make them better`
`15:50.11`	`tzanger`	`bkw_: actually I didn't know that, you were one of the biggest drivers behind it if memory serves (It often does not though)`
`15:50.20`	`bkw_`	`yes but I didn't code on it :P`
`15:50.24`	`Cresl1n`	`coppice: so obviously, if you have suggestions for how to do so, then we would like to try to use them`
`15:50.33`	`mog_work`	`bkw seems to pop up quite a bit.....`
`15:50.35`	`bkw_`	`mog_work, cutting fat away isn't coding`
`15:51.06`	`*** join/#asterisk objRobMitch (n=chatzill@c-24-1-203-134.hsd1.tx.comcast.net)`
`15:51.08`	`bkw_`	`Asterisk has so much fat it needs to be put on a diet :P`
`15:51.20`	`konfuzed`	`perhaps some one can confirm for me which OpenSource license it is that * is available under`
`15:51.29`	`mog_work`	`big is beautiful ^_^`
`15:51.32`	`sivana`	`konfuzed: it's dual licensed`
`15:51.37`	`mog_work`	`asterisk is GPL`
`15:51.45`	`mog_work`	`and is available for other licensing from digium`
`15:51.48`	`Cresl1n`	`bkw_: so you say that on one side, then you talk about the time that it take to get new feature patches in on the other... hrm.. makes a LOT of sense`
`15:51.49`	`coppice`	`except when it isn't`
`15:51.53`	`fugitivo`	`gpl2 sucks`
`15:51.57`	`konfuzed`	`hold on`
`15:52.07`	`BoRiS`	`wait for gpl revision 3`
`15:52.11`	`BoRiS`	`coming up soon`
`15:52.16`	`fugitivo`	`it'll suck`
`15:52.16`	`*** part/#asterisk nguyep (n=chatzill@64.34.203.231)`
`15:52.17`	`Beirdo`	`meh, whatever`
`15:52.18`	*** join/#asterisk james` (n=james@85.234.139.77)
`15:52.50`	`konfuzed`	`if asterisk is available under GPL then whats wrong with someone else making a GPL tone detector or anythign else?`
`15:52.57`	`mog_work`	`you coukld`
`15:53.13`	`mog_work`	`but it wont be put into the main tree with out disclaiming`
`15:53.30`	`sivana`	`konfuzed: disclaiming means that Digium can use it as they see fit`
`15:53.38`	`bkw_`	`aka sell it in ABE`
`15:53.38`	`*** join/#asterisk rainkid (n=rainkid@gemini.os5.com)`
`15:53.39`	`anthm`	`the distro cannot contain anything they cannot completely sell to someone or it would invalidate the existing agreements`
`15:53.43`	`konfuzed`	`so does GPL doesnt it`
`15:53.46`	`*** join/#asterisk sachse (n=sachse@86.56.32.11)`
`15:53.53`	`coppice`	`konfuzed: * has no plug in type of scheme. anything external plays endless catchup`
`15:53.55`	`Cresl1n`	`bkw_: so what's wrong with that?`
`15:54.00`	`sachse`	`hi all`
`15:54.07`	`Cresl1n`	`bkw_: is it wrong for you to make money off of using Asterisk in your ITSP?`
`15:54.11`	`*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)`
`15:54.14`	`fugitivo`	`why not making a new snv or cvs tree with only 100% GPL code?`
`15:54.16`	james`	`My CID if i called another extention is correct, but if i call a context i have setup the CID always is "device" can any one shead some light on this?`
`15:54.25`	`rainkid`	`how do you get the caller id of an incoming call?`
`15:54.33`	`bkw_`	`Cresl1n, Nope not at all.`
`15:54.42`	`sachse`	`has some1 experience with asterisk and freenet in germany?`
`15:54.43`	`Cresl1n`	`bkw_: maybe you should be a little more fair with how you think`
`15:54.52`	`anthm`	`you should not bother argueing moot points I am happy to hear them say they know there is a problem so I am gonna see what becomes of that before I dig up any more code.`
`15:55.52`	`tzanger`	`rainkid: ${CALLERID(all)} or variants. "show function CALLERID"`
`15:56.56`	`anthm`	`btw where is my svn branch did i get one?`
`15:57.18`	`konfuzed`	`ok so makie it an external patch project that can then only be had manually by system operators . do at your own risk, unsupported and not in the main tree`
`15:57.34`	`Cresl1n`	`anthm: I didn't think we were under the impression that you wanted anything to do with asterisk anymore`
`15:57.46`	`*** join/#asterisk santiago (n=santiago@208.195.215.97)`
`15:57.56`	`anthm`	`what a waste that would be i can practicly recite the api calls`
`15:58.16`	`sivana`	`can I get one too, an svn branch, for my c# conversion`
`15:58.31`	`Uther_P`	`ack`
`15:58.36`	`sachse`	`problem: freenet asterisk 1.0.9 gentoo kernel 2.6.12-r6: outgoing calls works, incomming not. sipgate work in both directions. help?!`
`15:58.40`	`Cresl1n`	`and one for my J++ conversion too :-)`
`15:58.44`	`coppice`	`sivana: c# conversion? is this a religious thing? :-)`
`15:58.49`	`rainkid`	`thank s`
`15:58.51`	`Uther_P`	`hkaha`
`15:59.00`	`sivana`	`heh`
`15:59.05`	`BoRiS`	`c#=scary, j+=Very Nasty`
`15:59.09`	`*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)`
`16:00.01`	`anthm`	`I mean should I spend 2 years making asterisk do all this stuff then not use it or anything?`
`16:00.02`	`Uther_P`	`what was wrong with C++ that ms had to go and screw it up?`
`16:00.17`	`Uther_P`	`(not there needs to be a specific reason for ms to screw something up)`
`16:00.18`	`BoRiS`	`c++ is good`
`16:00.26`	`tzanger`	`acutally I have heard from many people that C# is quite nice`
`16:00.29`	`tzanger`	`I've never used it myself`
`16:00.40`	`MRH2`	`hi does the zaptel echo can only work for the external connected part of the call? SO you would still get echo on the asterisk side?`
`16:00.51`	`coppice`	`C++ is so nasty, it would be hard for MS to actually wreck it :-)`
`16:01.02`	`tzanger`	`I like plain old C`
`16:01.03`	`anthm`	`I like C+0`
`16:01.04`	`Cresl1n`	`MRH2: it depends on how long the the echo tail is`
`16:01.06`	`jbalcomb`	`MRH2: I believe so. Zapatel is just the Telco side`
`16:01.06`	`tzanger`	`picking up python`
`16:01.22`	`Cresl1n`	`MRH2: generally it should only need to do it for the call side`
`16:01.23`	`BoRiS`	`I prefer C-3 (Cubed) :-p`
`16:01.28`	`rkioko`	`hi guys`
`16:01.29`	`BoRiS`	`c+++`
`16:01.31`	`jbalcomb`	`QBASIC is best`
`16:01.35`	`BoRiS`	`LOL!`
`16:01.37`	`Uther_P`	`haha`
`16:01.41`	`Uther_P`	`qb45 rocks`
`16:01.43`	`lunk`	`gorllas.bas = best game on the planet`
`16:01.43`	`Uther_P`	`yay`
`16:01.44`	`*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)`
`16:01.45`	`BoRiS`	`I actually Bought the Qbasic 4.5 box`
`16:01.46`	`jbalcomb`	`damn right`
`16:01.47`	`Uther_P`	`haha`
`16:01.49`	`tzanger`	`lunk: hahaha`
`16:02.03`	`Uther_P`	`dont forget nibbles.bas`
`16:02.10`	`jbalcomb`	`haha.. i remember gorillas from my first programming class in highschool`
`16:02.11`	`BoRiS`	`and that wonder PC speaker music`
`16:02.14`	`coppice`	`MRH2: its the other way round.`
`16:02.16`	`lunk`	`i chuck exploding bananas at little worms`
`16:02.17`	`lunk`	`haha`
`16:02.22`	`BoRiS`	`wonderful`
`16:02.25`	`anthm`	`I heard locksmith 2.0 is out now you can copy floppy disks.`
`16:02.40`	`Uther_P`	`no way!`
`16:02.42`	`lunk`	`anthm: scotch tape has been around for years!`
`16:02.52`	`PoWeRKiLL`	`someone know about this error Jan 12 17:03:20 WARNING[9904]: chan_iax2.c:3732 iax2_trunk_queue: Maximum trunk data space exceeded to ?`
`16:03.10`	`coppice`	`lunk: many countries have had similar tape for just as long`
`16:03.16`	`tzanger`	`PoWeRKiLL: you filled up your iax2 trunk`
`16:03.19`	`tzanger`	`how many simultaneous calls?`
`16:03.34`	`MRH2`	`so echo can is for the person connected to the zaptel card only?`
`16:03.37`	`Uther_P`	`especially the scottish`
`16:03.45`	`anthm`	`the cheater answer is to turn up the constants of max trunk space`
`16:03.56`	`anthm`	`at the top of chan_iax.c`
`16:04.16`	`Uther_P`	`MRH2: echo can wouldn't serve any purpose to anything but whats connected to the zaptel`
`16:04.32`	`Uther_P`	`MRH2: voip isn't going to echo its packets :)`
`16:04.53`	`Uther_P`	`sidetone is a bitch`
`16:05.03`	`*** join/#asterisk oli1234 (n=olivier@vodsl-8055.vo.lu)`
`16:05.11`	`MRH2`	`yes it certainly is`
`16:05.33`	`PoWeRKiLL`	`tzanger how I did that ?`
`16:05.41`	`oli1234`	`hello, ihave certain problems to load sipusers form a mysql table... is there anybody who could help me?`
`16:05.41`	`coppice`	`MRH2: if you use a digital card, there will be no echo back to the caller. if you use an analogue card * cannot cancel the echo it causes, but it shouldn't really matter. the important thing is audio from an IP phone should not be reflected back to that phone. that is what will sound bad`
`16:05.43`	`tzanger`	`PoWeRKiLL: how many simultaneous calls were you trying to push through the trunk?`
`16:05.45`	`MRH2`	`wondering if it would be too long an echo to loop voip calls through zap? or even if it would be a good idea?`
`16:06.13`	`tzanger`	`MRH2: PRIs and CAS T1/E1s do not GENERATE echo. Hoewver you can still GET echo on them`
`16:06.16`	`PoWeRKiLL`	`tzanger : usually I have 10 calls`
`16:06.25`	`tzanger`	`what codec?`
`16:06.29`	`PoWeRKiLL`	`g729`
`16:06.34`	`tzanger`	`hmm`
`16:06.41`	`anthm`	`like i said turn up the constants`
`16:06.45`	`PoWeRKiLL`	`now when 1 calls arrive i got this error`
`16:06.46`	`Uther_P`	`voip loopback across 15 hops == perfect guitar reverb`
`16:06.47`	`anthm`	`they are very liberal`
`16:06.50`	`Uther_P`	`:D`
`16:06.56`	`tzanger`	`Uther_P: :-)`
`16:07.00`	`*** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it)`
`16:07.04`	`tzanger`	`anthm: you mean conservative?`
`16:07.10`	`*** join/#asterisk monachoi (n=bonvoyag@cpe-24-174-162-34.satx.res.rr.com)`
`16:07.19`	`anthm`	`i suppose i do`
`16:07.24`	`coppice`	`Uther_P: if you don't mind a 3.5kHz limited guitar :-)`
`16:07.43`	`tzanger`	`coppice: depends on the tune :-)`
`16:07.55`	`oli1234`	`I have certain problems to load sipusers form a mysql table... is there anybody who could help me? --> just contact me in private thx in advance`
`16:08.07`	`*** join/#asterisk FastJack (i=fastjack@p5091FE1E.dip.t-dialin.net)`
`16:08.14`	`Uther_P`	`coppice: sum it back to the original tone`
`16:08.24`	`Uther_P`	`heh, pull it out of phase too`
`16:08.43`	`*** part/#asterisk sachse (n=sachse@86.56.32.11)`
`16:09.17`	`PoWeRKiLL`	`tzanger any idea ?`
`16:09.39`	`coppice`	`isn't it sad that after nearly 20 years of ISDN, which was supposed to bring us wideband voice, we still use narrow band for almost all VoIP?`
`16:09.46`	`*** join/#asterisk zukzuk (n=c@p508709B7.dip0.t-ipconnect.de)`
`16:10.13`	`sivana`	`coppice: why is that`
`16:10.13`	`zukzuk`	`hey guys. does anybody, by chance, know a way to work around this problem: http://bugs.digium.com/view.php?id=5838&nbn=7 ?`
`16:10.19`	`zukzuk`	`i'm experiencing the exact same thing`
`16:10.35`	`tzanger`	`PoWeRKiLL: did you listen to anthm`
`16:10.44`	`Mimmus`	`I forgot nickname of a really valid guy who helped me a few minutes ago about a toll-free number not responding.... can anyone help me?`
`16:10.51`	`*** join/#asterisk Dorphalsig (n=Dorphals@200.71.58.39)`
`16:10.55`	`sivana`	`Mimmus: Cresl1n`
`16:11.02`	`Mimmus`	`ok, thanks`
`16:11.14`	`PoWeRKiLL`	`thanks anthm :)`
`16:11.22`	`Uther_P`	`a really 'valid' guy`
`16:11.24`	`Uther_P`	`haha`
`16:11.32`	`anthm`	`#define DEFAULT_TRUNKDATA 640 * 10`
`16:11.32`	`anthm`	`#define MAX_TRUNKDATA 640 * 200`
`16:11.40`	`coppice`	`sivana: because people tolerate any old crap, I guess. people like Skype, yet don't scream for wideband elsewhere`
`16:11.48`	`anthm`	`crank those and recompile`
`16:11.53`	`anthm`	`note it's a band aid`
`16:12.13`	`sivana`	`coppice: who does wideband right now?`
`16:12.17`	`*** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`16:12.41`	`PoWeRKiLL`	`anthm i have to change this #define TRUNK_CALL_START 0x4000 ?`
`16:12.50`	`anthm`	`i just pasted the 2`
`16:12.55`	`PoWeRKiLL`	`thanks`
`16:12.56`	`coppice`	`skype is the only major user. a number of UMTS users have wideband - if they call another suitable UMTS user`
`16:13.06`	`*** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`16:13.09`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway`
`16:13.20`	`coppice`	`anthm: 22kHz is a weird rate to use`
`16:13.29`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`16:14.00`	`anthm`	`one leg was a soundcard`
`16:14.10`	`anthm`	`i could have done 16 and 32 also`
`16:14.18`	`tzanger`	`All of my stuff is PSTN ended so wideband does absolutely dick-all for me`
`16:14.53`	`anthm`	`i'm concerned how to negotiate the wideband stuff seems like the rate in the sdp is only like a kinda sorta option`
`16:14.54`	`coppice`	`soundcards are a pain for VoIP. their sampling rates don't lock to anything - include tx not locking to rx`
`16:15.29`	`anthm`	`iax doesnt seem to have any rate element`
`16:15.38`	`anthm`	`so that will be fun`
`16:15.51`	`coppice`	`anthm: shouldn't be. if one end announces only 8kHz codecs, the other end certainly shouldn't choose something higher.`
`16:15.59`	`MRH2`	`thanks I am going to blame echo on the other party for the moment.`
`16:16.15`	`sivana`	`Session Description Protocol ?`
`16:16.16`	`anthm`	`well that act of announcing the 16k codec is what i am wondering about`
`16:16.23`	`sivana`	`Social Democratic Part?`
`16:16.26`	`coppice`	`IAX lacks a number of important things if it is to break into the big time.`
`16:16.42`	`coppice`	`Session Dementing Protocol`
`16:16.43`	`*** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net)`
`16:18.14`	`anthm`	`so at least it seems like spd has the concept of it but it's barely used so it's not likely it will be understood by much stuff`
`16:18.17`	`coppice`	`The CNG frame is useless. There is no proper allowance for sample rates. The text is not defined as being UTF-8. Various little odds and ends that nobody seems to care to sort out, but which will cripple it.`
`16:18.20`	`*** join/#asterisk dily_ (n=dily@host91-30.pool80105.interbusiness.it)`
`16:18.21`	`anthm`	`sdp i mean`
`16:19.19`	`Mimmus`	`sivana: now I have a different but seemingly related problem, can I try here or file a bug?`
`16:19.19`	`coppice`	`I think there should be no problems with SDP. Things that don't understand the rates will not support the related codecs. It should sort itself out`
`16:19.31`	`*** join/#asterisk www2 (n=www1985@cd4400448.cable.wanadoo.nl)`
`16:19.40`	`anthm`	`that's what i'm hoping for`
`16:20.08`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`16:20.41`	`*** join/#asterisk secure75 (n=mic@dslb-084-057-013-245.pools.arcor-ip.net)`
`16:20.42`	`jbalcomb`	`Should I be using uLaw or aLaw and what's the difference?`
`16:20.54`	`coppice`	`I think it should fall into place a lot better than T.38 :-) The spec for that fails to tie a whole mass of things down.`
`16:21.05`	`anthm`	`i guess iax could send an IE with khz in it but that should be fun getting it accepted`
`16:21.30`	`anthm`	`there is a bit of a difference between ulaw and alaw`
`16:22.05`	`coppice`	`jbalcomb: they won't talk to each other, but their quality is about the same`
`16:22.47`	`jbalcomb`	`coppice: is it correct that one is an american standard and the other is a european standard? if so, which is which?`
`16:23.19`	`coppice`	`ulaw = US, HK, Taiwan, Japan`
`16:23.20`	`coppice`	`Alaw = the rest of humanity`
`16:23.31`	`coppice`	`oh, i missed canada`
`16:23.35`	`tzanger`	`:-)`
`16:23.39`	`tzanger`	`don't worry, everyone does. :-)`
`16:23.42`	`Uther_P`	`its common`
`16:24.27`	`jbalcomb`	`coppice: excellent. i assume uLaw correlates to PCMU vocoder on my grandstream phones?`
`16:24.32`	`*** join/#asterisk jero (n=sflphone@savoirfairelinux.net)`
`16:24.37`	`Uther_P`	`yes`
`16:24.50`	`jero`	`hi`
`16:24.52`	`coppice`	`quite a few phones call them PCMU and PCMA`
`16:25.11`	`jbalcomb`	`excellent. i think all executive decisions regarding our codec setup have been made.`
`16:25.17`	`jbalcomb`	`thanks for the help yall`
`16:25.24`	`Cresl1n`	`anthm: IIRC, I think there's an IE for sample rate`
`16:25.38`	`anthm`	`oh that would be good`
`16:25.44`	`Cresl1n`	`anthm: (in IAX)`
`16:25.59`	`Cresl1n`	`I started working on wideband too, and that was one of the things mark mentioned to me`
`16:26.02`	`*** part/#asterisk www2 (n=www1985@cd4400448.cable.wanadoo.nl)`
`16:26.15`	`*** join/#asterisk Strom_C (n=strom@216-80-66-245.lem-bsr1.chi-lem.il.cable.rcn.com)`
`16:26.20`	`anthm`	`expressed in hz ?`
`16:26.33`	`Cresl1n`	`anthm: hrmm... not sure on that one`
`16:26.42`	`jbalcomb`	`ah snap, one more codec question. the grandstream codex FAQ is using kbps but the Cisco codec FAQ is using Kbps. is it kilobits or kilobytes that i should be thinking?`
`16:26.53`	`*** join/#asterisk rick222 (n=rick555@207.71.127.152)`
`16:28.12`	`*** join/#asterisk jijgeh (n=luken@static-66-182-95-76.bbsc.net)`
`16:28.15`	`anthm`	`yah i see unsigned short samprate; in iax2-parser.h`
`16:29.04`	`Cresl1n`	`anthm: cool, yeah, I thought there was one`
`16:29.06`	`Mimmus`	`Cresl1n: now I have a different but seemingly related problem, can I disturb you again or is there a better choice?`
`16:31.17`	`*** join/#asterisk uther (n=uther_p@66.180.120.82)`
`16:32.22`	`mog_work`	`woot 239!`
`16:32.33`	`wunderkin`	`oops, found 2 more bugs`
`16:32.36`	`Cresl1n`	`mog_work: yeppers`
`16:32.44`	`Cresl1n`	`Mimmus: ???`
`16:33.11`	`Mimmus`	`Cresl1n: yes, I'm`
`16:33.15`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`16:33.26`	`anthm`	`i wonder why it's unsigned short considering how stingy the rest of the lements are it could have been char since in khz the vals are all < 50`
`16:33.36`	`Mimmus`	`Cresl1n: different number and slightly different problem`
`16:33.45`	`anthm`	`at least it's there already`
`16:33.46`	`Mimmus`	`Cresl1n: but always an automatci responder`
`16:33.52`	`Cresl1n`	`anthm: so you could send 8600hz audio :-)`
`16:34.36`	`Cresl1n`	`anthm: here wait... so you can send 8633hz audio :-)`
`16:35.34`	`*** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com)`
`16:35.41`	`anthm`	`hmm looks like there are hard constants for various rates in a bitmask`
`16:35.48`	`*** join/#asterisk montag___ (n=montag@host187-252.pool8175.interbusiness.it)`
`16:35.53`	`anthm`	`8 11 16 22 44 48`
`16:36.03`	`montag___`	`it's there a way to define custom greetings for every voicemail mailbox ?`
`16:36.17`	`dos000`	`anyone has idea about a cheap 2 port sip 2 pstn gateway ?`
`16:36.19`	`aminorex`	`oh that england had but one head so that i might strike it off`
`16:36.50`	`wunderkin`	`montag___, yes i use the temp greeting for that`
`16:37.16`	`*** join/#asterisk DFS (n=bwarner@65.113.208.11)`
`16:37.31`	`DFS`	`Anyone available for a dialplan question?`
`16:37.35`	`montag___`	`wunderkin: ?`
`16:37.45`	`anthm`	`aha so you must have the same bitmap on both sides and you actually send the unsigned version of that paticular constant not the desired speed`
`16:37.47`	`Mimmus`	`montag___: if I remember well, hit '3' for special functions`
`16:38.02`	`Cresl1n`	`anthm: those are some funny sample rates`
`16:38.12`	`DFS`	`Anyone available for a dialplan question?`
`16:38.21`	`montag___`	`but i want to manage this file from filesystem, not from user dtmf interface`
`16:38.33`	`{zombie}`	`DFS: just ask the question, don't ask if you can ask`
`16:38.39`	`{zombie}`	`and please don't repeat yourself`
`16:38.48`	`coppice`	`anthm: they miss an important one for telephony - 32`
`16:38.51`	`Cresl1n`	`8 16 32 and 48 should probably be in there`
`16:38.59`	`Mimmus`	`montag___: they are under /var/spool/asterisk/voicemail`
`16:39.00`	`anthm`	`yah were is 32 ?`
`16:39.06`	`wunderkin`	`montag___, funny thing.. they are saved on the filesystem.. so if you do it from the menu and look in the directory you will see how it works`
`16:39.10`	`Cresl1n`	`but I don't know about the non even multiple choices`
`16:39.16`	`DFS`	`Zombie>>I am trying to set up a dialplan where I can call other voip users on another asterisk server in a diff. network`
`16:39.22`	`coppice`	`they miss 192 as well`
`16:39.36`	`Cresl1n`	`oh yeah, and 384 too :-P`
`16:39.45`	`coppice`	`don't be silly`
`16:40.18`	`montag___`	`wunderkin: ok, but you know the name for busy and unavailable files ?`
`16:40.20`	`dily_`	`anyone use bristuff?`
`16:40.31`	`coppice`	`192k, 24 bit 7.1 is bound to be de rigeur for audio conferencing this year`
`16:40.39`	`wunderkin`	`montag___, you can research that the same way`
`16:40.42`	`{zombie}`	`DFS: Dial(IAX2/user:pass@remoteserver/XXXX)`
`16:40.51`	`anthm`	`so when you convert it to bits`
`16:40.57`	`Cresl1n`	`coppice: ah, didn't realize that`
`16:41.09`	`montag___`	`ok thanks`
`16:41.23`	`Mimmus`	`montag___: unavail.wav and busy.wav (.WAV too)`
`16:41.32`	`*** join/#asterisk grandy (n=mmmurf@pcp05305753pcs.wanarb01.mi.comcast.net)`
`16:41.46`	`DFS`	`Zombie: Where do you place this...in extentions.conf or in IAX?`
`16:42.12`	`{zombie}`	`you put that in your extensions.conf`
`16:42.19`	`*** join/#asterisk ffs_04 (n=jbon@modemcable071.144-80-70.mc.videotron.ca)`
`16:42.37`	`anthm`	`nothing = 1`
`16:42.37`	`anthm`	`8k = 2`
`16:42.37`	`anthm`	`16k = 4`
`16:42.37`	`anthm`	`22k = 8`
`16:42.37`	`anthm`	`44k = 16`
`16:42.38`	`anthm`	`48k = 32`
`16:43.12`	`anthm`	`the seem strikingly similar to just sending the rate you want rounded to nearest khz`
`16:43.18`	`dos000`	`anyone know a 2 port gw (not ata) that will allow phone<->ata<->internet<->gw<->pstn ?`
`16:43.42`	`{zombie}`	`DFS: http://voip-info.org/wiki/view/Asterisk+Connect+2+servers would be good reading`
`16:44.09`	`coppice`	`i wonder what the difference between an ATA and a GW might be :-\`
`16:44.31`	`dos000`	`coppice, no fxo on the ata normally`
`16:44.42`	`rue_work`	`[TK]D-Fender you up?`
`16:44.49`	`Mimmus`	`Cresl1n: is it a good idea to open a bug on digium.com or is it better to wait here?`
`16:45.14`	`coppice`	`dos000: so you cook up your own terminology, and expect everyone to understand? :-\`
`16:45.28`	`dos000`	`coppice, even if you have an fxo interface you can only originate not terminate`
`16:45.34`	`DFS`	`zombie: Is there anything else I need to add? Just this statement with my info in extentions.conf?`
`16:45.55`	`{zombie}`	`DFS: I think you need to do a whole lot more reading...`
`16:46.07`	`anthm`	`what's the max val of unsigned short?`
`16:46.09`	`{zombie}`	`don't expect you can just throw random statements into your asterisk config files and make things wrk`
`16:47.10`	`jbalcomb`	`one more codec question. the grandstream codex FAQ is using kbps but the Cisco codec FAQ is using Kbps. is it kilobits or kilobytes that i should be thinking?`
`16:47.17`	`coppice`	`anthm: is this a trick question?`
`16:47.18`	`Beirdo`	`anthm: 2^16 - 1`
`16:47.35`	`Uther_P`	`jbalcomb: bits`
`16:47.45`	`Beirdo`	`65535`
`16:48.09`	`anthm`	`so too small to send hz`
`16:48.11`	`Cresl1n`	`Mimmus: it's always better to verify here or on the mailing list that it's actually a bug before you post one (like earlier with the 'r' flag in the Dial command)`
`16:48.21`	`anthm`	`but big enough to send rounded khz`
`16:48.23`	`jbalcomb`	`Uther_P: ah, most peculiar that Ciscos page would be incorrect. That certainly explains my confusion in the amount of traffic I'm seeing. Thank you.`
`16:48.34`	`Uther_P`	`usually kilobytes per second is denoted as k/s`
`16:48.41`	`coppice`	`the maximum value of a short int is when it saves 2 bytes of memory and squeezes the product into a much cheaper MCU or DSP :-`
`16:49.04`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`16:49.17`	`Mimmus`	`Cresl1n: this is correct. Now I try to explain (it isn't so simple)`
`16:49.27`	`*** join/#asterisk A-Tuin\|work (n=A-Tuin@nat.office.legend.net.uk)`
`16:49.33`	`rue_work`	`WARNING[19687] chan_iax2.c: Received mini frame before first full voice frame`
`16:49.36`	`Dorphalsig`	`btw, checking my zaptel modules... I noticed I have wct4xxp and tor2 up`
`16:49.36`	`anthm`	`i'm trying to figure out why the iax code has the rates in a bitmask is there a condition where you can have 2 rates at once ?`
`16:49.38`	`rue_work`	`anyone know where that commes from?`
`16:49.42`	`Mimmus`	`Cresl1n: I have (PSTN PRI) -- Asterisk --- Alcatel PBX --- analog phones`
`16:49.47`	`Uther_P`	`I would like anyone to find somewhere where bps is used to denote bytes per second`
`16:49.49`	`Dorphalsig`	`shouldnt I just have one of them?`
`16:49.50`	`coppice`	`i remember once spending over a week getting one instruction out of a DSP loop :-\`
`16:49.55`	`Mimmus`	`Cresl1n: ans some VoIP phones directly connected to Asterisk`
`16:50.15`	`anthm`	`since nothing uses it i was brainstorming other ways to send the rate in the constraints of the unsigned short it is declared as`
`16:50.26`	`*** join/#asterisk bhickey (n=chatzill@212.2.174.21)`
`16:50.31`	`coppice`	`anthm: of course. if a phone supports 8k and 16k you set two bits`
`16:50.43`	`Mimmus`	`Cresl1n: calling a number with automatci responder from analog phones doesn't work (NONSWER after two rings), from Voip phones works`
`16:51.04`	`Cresl1n`	`coppice: that sounds like it could cause problems`
`16:51.05`	`coppice`	`dunno why the IE can't have a list of shorts with all the possible rates, though`
`16:51.43`	`*** join/#asterisk }btorch{ (n=kvirc@208.63.19.172)`
`16:52.25`	`Cresl1n`	`coppice: well, I take that back. I guess it depends on how it 's used`
`16:52.40`	`}btorch{`	`when someone asks what voice standards * can support are they talking about the technology like SIP, GSM`
`16:53.07`	`coppice`	`Cres11n: its dumb trying to squeeze this down and loose flexibility. its only sent infrequently`
`16:54.04`	`Cresl1n`	`coppice: you mean with doing it as a bit mask?`
`16:54.16`	`Cresl1n`	`coppice: I think there's truth to that`
`16:54.27`	`anthm`	`you can send several ie with the same name correct?`
`16:54.49`	`anthm`	`you also have no way to tie which rate goes with which codec`
`16:54.58`	`coppice`	`why should you? an IE has variable length, so it can contain a list of things`
`16:55.02`	`Cresl1n`	`anthm: yeah, that's what I was concerned about`
`16:55.07`	`_Sam--`	`does any know if that cheap DLINK packet prioritizer recognizes IAX? http://www.voipsupply.com/product_info.php?manufacturers_id=45&products_id=1168`
`16:55.19`	`*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)`
`16:55.29`	`Cresl1n`	`anthm: so if you blindly advertised 8 and 16, but you don't support 16 on all 16 capable codecs, that's a problem`
`16:56.14`	`coppice`	`well, the whole rate concept is restrictive. take AMR WB for example. what is its rate?`
`16:57.51`	`Cresl1n`	`coppice: yeah, I think that the way rates are done are going to have to be rethought`
`16:57.56`	`*** join/#asterisk cfh (n=luca@82.193.23.6)`
`16:57.58`	`anthm`	`maybe send mutiple capability and mutiple rate per groups`
`16:58.15`	`coppice`	`i think specifying rates is a fundamentally bad idea. you should just specify the codecs. they defines their capabiliites`
`16:58.15`	`anthm`	`send 1 cap with ulaw alaw then send 1 rate with 8 and 16`
`16:58.18`	`anthm`	`then send the rest`
`16:58.23`	`anthm`	`and send a sigle 8`
`16:58.50`	`nextime`	`wath's the best ( as a stability and performance ) h323 channel, h323, oh323 or ooh323?`
`16:58.54`	`coppice`	`anthm: just what about AMR WB? or speex? they can dynamically change their rates, to deal with congestion`
`16:59.11`	`anthm`	`yes`
`16:59.20`	`cfh`	`is there a solution to config the fast numbers on asterisk server?`
`16:59.27`	`anthm`	`maybe we should hack all the other codecs to be able to negotiate thier own rate`
`16:59.28`	`watchy-`	`im sitting here naked, i just took a shower`
`16:59.32`	`watchy-`	`i feel so sexy`
`16:59.41`	`Beirdo`	`TMI`
`16:59.56`	`watchy-`	`u sure`
`17:00.08`	`Beirdo`	`absolutely`
`17:00.18`	`anthm`	`maybe sdp over IE =D`
`17:00.22`	`*** join/#asterisk zapotecz (n=surfer@217.201.198.236)`
`17:00.23`	`rue_work`	`as some of you may have noticed, I'm not verry farmiliar with asterisk. I'm currently trying to correct issues with a PSTN machine that kb1canobie assembled, who you may know of. I could really use some help going through teh errors on the system while I try to correct some issues that are making the people in the office really agitated (theyre damanding that the phone system be replaced completely) the first thing I want to resolv is the mos`
`17:00.33`	`coppice`	`a list of codecs, detailed enough to define the specific variants supported, should be a complete description`
`17:00.39`	`*** join/#asterisk buzzyd (n=buzzyd@82-45-247-173.cable.ubr01.enfi.blueyonder.co.uk)`
`17:01.22`	`dos000`	`coppice, still no idea about what i asked ?`
`17:01.44`	`rue_work`	`I also verrymch need to fix the voicemail, which keeps recording blank messages`
`17:01.58`	`dos000`	`rue_work, tow !`
`17:02.01`	`buzzyd`	`Hi All, anyone know how I can setup voicemail prompts instead of using the default american voiced ones when leaving voicemail messages`
`17:02.51`	`rue_work`	`buzzyd the files you want to re-record are in the directory /var/lib/asterisk/sounds/`
`17:02.52`	`Uther_P`	`buzzyd: eh?`
`17:02.59`	`buzzyd`	`I would like my users to be able to set their own message but I can't see where to configure it`
`17:03.16`	`Mimmus`	`buzzyd: language setting set also messages for voicemail but you need sounds file for your lang`
`17:03.17`	`anthm`	`enough of this dealing with issues that control the outcome of any success in the near future lets fix config issues`
`17:03.19`	`rue_work`	`buzzyd if I understood you right`
`17:03.29`	`buzzyd`	`rue_work that would change it for everyone though`
`17:03.32`	`*** join/#asterisk masonf (n=masonf@dungle.vineyard.net)`
`17:03.43`	`rue_work`	`buzzyd sorry I misunderstood`
`17:03.46`	`zapotecz`	`hi all`
`17:03.54`	`rue_work`	`and I'm not verry farmiliar iwth asterisk`
`17:03.58`	`Mimmus`	`buzzyd: and any user can record his/her message hitting '3'`
`17:03.58`	`buzzyd`	`I just want it so each person can have their own message instead of playing a standard one for all`
`17:03.58`	`zapotecz`	`no one has used the patch for the bearer?`
`17:04.02`	`zapotecz`	`http://bugs.digium.com/view.php?id=3547&nbn=26#bugnotes`
`17:04.14`	`masonf`	`what are some possible causes for the message: Unable to open Asterisk database?`
`17:04.18`	`rue_work`	`dispite that I need to fix a number of issues on a system`
`17:04.20`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
`17:04.33`	`rue_work`	`which I could really use someone talking me though`
`17:04.35`	`Mimmus`	`masonf: permissions?`
`17:05.05`	`masonf`	`Ill try running as root....`
`17:05.06`	`buzzyd`	`Mimmus, anyway of doing it without using that app`
`17:05.30`	`Mimmus`	`buzzyd: yes, record and save message under /var/spool/asterisk/voicemail/...`
`17:05.46`	`Mimmus`	`masonf: no no, usually Asterisk runs as asierisk user`
`17:05.54`	`Mimmus`	`masonf: asterisk user`
`17:05.55`	`masonf`	`yeah its permissions... now I need to find what files it wants.`
`17:06.00`	`rue_work`	`in http://pastebin.com/502582 that context, does anyone know what 'outage' should sound like?`
`17:06.11`	`buzzyd`	`mimmus: ok but how would I then link that to each account`
`17:06.16`	`DFS`	`zombie: Can you specifiy the host as an IP address when creating the REC server?`
`17:06.25`	`dily_`	`exit`
`17:06.27`	`dos000`	`buzzyd, check out theese guys http://actor.loquendo.com/actordemo/default.asp?language=en`
`17:06.36`	`Mimmus`	`buzzyd: there is a directory for any extension under /var/spool/asterisk/voicemail/default/...`
`17:07.39`	`*** join/#asterisk dasuberdavid (n=david@gateway.digium.com)`
`17:08.13`	`*** join/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it)`
`17:10.39`	`buzzyd`	`Thanks guys, I see it now ;)`
`17:11.04`	`DFS`	`zombie: when specifying the host on the REC server, can you use the IP address`
`17:11.13`	`Mimmus`	`buzzyd: you are welcome`
`17:11.41`	`Mimmus`	`hey people, even Mimmus is able to help someone!`
`17:12.29`	`DFS`	`mimmus:you familiar with configuring two asterisks to conduct calls between the two on two diff. networks?`
`17:12.36`	`*** join/#asterisk juice (n=juice@209.33.109.45)`
`17:12.54`	`Mimmus`	`DFS: using IAX?`
`17:13.08`	`DFS`	`yes...I've read the text on [REC_SERVER]`
`17:13.08`	`DFS`	`type=user`
`17:13.08`	`DFS`	`host=my.calling.server.ca`
`17:13.08`	`DFS`	`secret=mysecret`
`17:13.08`	`DFS`	`context=local`
`17:13.09`	`DFS`	`trunk=yes`
`17:13.44`	`DFS`	`this is where I'm confused....based on the site reading :http://voip-info.org/wiki/view/Asterisk+Connect+2+servers`
`17:14.05`	`DFS`	`where the host is my.calling.server...example... can u use an IP address instead?`
`17:14.17`	`Mimmus`	`DFS: yes, Ip is good`
`17:14.39`	`DFS`	`mimmus: thanks...wasn't for sure if it would still work...`
`17:15.15`	`Mimmus`	`I'm not sure what context stands for`
`17:15.44`	`*** join/#asterisk roulduke_ (i=yz6mgq5v@p508D0F3D.dip0.t-ipconnect.de)`
`17:17.17`	`mocker`	`I'm having a problem w/ Asterisk receiving faxes. The tif files appear to be all crunched up into about 1 inch instead of looking like a normal fax page.`
`17:17.20`	`mocker`	`Is that normal?`
`17:17.24`	`DFS`	`mimmus: do you create a new secret for IAX or do you use the current secret for registering devices?`
`17:18.55`	`*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)`
`17:18.57`	`Mimmus`	`DFS: secret is a 'password' between two peers`
`17:20.00`	`DFS`	`mimmus: correct..this password I have is different for each asterisk...which do I use..or do I create a new one`
`17:21.59`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`17:22.21`	`masonf`	`any ideas what files would be giving be permissions issues Ive already checked /var/log /etc/asterisk /var/spool and /var/run`
`17:23.25`	`fulgas`	`strace asterisk`
`17:23.47`	`fulgas`	`and check for the permissions problem`
`17:24.10`	`*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)`
`17:24.14`	`*** part/#asterisk cfh (n=luca@82.193.23.6)`
`17:24.36`	`generalhan`	`whats up everyone ? !`
`17:24.41`	`DFS`	`mimmus: what is the [mycontext] ? What do you specify for that?`
`17:25.39`	`*** join/#asterisk A-jay (n=quirc@62.217.245.194)`
`17:26.24`	`generalhan`	`can anyone help me out with a compling problem im having with zaptel-1.2.1 ???`
`17:27.29`	`Mimmus`	`DFS: I used the same on both servers`
`17:27.49`	`*** join/#asterisk rkioko (n=rkioko@196.200.26.42)`
`17:28.16`	`Mimmus`	`DFS: if you like, I can post my con on pastebin.com`
`17:29.25`	`DFS`	`mimmus: that would be great...this project is confusing`
`17:29.53`	`Mimmus`	`DFS: just a moment...`
`17:32.30`	`[TK]D-Fender`	`rue_work : Here`
`17:32.55`	`*** join/#asterisk denon (i=denon@synapse.subneural.net)`
`17:32.55`	`*** mode/#asterisk [+o denon] by ChanServ`
`17:33.21`	`[TK]D-Fender`	`rue_work : that last pastebin of yours is very wrong. When you're back I'll help you fix it up`
`17:33.24`	`watchy`	`i got voicemail setup`
`17:33.27`	`watchy`	`how do i access it?`
`17:34.16`	`[TK]D-Fender`	`watchy : set up an extension that your phones can dial like "exten => 1234,1,VoicemailMain"`
`17:34.26`	`watchy`	`oh`
`17:34.33`	`DFS`	`mimmus: who will you post as`
`17:34.35`	`Mimmus`	`DFS: here http://pastebin.com/502626`
`17:34.37`	`watchy`	`so if i dial it from the actuall phone`
`17:34.41`	`watchy`	`it'll let me hear vM?`
`17:34.41`	`DFS`	`mimmus: thanks`
`17:35.09`	`[TK]D-Fender`	`watchy : that will bring you to the VM "main" where it'll ask you which VM box & password and then let you listen`
`17:35.19`	`watchy`	`ah`
`17:35.21`	`Mimmus`	`DFS: a small error, look here: http://pastebin.com/502627`
`17:35.26`	`watchy`	`and using variables i can auto goto the box?`
`17:36.16`	`[TK]D-Fender`	`watchy : like this - "exten => *98,1,VoicemailMain(${CALLERID(num)}@default)`
`17:36.22`	`watchy`	`sweet`
`17:36.33`	`watchy`	`so whats 98? literally 98?`
`17:37.00`	`[TK]D-Fender`	`watchy : that will assume your phones callerid is the same as its VM box #. You can script it up any which way you want like say "if its 555 then use box 222" or whatever`
`17:37.20`	`[TK]D-Fender`	`watchy : exactly like *98 (north american standard telco VM style)`
`17:37.20`	`watchy`	`yea`
`17:37.22`	`*** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com)`
`17:37.26`	`watchy`	`thats badass`
`17:37.34`	`DFS`	`mimmus: looking at the latest post...what factor do you decide which will be the REC server?`
`17:37.37`	`[TK]D-Fender`	`watchy, here, hold on, a gift for you...`
`17:37.41`	`watchy`	`thanks`
`17:37.43`	`Mimmus`	`DFS: REC?`
`17:37.48`	`DFS`	`mimmus: receive`
`17:38.19`	`PMantis`	`Howdy! Are there any Asterisk supported phones that act like an operator phone (can see which extens are in use, etc)`
`17:38.31`	`Mimmus`	`DFS: ah, it's a bidirectional trunnk (both peers)`
`17:38.52`	`DFS`	`mimmus: so you have to place this conf in both iax.conf in both servers?`
`17:38.59`	`[TK]D-Fender`	`watchy : here's a sample "features" context to add to your setup and include in your phone's main one. http://pastebin.com/502635`
`17:39.24`	`[TK]D-Fender`	`PMantis : SNOM, Polycom using SIP, CISCO's with SCCP.`
`17:39.39`	`trixter`	`I think etel has some issues scheduling.. they give phil zimmerman 15 minutes to talk about voip security but give me 1 hour for click2call.. mine is really only 15 minutes of stuff, his should be at least 1 hour`
`17:39.40`	`[TK]D-Fender`	`PMantis : Also Grandstream GXP-2000`
`17:40.01`	`Mimmus`	`DFS: yes`
`17:40.05`	`PMantis`	`[TK]D-Fender, Ok, I was looking to use a Grandstream, since it has paging capabilities in the latest firmware`
`17:40.10`	`DFS`	`mimmus: both servers must mirror each config then..`
`17:40.34`	`watchy`	`tkd: thanks man`
`17:40.34`	`DFS`	`mimmus: of course inversing the info for the other...`
`17:40.42`	`Mimmus`	`DFS: yes`
`17:40.49`	`masonf`	`for the record I need asterisk to be able to read write /usr/local/share/asterisk (problem solved thans mimus)`
`17:40.53`	`[TK]D-Fender`	`PMantis : plenty of ways do do paging on others. Unless you're really short of cash I'd suggest going with the Polycom IP 601, or at least the SNOM 360.`
`17:40.54`	`Mimmus`	`but I'm not a guru`
`17:41.27`	`Mimmus`	`how can I fetch last three chars from a var????? I forget it`
`17:42.39`	`PMantis`	`[TK]D-Fender, And it can show the status of a remote SIP extension? (I can't imagine how the setup works in *)`
`17:42.48`	`idpromnut`	`question: is there a listing (like a reference) of all dialplan functions/macros?`
`17:43.29`	`*** part/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk)`
`17:43.35`	`rue_work`	`app.c: No audio available on IAX2/astpbx-woodbay-3?? < - I think that has something to do with my voicemail audio problem`
`17:44.08`	`[TK]D-Fender`	`PMantis : yes, that is the exact point for it.`
`17:44.25`	`DFS`	`mimmus: do you set the type as user, friend or peer?`
`17:44.25`	`PMantis`	`[TK]D-Fender, Ok, I'll have to take your word for it. :)`
`17:44.39`	`[TK]D-Fender`	`PMantis : exten => 1000,hint,SIP/1000`
`17:44.44`	`[TK]D-Fender`	`PMantis : exten => 1000,1,Dial(SIP/1000,20)`
`17:44.50`	`[TK]D-Fender`	`like that in *`
`17:44.54`	`[TK]D-Fender`	`thats all`
`17:44.57`	`PMantis`	`hint? hmmmmm`
`17:45.05`	`[TK]D-Fender`	`its a priority on the exten`
`17:45.07`	`Mimmus`	`DFS: peer if they are peers!`
`17:45.32`	`[TK]D-Fender`	`then theres the setup on the phone istelf which varies between mfg's`
`17:45.55`	`*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)`
`17:45.55`	`*** join/#asterisk chapeaurouge (n=chap@85.201.81.201)`
`17:46.12`	`DFS`	`mimmus: just checking....`
`17:46.47`	`rue_work`	`how can I tell which errors are actaully causing me problems?`
`17:47.07`	`[TK]D-Fender`	`rue_work : Ok, first, whats not working to your satisfaction?`
`17:48.07`	`rue_work`	`well, we keep haiving lose all the audio on voicemails`
`17:48.16`	`rue_work`	`it records long blank files`
`17:48.22`	`PMantis`	`[TK]D-Fender, Thanks, I found a wiki entry on voip-info`
`17:48.31`	`watchy`	`does sjphone support voicemail?`
`17:48.33`	`PMantis`	`[TK]D-Fender, Makes more sense now.`
`17:48.52`	`[TK]D-Fender`	`rue_work : Where are the calls coming in from? If you leave a message directly from a phone connected to * do you get sound then?`
`17:49.10`	`rue_work`	`for the most part, messages work`
`17:49.26`	`[TK]D-Fender`	`watchy : typically you don't let the SIP client do its own voicemail handling... you do it in the server.`
`17:49.28`	`rue_work`	`these calls are comming in from a T1 to our PSTN machine`
`17:49.35`	`watchy`	`hrm`
`17:49.40`	`watchy`	`oops`
`17:49.51`	`[TK]D-Fender`	`rue do incoming calls have audio at all?`
`17:49.54`	`watchy`	`ah`
`17:49.58`	`watchy`	`it does support voicemail`
`17:49.59`	`rue_work`	`yes`
`17:50.04`	`watchy`	`it just notified sjphone`
`17:50.26`	`[TK]D-Fender`	`rue_work : So only in voicemail you lose all audio?`
`17:50.29`	`rue_work`	`[TK]D-Fender it happens intermittently`
`17:50.51`	`rue_work`	`that voicemails comming in on the t1 have no audio`
`17:51.10`	`[TK]D-Fender`	`rue_work : Well it would basically mean ALL CALLS on the T1 then.`
`17:51.36`	`rue_work`	`were using it right now, all the calls are fine`
`17:51.40`	`[TK]D-Fender`	`pastebin the CLI of a call coming in and trying to leave a VM.`
`17:51.56`	`watchy`	`hey tk: what do put in sip.conf to tell the phone its voicemail # so my VM button works?`
`17:52.09`	`rue_work`	`sorry, can you give me more detail on how to do that?`
`17:52.15`	`[TK]D-Fender`	`watchy : thats not sip.conf's job, thats a setting on your PHONE.`
`17:52.28`	`watchy`	`oh`
`17:52.33`	`rue_work`	`so far, this seems to be limited to the voicemail`
`17:52.36`	`justinu`	`is fender singlehandedly helping 5 newbies at once again?`
`17:52.38`	`watchy`	`so i'd push that out with like sipdefault.cnf?`
`17:52.39`	`[TK]D-Fender`	`rue_work : copy the CLI output of a call that is attempting to leave a VM and shove it in a pastebin.`
`17:53.14`	`rue_work`	`[TK]D-Fender from /var/log/asterisk/full ?`
`17:53.21`	`[TK]D-Fender`	`Actually, only 3 this time :)`
`17:53.25`	`watchy`	`messages_uri`
`17:53.26`	`watchy`	`<PROTECTED>`
`17:53.26`	`watchy`	`<PROTECTED>`
`17:53.27`	`watchy`	`ah!`
`17:53.32`	`[TK]D-Fender`	`rue_work : no from "asterisk -rvvvvvv"`
`17:53.39`	`rue_work`	`ok`
`17:54.08`	`[TK]D-Fender`	`watchy : so you'd set that to either 98 or 97[box] per the context I gave you`
`17:54.33`	`[TK]D-Fender`	`watchy : since all of my home uses 1 box I use *970 (box 0) on my SPA-941's VM key`
`17:54.36`	`hardwire`	`ok..`
`17:54.44`	`hardwire`	`is there a good test suite for measuring rtp loss`
`17:54.58`	`justinu`	`hardwire: not really`
`17:55.13`	`watchy`	`thanks tk`
`17:55.16`	`hardwire`	`I am trying to measure loss using icmp.. which most routers basically filter or throttle.`
`17:55.22`	`justinu`	`hardwire: you need to rely on RTCP which asterisk doesn't support, but there's a dodgy patch for`
`17:55.41`	`hardwire`	`justinu: I was thinking they would just have to agree on a pattern. and measure loss with pattern matching.`
`17:56.14`	`justinu`	`RTCP is the answer`
`17:56.17`	`hardwire`	`or send chunks w/ a crc.. and just feather out the results.`
`17:56.33`	`*** join/#asterisk detatch (i=detent@dhcp-100.fresno-dc2.brandxnet.com)`
`17:56.34`	`hardwire`	`justinu: I just want to measure the loss.. not get around it.`
`17:56.45`	`justinu`	`you should read about what RTCP does then`
`17:56.49`	`justinu`	`it's for instrumentation`
`17:56.55`	`hardwire`	`ah`
`17:57.13`	`hardwire`	`you could use it uotside of asterisk I presume`
`17:57.19`	`justinu`	`yes`
`17:57.25`	`hardwire`	`thats all I would need`
`17:57.26`	`hardwire`	`appreciated.`
`17:57.37`	`justinu`	`a lot of media gateways support RTCP`
`17:57.44`	`detatch`	`hey everybody`
`17:57.44`	`hardwire`	`why`
`17:57.44`	`justinu`	`and most SIP phones do`
`17:58.05`	`justinu`	`because people want to know what the QoS is like`
`17:58.22`	`hardwire`	`http://en.wikipedia.org/wiki/Rtcp`
`17:58.25`	`hardwire`	`you should write about it :)`
`17:58.32`	`justinu`	`heh`
`17:58.48`	`detatch`	`can someone answer a question about my 1.2.1 extensions.conf?`
`17:58.55`	`hardwire`	`http://www.voip-info.org/wiki/view/RTCP`
`17:58.56`	`hardwire`	`hah`
`17:58.56`	`[TK]D-Fender`	`BBIAB`
`17:58.59`	`*** join/#asterisk BladeRunner05 (n=feelme@81.174.56.54)`
`17:59.08`	`*** join/#asterisk Switchplaces (n=me@72.29.237.163)`
`17:59.15`	`justinu`	`hardwire: all you need to know: http://www.faqs.org/rfcs/rfc3550.html`
`17:59.29`	`detatch`	`im upgrading from 1.0.3 to 1.2.1`
`17:59.32`	`[TK]D-Fender\|AFK`	`detatch : Pastebin it, and ask your questions I'll be back soon`
`17:59.32`	`hardwire`	`ok the control protocol..`
`17:59.34`	`[TK]D-Fender\|AFK`	`~pb`
`17:59.38`	`jbot`	`i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`17:59.38`	`hardwire`	`same as rtp but + some`
`17:59.39`	`detatch`	`thanks`
`18:00.09`	`*** join/#asterisk coolhp (n=crap@mtl149-99-190-66.dedicated.sprintdsl.ca)`
`18:00.14`	`justinu`	`it's actually not the same as RTP`
`18:00.28`	`justinu`	`RTP is used for carrying time sensitive data (like voip packets)`
`18:00.35`	`BladeRunner05`	`I'm troubling installing astGUIclient + vicidial.... I'm getting error running: ADMIN_area_code_populate.pl`
`18:00.35`	`hardwire`	`ok`
`18:00.41`	`justinu`	`RTCP is used to monitor the performance of the forward/backwards streams`
`18:00.50`	`coolhp`	`Good day all ! I was wondering : Which of the following is better/more advanced : chan_skinny, chan_sccp (from SF) or chan_sccp2 (from berlios) ?`
`18:01.03`	`rue_work`	`[TK]D-Fender\|AFK http://pastebin.com/502653`
`18:01.04`	`rue_work`	`:/`
`18:01.19`	`rue_work`	`but that is a bad example, because it worked`
`18:01.36`	`*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)`
`18:01.41`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`18:01.49`	`hardwire`	`justinu: I suppose one form of doing this is to send an rtp stream to an rtp echo server.`
`18:02.02`	`BladeRunner05`	`I'm troubling installing astGUIclient + vicidial.... I'm getting error running: ADMIN_area_code_populate.pl error is at http://pastebin.com/502654`
`18:02.07`	`justinu`	`hardwire: yeah, but then you wouldn't know if the loss was on the forward stream or the backwards stream`
`18:02.17`	`hardwire`	`justinu: sometimes I just don't want to know.`
`18:02.28`	`justinu`	`then why bother with qos at all? :P`
`18:02.34`	`hardwire`	`because I love my customers.`
`18:02.42`	`hardwire`	`hmmphm`
`18:02.47`	`justinu`	`most ATAs do RTCP also`
`18:02.55`	`hardwire`	`heh.. you could rtp a stream to one place.. then have it tcp the results back.`
`18:03.01`	`Switchplaces`	`must go today 2 alienware area51-m 7700 notebooks. price 600 for 2. message me if interested on msn at mcsltd1@hotmail.com, aim at ogd443 or yahoo at thishastogotoday. do have an auction set up on yahoo auctions for these.`
`18:03.17`	`hardwire`	`why is asterisk not on this RTCP bandwagon?`
`18:03.24`	`hardwire`	`I would assume it just comes with the territory`
`18:03.25`	`detatch`	`switch`
`18:03.26`	`detatch`	`go away`
`18:03.27`	`detatch`	`hah`
`18:03.29`	`justinu`	`hardwire: that's a good question... i would ask digium that`
`18:03.36`	`hardwire`	`I think I will.`
`18:03.38`	`hardwire`	`Attn: Digium`
`18:03.43`	`hardwire`	`Subject: RTCP in asterisk`
`18:03.44`	`NDT`	`asterisk have anyway to determine if a human answered or an answering machine without interaction like pressing a number etc?`
`18:03.46`	`hardwire`	`<PROTECTED>`
`18:03.51`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`18:03.54`	`hardwire`	`I have no idea what this really does.. can you implement it?`
`18:03.56`	`hardwire`	`.`
`18:04.00`	`NDT`	`some sort of positive call acceptance`
`18:04.30`	`detatch`	`I just upgraded from 1.0.3 to 1.2.1, we run a call center`
`18:04.31`	`justinu`	`hardwire: http://bugs.digium.com/view.php?id=2863`
`18:04.38`	`*** join/#asterisk lorinc (n=ang@caracas-3803.adsl.interware.hu)`
`18:04.52`	`detatch`	`i see a lot of messages in my /var/log/asterisk/messages about the timeout context in my extensions.conf`
`18:04.55`	`justinu`	`it's been on the digium bug tracker for over a year`
`18:05.03`	`Katty`	`hi lads.`
`18:05.11`	`detatch`	`ive posted a sample extension and the error in pastebin`
`18:05.27`	`rob0`	`afternoon Katty`
`18:05.32`	`hardwire`	`justinu: heh`
`18:05.54`	`hardwire`	`nobody seems like they want to adopt the patch`
`18:06.02`	`justinu`	`i made it work`
`18:06.14`	`justinu`	`but I haven't released it back`
`18:06.16`	`Katty`	`A-jay: please don't talk to me in private. it's rather annoying.`
`18:06.16`	`*** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk)`
`18:06.20`	`Katty`	`A-jay: instead, if you want help, talk in here.`
`18:06.43`	`nextime`	`is it possible to detect the nature of address indicator from a incoming call on one of the 3 h323 channels available?`
`18:06.49`	`lo_tech`	`no cyb0r the kat!`
`18:07.07`	`hardwire`	`justinu: they don't like you not having a disclaimer!`
`18:07.13`	`justinu`	`i know`
`18:07.18`	`hardwire`	`I have one on file there`
`18:07.20`	`hardwire`	`whardier`
`18:07.22`	`justinu`	`but I'm not willing to sign my life away just yet`
`18:07.33`	`hardwire`	`justinu: well its a patch.. for asterisk.`
`18:07.37`	`Switchplaces`	`must go today 2 alienware area51-m 7700 notebooks. price 600 for 2. message me if interested on msn at mcsltd1@hotmail.com, aim at ogd443 or yahoo at thishastogotoday. do have an auction set up on yahoo auctions for these.`
`18:07.41`	`hardwire`	`not like you are going to apply it anywhere else.`
`18:07.47`	`justinu`	`thank you`
`18:07.49`	`justinu`	`whoever did that`
`18:08.19`	`justinu`	`hardwire: if someone was actually willing to go over the code, i'd be more than happy to show them what's wrong and how to fix it, but no one seems to care.`
`18:08.21`	`hardwire`	`you don't want the laptops?`
`18:08.29`	`justinu`	`so I don't care about posting the patch`
`18:08.33`	`hardwire`	`justinu: yeh they would liekt o adopt more developers`
`18:08.33`	`hardwire`	`hehe`
`18:08.46`	`hardwire`	`give it to file.. file will eat anything.`
`18:08.54`	`*** mode/#asterisk [+b !@72.29.237.163] by denon`
`18:08.55`	`justinu`	`file just ignores me`
`18:08.58`	`denon`	`I dont think it was a real kline`
`18:09.00`	`hardwire`	`yeh`
`18:09.01`	`denon`	`I think it was just his quit msg`
`18:09.16`	`hardwire`	`file is a snobby wobby knob sometimes..`
`18:09.36`	`justinu`	`again, people don't want to work on it, i'm not gonna cram it down their throats`
`18:09.53`	`hardwire`	`don't you know thats how shit gets done?`
`18:10.04`	`justinu`	`i don't work like that`
`18:10.12`	`hardwire`	`the most successfull people in the world spend their time on planes so they can go cram their crap down as many throats as possible.`
`18:10.18`	`justinu`	`if people are going to be insular, i'll just keep it to myself as well`
`18:10.43`	`hardwire`	`hyperlinks IVR sucks`
`18:11.08`	`hardwire`	`insular is a good word of the day :)`
`18:11.23`	`hardwire`	`concidering I work on an island.. or off island with people of an island.`
`18:13.00`	`Katty`	`rob0: allo (=`
`18:14.01`	`*** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com)`
`18:14.12`	`justinu`	`which island?`
`18:14.20`	`hardwire`	`st paul island ak`
`18:14.24`	`justinu`	`cool`
`18:14.54`	`hardwire`	`the people of the world should comply with me putting them on hold when waiting to connect to them`
`18:15.18`	`*** join/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com)`
`18:15.25`	`rue_work`	`http://pastebin.com/502671 <- this error dosn't look good either`
`18:15.30`	`Mimmus`	`where can I look if Asterisk doesn't correctly bridge a call?`
`18:15.39`	`*** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`18:15.49`	`*** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`18:16.34`	`hardwire`	`Mimmus: maybe the lost packets under the bridge know the answers.`
`18:16.50`	`generalhan`	`When i try to compile asterisk 1.2.1 (after compiling libpri and zaptel) im getting this error :: collect2: ld returned 1 exit status make: *** [asterisk] Error 1 :: anyone know whats going on ?`
`18:17.01`	`*** join/#asterisk nside (i=O@Toronto-HSE-ppp3770629.sympatico.ca)`
`18:17.05`	`rue_work`	`linker error?`
`18:18.02`	`nside`	`anyone here played with libiaxclient?`
`18:18.03`	`Mimmus`	`hardwire: only with a number!!!`
`18:18.31`	`mut`	`O_O`
`18:18.34`	`hardwire`	`you are a crazy little man.`
`18:18.36`	`mut`	`alo hardwire`
`18:18.54`	`Mimmus`	`hardwire: I have (PRI) -- Asterisk -- Alcatel PBX --- analog phones`
`18:19.07`	`hardwire`	`Mimmus: I am not going to know the answer..`
`18:19.35`	`Mimmus`	`dont' worry,hardwire`
`18:19.46`	`Mimmus`	`I try anyway...`
`18:19.50`	`hardwire`	`heh`
`18:20.06`	`Mimmus`	`analog phones are unable to call a number (an automatic responder)`
`18:20.16`	`Mimmus`	`voip phones, directlyconnected to Asterisk, yes!`
`18:20.41`	`[TK]D-Fender\|AFK`	`detatch : I didn't see taht pastebin of your extensions.conf so I can help you out...`
`18:20.41`	`Mimmus`	`rest of the world works!`
`18:20.51`	`Dorphalsig`	`I have an E1 link and a TE400 card, every few hours * issues this warning: No D-Channels found. Using Primary on channel 16 anyway and then nothing works again untill I restart zaptel, could anybody give me a hand?`
`18:21.02`	`*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`18:21.51`	`Mimmus`	`can be echocancel=800?`
`18:23.01`	`*** join/#asterisk \|omni\| (n=rob@207.88.74.98)`
`18:23.13`	`detatch`	`i just posted it`
`18:23.18`	`*** part/#asterisk nside (i=O@Toronto-HSE-ppp3770629.sympatico.ca)`
`18:23.41`	`detatch`	`TK: http://pastebin.com/502659`
`18:23.45`	`[TK]D-Fender`	`detatch : paste the link here please...`
`18:23.47`	`[TK]D-Fender`	`thx`
`18:23.53`	`detatch`	`but you know what i think i just figured it out`
`18:24.23`	`[TK]D-Fender`	`detatch : you have 2 exten entries for t in there!`
`18:24.24`	`detatch`	`its because ive got two t,1`
`18:24.35`	`detatch`	`it should work if i did t,2 right`
`18:24.36`	`detatch`	`?`
`18:25.12`	`[TK]D-Fender`	`might be a good idea. mind you I have no idea what you do in those macro's...`
`18:25.42`	`detatch`	`basically play a greetin message "press one for tech support, 2 for blah" and then macro systatus gives a network status and dumps into a queue`
`18:25.43`	`[TK]D-Fender`	`detatch : Maybe pastebin more of your setup. I think there might be some optimising to do...`
`18:26.02`	`rue_work`	`[TK]D-Fender http://pastebin.com/502671 that look suspicious to you? The voicemail I did worked, so the data will be no good t you but this clip was somemthing I saw going by, there were no files int eh users voicemail directory`
`18:26.40`	`Mimmus`	`does anyone know look at 'pri debug' output to find why * fails to call a number?`
`18:26.42`	`rue_work`	`line 20 is what I'm wondering about`
`18:26.53`	`[TK]D-Fender`	`rue_work : Can you try again without the exdcessive debug?`
`18:27.05`	`detatch`	`Tk, thanks, i feel like such an ass`
`18:27.21`	`rue_work`	`[TK]D-Fender this is an intermittent problem, I cant just make it happen`
`18:27.38`	`detatch`	`TK, the funny thing is 1.0.3 didnt mind a dialplan like that at all`
`18:27.38`	`[TK]D-Fender`	`rue_work : No decetable pattern to it?`
`18:27.42`	`*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)`
`18:27.47`	`rue_work`	`[TK]D-Fender no`
`18:28.18`	`mut`	`anyone here much of a cisco pix expert?`
`18:28.36`	`Rawplayer`	`#cisco`
`18:30.32`	`[TK]D-Fender`	`Umm... Wiki seem down for everyone?`
`18:31.30`	`*** join/#asterisk Seggy (i=rbutler@tsss.org)`
`18:31.34`	`*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`18:32.25`	`Mimmus`	`Cresl1n: here?`
`18:32.33`	`detatch`	`seems so`
`18:35.53`	`Mimmus`	`detatch: uh?`
`18:36.03`	`detatch`	`hmm`
`18:36.04`	`detatch`	`?`
`18:36.22`	`Mimmus`	`detatch: same person!`
`18:36.40`	`detatch`	`sorry`
`18:36.41`	`detatch`	`you lost me`
`18:37.06`	`Mimmus`	`I lost myself too.. here it's 19:37`
`18:37.28`	`*** join/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`18:38.02`	`detatch`	`uhg`
`18:38.06`	`detatch`	`too early for me`
`18:38.12`	`*** part/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net)`
`18:38.26`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`18:38.39`	`Mimmus`	`detatch: any residual chance to got some help?`
`18:38.49`	`*** join/#asterisk dwildes (n=dwildes@209.164.237.195)`
`18:38.56`	`*** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com)`
`18:39.13`	`detatch`	`what cha got`
`18:39.21`	`hardwire`	`http://www.voipsupply.com/product_info.php?products_id=1002`
`18:39.22`	`Mimmus`	`get !`
`18:39.26`	`hardwire`	`anybody use this?`
`18:39.57`	`justinu`	`no, but that stuff looks hot`
`18:40.41`	`hardwire`	`its cute`
`18:40.58`	`*** join/#asterisk evilrabbi (i=evilrabb@64.123.157.130)`
`18:41.14`	`evilrabbi`	`has anyone here used AMP?`
`18:41.43`	`DFS`	`mimmus: the peer to peer worked!!`
`18:41.54`	`*** join/#asterisk lesouvage (n=lesouvag@82.74.11.143)`
`18:41.56`	`Mimmus`	`DFS: well, I'm happy`
`18:42.08`	`DFS`	`mimmus: thanks for the assistance`
`18:42.47`	`Mimmus`	`DFS: I'm happy to receive (99%) and give (1%) help`
`18:43.06`	`detatch`	`arent we all?`
`18:43.26`	`rob0`	`my ratio tends to differ quite a bit ... but not in this channel. :)`
`18:43.49`	`fndude`	`Is there a test number I can dial for pots that will give me my originating number?`
`18:44.15`	`Mimmus`	`detatch: you can still have a chance to improve looking at http://pastebin.com/502717`
`18:44.27`	`Mimmus`	`and find why this call is not answered!`
`18:44.56`	`evilrabbi`	`fndude yeah`
`18:44.57`	`evilrabbi`	`brb`
`18:46.02`	`[TK]D-Fender`	`evilrabbi : Please try #amportal`
`18:46.41`	`*** join/#asterisk ToTo (n=ToTo@host243-91.pool8260.interbusiness.it)`
`18:47.14`	`[TK]D-Fender`	`I tend to give (95%) and get (%5) here`
`18:47.51`	`*** join/#asterisk Defraz (n=t0tal@72.24.26.215)`
`18:48.02`	`evilrabbi`	`fndude they number doesn't work anymore`
`18:48.19`	`evilrabbi`	`goto somethingl ike oldschoolphreak.com`
`18:48.21`	`evilrabbi`	`they should have one`
`18:48.42`	`evilrabbi`	`www.oldskoolphreak.com`
`18:48.43`	`[TK]D-Fender`	`fndude : call someone with callerID.`
`18:49.53`	`evilrabbi`	`btw [TK]D-Fender thank you =)`
`18:50.41`	`*** join/#asterisk quiardon (n=chiardon@200.71.58.39)`
`18:51.34`	`quiardon`	`hello`
`18:51.48`	`Dorphalsig`	`erre`
`18:52.30`	`fndude`	`Fender: yeah I tried that I am getting 'unknown'.`
`18:54.34`	`fndude`	`Heh sux. I finally understand asterisk and can use it, now I am having to figure out the pots side to the mess. Go figure.`
`18:54.57`	`[TK]D-Fender`	`fndude : unknonw name, number or both?`
`18:55.25`	`fndude`	`[TK]D-Fender: unknown.`
`18:55.38`	`*** join/#asterisk elvisthedj (n=Johnny@th20.montanavision.com)`
`18:56.01`	`[TK]D-Fender`	`fndude : try 514-940-8223. I'll see if I can see it on my end`
`18:56.22`	`[TK]D-Fender`	`nope`
`18:56.24`	`[TK]D-Fender`	`nothing`
`18:56.29`	`[TK]D-Fender`	`guess you're blocked`
`18:56.31`	`elvisthedj`	`ok, i'm ready to paypal somebody to get this cisco 7940 working :)`
`18:56.40`	`fndude`	`boooo.`
`18:56.41`	`elvisthedj`	`somebody name your price`
`18:56.53`	`[TK]D-Fender`	`fndude : call your telco and have them ID the line.`
`18:56.55`	`fndude`	`[TK]D-Fender: thx anyway.`
`18:57.22`	`evilrabbi`	`elvisthedj have you updated the firmware yet?`
`18:58.59`	`Mimmus`	`it's 20:00, I'm dead, bye`
`18:59.34`	`Katty`	`what's it mean when traceroute just gives you *s?`
`18:59.37`	`Katty`	`like 1. * * *`
`18:59.40`	`fndude`	`[TK]D-Fender: well the lowdown is, I tried to get termination to DID yesterday, some how the provider used my cell # . After the confusion, they told me that they had not gotten a new number, and were waiting for some reason. Is it that hard to get or change these numbers?`
`18:59.41`	`*** join/#asterisk Cinen-Alt (n=Cinen@64.128.219.131)`
`18:59.53`	`elvisthedj`	`Katty, maybe icmp is blocked for you?`
`18:59.58`	`Mimmus`	`thanks again to all, especially Cresl1n`
`19:00.02`	`elvisthedj`	`Katty, can you ping things?`
`19:00.02`	`Katty`	`elvisthedj: huh?`
`19:00.08`	`Katty`	`mister fender.`
`19:00.15`	`Katty`	`what do *s mean in traceroute?`
`19:00.23`	`justinu`	`no response`
`19:00.28`	`*** join/#asterisk _Sam-- (n=sam@dca.kneedraggers.com)`
`19:02.53`	`*** join/#asterisk vandien (i=sted@aditu.dahltronics.de) [NETSPLIT VICTIM]`
`19:03.36`	`*** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr)`
`19:03.39`	`jhiver`	`hi all`
`19:03.51`	*** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) [NETSPLIT VICTIM]
`19:04.09`	`*** join/#asterisk KryoStoffer (n=kri@helium.kri.dk) [NETSPLIT VICTIM]`
`19:04.10`	`jhiver`	`is anybody using Asterisk in combination with SER? I am trying to forward * calls to phones registered to SER without much success`
`19:04.43`	`jhiver`	`it goes`
`19:04.45`	`jhiver`	`Executing Dial("Zap/3-1", "SIP/jhiver@ser.ykoz.net") in new stack`
`19:04.49`	`jhiver`	`which is fine`
`19:04.59`	`_Sam--`	`i asked earlier and apologize if someone asnwered..i didnt see...but does this cheap D-link packet prioritizer recognize IAX so that it can prioritize it? http://www.voipsupply.com/product_info.php?manufacturers_id=45&products_id=1168`
`19:05.03`	`jhiver`	`but then it says "Failed to authenticate on INVITE to '"0" <sip:0@83.206.114.91>;tag=as13a66c99'"`
`19:05.03`	`*** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) [NETSPLIT VICTIM]`
`19:05.03`	`*** join/#asterisk alrs (n=lars@69-160-242-101.vnnyca.adelphia.net)`
`19:05.03`	`*** join/#asterisk file (n=joshnet@mctnnbsa24w-142167048221.nb.aliant.net) [NETSPLIT VICTIM]`
`19:05.04`	`*** join/#asterisk hnupik (n=hnupik@chello082119119139.chello.sk) [NETSPLIT VICTIM]`
`19:05.04`	`*** join/#asterisk CoolAcid (n=jason@216.99.98.39) [NETSPLIT VICTIM]`
`19:05.04`	`*** join/#asterisk Cinen (n=Cinen@vpn.triadtelecom.com)`
`19:05.04`	`*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM]`
`19:05.04`	`*** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) [NETSPLIT VICTIM]`
`19:05.04`	`*** join/#asterisk memic (n=memic@chicago089.server4free.de) [NETSPLIT VICTIM]`
`19:05.04`	`*** mode/#asterisk [+o twisted[asteria]] by irc.freenode.net`
`19:05.22`	`*** join/#asterisk memic (n=memic@chicago089.server4free.de)`
`19:05.36`	`rue_work`	`[TK]D-Fender http://pastebin.com/502671 I'm trying to find things that might point to the source of the problems...`
`19:05.39`	`*** join/#asterisk Cinen-Alt (n=Cinen@vpn.triadtelecom.com)`
`19:08.36`	`*** join/#asterisk bhickey_ (n=chatzill@212.2.174.21)`
`19:09.30`	`PoWeRKiLL`	`if I have 3 asterisk box logging cdr to a central mysql server what is the probabilty to get same uniqueid ?`
`19:09.33`	`*** part/#asterisk secure75 (n=mic@dslb-084-057-013-245.pools.arcor-ip.net)`
`19:10.08`	`*** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)`
`19:10.09`	`jhiver`	`PoWeRKill, depends what "uniqueid" are you referring to?`
`19:10.15`	`*** part/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com)`
`19:11.59`	`jhiver`	`damn this chan is dead`
`19:13.08`	`watchy`	`lets hug`
`19:13.21`	`jhiver`	`(Asterisk \| Emacs\| Gnome \| PHP) is (Better than\| Worse than \| Blows the socks of) (OpenPBX \| Vi \| KDE \| Perl)!`
`19:13.33`	`jhiver`	`welcome to the _META_TROLL_`
`19:13.36`	`Nugget`	`you forgot mysql \| postgresql.`
`19:13.41`	`jhiver`	`damn :)`
`19:14.02`	`rob0`	`are those indexed arrays?`
`19:14.14`	`docelm0`	`my dCAP plaque arrived today!!!!`
`19:14.18`	`docelm0`	`YIPPIE!`
`19:14.28`	`*** join/#asterisk Paol1 (n=paolo@217.220.155.234)`
`19:14.39`	`docelm0`	`say was 1.2 released before or after astricon?`
`19:14.45`	`docelm0`	`05`
`19:15.00`	`rob0`	`can you say, for instance, "Asterisk is Blows the socks of Perl"?`
`19:15.54`	`[TK]D-Fender`	`Apples > Oranges`
`19:16.02`	`*** part/#asterisk Paol1 (n=paolo@217.220.155.234)`
`19:16.27`	`jhiver`	`yeah why not`
`19:16.36`	`quiardon`	`hello`
`19:16.36`	`jhiver`	`you _can_ say it, doesn't mean it has to make sense :)`
`19:16.37`	`rob0`	`Oranges is Blows the socks of Apples`
`19:16.48`	`Nugget`	`I quit smoking 8 years, 8 months, 1 week, 18 hours, 16 minutes, and 47 seconds ago. During that time, I would have smoked 69,764 cigarettes. (That's like smoking a 3.30 mile-long cigarette) By quitting, I've saved $12,208.70! (That's 9 Apple 23" Cinema Displays and change) I've avoided inhaling 1.81 kg of tar, 111 grams of nicotine, and 1.12 kg of carbon monoxide.`
`19:16.50`	`rob0`	`true, if it made sense it might not be a good troll`
`19:17.09`	`*** part/#asterisk pigpen (n=mark@fw.seamans.cc)`
`19:17.59`	`tzafrir_home`	`1.2 was released awy after astricon`
`19:18.09`	`docelm0`	`shit..`
`19:18.15`	`docelm0`	`I have to take the damn test again..`
`19:18.44`	`Zodiacal`	`anyone know if cisco 7900 serise phones work well with sip & *? the forth paragraph on this page titled "Note:" seems to say other wise... http://www.voip-info.org/wiki/view/Setup+SiP+on+7940+-+7960`
`19:19.03`	`tzafrir_home`	`anybody heard anything about asterisk and libjingle (googletalk)?`
`19:19.13`	`Nugget`	`the cisco sip firmware isn't quite as fancy as the native firmware, but the phones work just fine with asterisk.`
`19:19.37`	`Zodiacal`	`nagget not as fancy?`
`19:19.51`	`Zodiacal`	`nagget can they still grab xml?`
`19:19.56`	`marcus2`	`man`
`19:20.01`	`marcus2`	`meetme works great for us with zap channels`
`19:20.05`	`Nugget`	`yes, but not all the stuff that the SCCP firmware can.`
`19:20.09`	`Nugget`	`and you can't push XML to the phone`
`19:20.13`	`marcus2`	`but when we start adding iax2 channels, theres all sorts of echo and wierd shit`
`19:20.28`	`tzafrir_home`	`mog_work, hmm, how about HURD?`
`19:21.57`	`Zodiacal`	`nugget which brand would you recomend?`
`19:22.01`	`Zodiacal`	`nugget polycom?`
`19:22.19`	`mog_work`	`mmm polycom`
`19:22.25`	`Nugget`	`I have 7960's, personally, but lots of people I respect favor the polycoms.`
`19:22.29`	`Nugget`	`avoid grandstream.`
`19:22.30`	`justinu`	`mmm, chicken nuggets`
`19:23.02`	`rue_work`	`tzafrir_home hu me?`
`19:23.11`	`watchy`	`i only have 2 7960s but i like them`
`19:23.16`	`watchy`	`i havent tried a poly`
`19:23.26`	`[TK]D-Fender`	`Zodiacal : How many phones do you need? How many need speakerphone? How many "receptionist" type phones?`
`19:23.31`	`dogtanian`	`Zodiacal: 7960 works excellently with SIP`
`19:23.44`	`dogtanian`	`Zodiacal: and *`
`19:23.55`	`watchy`	`yea it does`
`19:24.04`	`[TK]D-Fender`	`Polycom is noticably cheaper than Cisco for the same class of phones.`
`19:24.07`	`Zodiacal`	`[td]d-fender i was planing to get six 7960's and four 7912's`
`19:24.17`	`justinu`	`any polycom isn't quite so nazi about firmware`
`19:24.23`	`rue_work`	`[TK]D-Fender I didn't show you teh extentions file did I?`
`19:24.30`	`dogtanian`	`yeah - same here... i went straight to buying 2x 7960's and i don't regret it at all`
`19:24.32`	`Zodiacal`	`im new to asterisk obvously, does asterisk support SCCP very well?`
`19:24.35`	`*** part/#asterisk Mystique (n=mystique@mystique.poklib.org)`
`19:24.46`	`justinu`	`ask qwell about sccp :P`
`19:24.51`	`[TK]D-Fender`	`Zodiacal : Then I'd say get IP601's instead of 7960's, and IP301's instead of 7912's and save a lot of $`
`19:25.03`	`Nugget`	`you'll definitely be swimming upstream if you want to give SCCP a go with asterisk.`
`19:25.04`	`watchy`	`i'm using 7960s to learn *`
`19:25.15`	`[TK]D-Fender`	`rue_work : no, and you never gave me CLI output WITHOUT the excess debugging.`
`19:25.29`	`*** join/#asterisk pigpen (n=mark@fw.seamans.cc)`
`19:25.31`	`Strom_C`	`I use the 7960s with SIP firmware without problems`
`19:25.33`	`rue_work`	`ok, let me start with extentions.conf`
`19:26.03`	`watchy`	`i had alot of probs with my 7960. You can fuck it up if you tell it a image file in the .cnf and you aint got it`
`19:26.11`	`Strom_C`	`hi`
`19:26.11`	`watchy`	`i didnt have sip images to fix my phone either`
`19:26.19`	`watchy`	`i finnaly got them though`
`19:26.37`	`Zodiacal`	`strom_c have you used xml a lot with it? i would like to program dynamic content for it..`
`19:26.46`	`Strom_C`	`no`
`19:26.53`	`*** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br)`
`19:27.15`	`[TK]D-Fender`	`7912G = $210, IP300 = $114. 7960G = $340, IP601 = $250. Easy choice.`
`19:27.22`	`dogtanian`	`the cisco 79xx's are a bit of a pig about upgrading their firmware... you normally have to upgrade version-by-version`
`19:27.24`	`Strom_C`	`I use it just as a high-quality digital desk set`
`19:27.29`	`Zodiacal`	`tkd-fender not really my money :P`
`19:27.38`	`justinu`	`the ip601 is gets an A+ in my book`
`19:27.39`	`brad_mssw`	`anyone have a recommended voip supplier that provides iax access ?`
`19:27.42`	`justinu`	`the speakerphone is just magic`
`19:27.43`	`Zodiacal`	`tkd-fender i just want the best features...`
`19:27.50`	`[TK]D-Fender`	`Zodiacal : Well also not so much a PITA to get firmware either :)`
`19:28.13`	`[TK]D-Fender`	`Zodiacal : Features-wise they are near-par. Both great phones.`
`19:28.21`	`justinu`	`featurewise, i don't think cisco has anything on polycom`
`19:28.21`	`Zodiacal`	`tkd-fender yeah i have read that about it.. but i think i have access to the firmware... well i'll know in a few days`
`19:28.37`	`*** join/#asterisk pigpen (n=mark@fw.seamans.cc)`
`19:28.54`	`Zodiacal`	`that one comment on that wiki posted a few mins ago makes me nervous. talks about xml time outs and not being able to create sip soft keys...`
`19:29.10`	`[TK]D-Fender`	`Zodiacal : Other thing is the Cisco's were designed for Cisco pre-std PoE and don't include a wall adapter all the time. The exact opposite for Polycom.`
`19:29.24`	`justinu`	`only the 601`
`19:29.31`	`Zodiacal`	`yeah i was going to get a poe switch.. ~350..`
`19:29.34`	`justinu`	`be careful, the 501 is cisco CDP`
`19:29.40`	`Zodiacal`	`$350`
`19:29.47`	`justinu`	`not 802.3af PoE`
`19:29.51`	`[TK]D-Fender`	`justinu : huh?!`
`19:30.04`	`[TK]D-Fender`	`IP 501 = 802.3af....`
`19:30.07`	`justinu`	`the 501 out of the box won't work with a PoE switch`
`19:30.12`	`justinu`	`you need a special cable`
`19:30.16`	`Zodiacal`	`justinu cisco's arn't 802.3af?`
`19:30.17`	`Zodiacal`	`yikes`
`19:30.21`	`justinu`	`nope`
`19:30.25`	`justinu`	`you need a converter`
`19:30.26`	`Zodiacal`	`that sucks`
`19:30.29`	`justinu`	`it does`
`19:30.45`	`[TK]D-Fender`	`justinu : correct you need an adapter cable, but with it you do get 802.3af. Cisco you're STUCK with pre-802.3af standard`
`19:30.56`	`Zodiacal`	`cisco's wall adapters are about 30bux (3rd party)`
`19:31.22`	`justinu`	`fender: actually, i think the same adaptors work on the ip501 and cisco 7960`
`19:31.23`	`[TK]D-Fender`	`Cisco = good \| != cheap :D`
`19:31.27`	`Zodiacal`	`tk-dender know what that cable is called?`
`19:31.54`	`[TK]D-Fender`	`justinu : SOME releases of the 7960G were dual-compliant, but only the more recent stuff. the 7912 = DOA`
`19:31.58`	`Zodiacal`	`i see the $ u are talking about now..`
`19:32.01`	`Zodiacal`	`extra power cables etc..`
`19:32.04`	`[TK]D-Fender`	`Zodiacal : which cable?`
`19:32.14`	`Zodiacal`	`[TK]D-Fender> justinu : correct you need an adapter cable, but with it you do get 802.3af. Cisco you're STUCK with pre-802.3af standard`
`19:32.18`	`*** join/#asterisk enemy^x (n=null@85.196.70.98)`
`19:32.23`	`*** join/#asterisk pigpen (n=mark@fw.seamans.cc)`
`19:32.30`	`[TK]D-Fender`	`Zodiacal : here, check this out http://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-36931336704.htm`
`19:32.38`	`enemy^x`	`I was disconnected in case someone answered my last q`
`19:32.58`	`justinu`	`if you want to use 802.3af with cisco, check into these: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-44101559296.htm`
`19:33.04`	`rue_mohr`	`[TK]D-Fender you get that pastebin?`
`19:33.07`	`[TK]D-Fender`	`You'd only need a PoE cable for IP301/501 and Atacom hasa great deal on the 301 with it.`
`19:33.19`	`[TK]D-Fender`	`rue_mohr :nope`
`19:33.28`	`[TK]D-Fender`	`errr.. yes!`
`19:33.28`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`19:33.43`	`rue_mohr`	`ok, sorry, not that I dont trust anyone but :)`
`19:33.51`	`chiardon`	`hello`
`19:34.27`	`chiardon`	`nothing about our unavailable D channel problem? we are having a big comunications emergency!!`
`19:35.15`	`*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)`
`19:36.12`	`gambolputty`	`Is Unix time in a variable at all in *?`
`19:37.59`	`chiardon`	`Should I do somwe IRQs set ups to manage or compensate some timming inconsitencies between card and hdsl modem from an E!???`
`19:38.37`	`*** join/#asterisk _di (n=disnider@papados.sferos.com)`
`19:38.58`	`justinu`	`chiadron: timing is likely your problem, but I'm not experienced in solving issues like that with your specific T1 board`
`19:39.02`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`19:39.21`	`brad_mssw`	`anyone have any experience with vonwordwide (aka VON) ?`
`19:40.15`	`chiardon`	`Exquse .. E1 board!!`
`19:40.29`	`justinu`	`or E1 board`
`19:41.01`	`*** join/#asterisk BladeRunner05 (n=gianni@adsl-52-214.37-151.net24.it)`
`19:41.43`	`chiardon`	`justiniy . . .tring to do some IRQs modificaytions on the CPU Bios could help to compensate the timmings unsincronized??`
`19:41.55`	`justinu`	`it's certainly possible, yes`
`19:41.58`	`*** join/#asterisk _di (n=disnider@papados.sferos.com)`
`19:42.55`	`chiardon`	`what irq should be assinged?`
`19:43.40`	`justinu`	`chiardon: paste /proc/interrupts`
`19:44.03`	`justinu`	`make sure e1 card module isn't sharing irq with anything else`
`19:44.14`	`brad_mssw`	`anyone using broadvoice, and knows how many simultaneous calls they allow on an account?`
`19:44.20`	`justinu`	`one, i think`
`19:45.07`	`brad_mssw`	`hmm`
`19:45.12`	`elvisthedj`	`ok, we'll start the bidding at 10 bucks for somebody to help me upgrade this 7940 firmware.. (paypal required) .. 10 bucks.. anybody?`
`19:45.14`	`_di`	`hi guys, i have one problem, just installed new asterisk , with sample confs. i have lill problem with sip, it dont want to work. asterisk is loading normally, without any warnings, but it dont bind on default sip port e.g. 5060`
`19:45.34`	`justinu`	`elvisthedj: i'd help you, but dunno about cisco specifics`
`19:45.38`	`justinu`	`only polycom`
`19:46.00`	`chiardon`	`I cant see the damn card :(`
`19:46.10`	`elvisthedj`	`justinu, I'd pay cisco instead of someone on the asterisk channel.. but i hate cisco.. specifically`
`19:46.17`	`elvisthedj`	`:)`
`19:46.22`	`justinu`	`lol`
`19:46.23`	`*** join/#asterisk Meaty (n=cp_simbu@office.abi.ca)`
`19:46.33`	`justinu`	`chiardon: paste cat /proc/interrupts`
`19:47.27`	`*** join/#asterisk marv[work] (n=timr@64.89.118.139)`
`19:47.39`	`elvisthedj`	`i'll throw this out in case there are some cisco folk looking .. 7940 - bought off of ebay - want to either update sccp or change to sip.. don't care.`
`19:47.53`	`*** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca)`
`19:48.06`	`chiardon`	`www.pastebin.com/502814`
`19:48.35`	`elvisthedj`	`phone asks tftp for sep<mac> file.. (get's it).. then asks for 7960-font.xml .. then loops. never asks for a load or the os79xx.txt file`
`19:48.37`	`chiardon`	`justinu www.pastebin.com/502814`
`19:49.05`	`justinu`	`chiardon: zaptel loaded?`
`19:49.06`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`19:49.30`	`brad_mssw`	`anyone have a recommended iax or sip provider ?`
`19:49.32`	`chiardon`	`trying to...`
`19:49.56`	`elvisthedj`	`brad_mssw, I use teliax and junctionnetworks for iax .. broadvoice for sip`
`19:49.56`	`brad_mssw`	`need to be able to have 10+ simultaneous calls active`
`19:50.10`	`elvisthedj`	`brad_mssw, then scratch broadvoice for sip`
`19:50.12`	`brad_mssw`	`elvisthedj: we're using teliax now, it's not working right anymore ...`
`19:50.24`	`brad_mssw`	`elvisthedj: latency has shot through the roof`
`19:50.34`	`brad_mssw`	`elvisthedj: 70+ms`
`19:50.39`	`elvisthedj`	`brad_mssw, weird.. i haven't had problems (i just tried to tab complete the word problems .. wow)`
`19:50.43`	`justinu`	`70ms isn't usually a problem`
`19:50.43`	`chiardon`	`trying to load it`
`19:51.11`	`brad_mssw`	`justinu: yeah, 70ms at best ... and we try to fax over it ... when they were around 50ms, it was fine`
`19:51.19`	`justinu`	`ah, fax`
`19:51.25`	`elvisthedj`	`brad_mssw, have you looked at junctionnetworks? i've never checked into that number of simultaneous channels, but i've had good service`
`19:51.26`	`brad_mssw`	`justinu: dropped calls and choppyness lately though too`
`19:51.26`	`justinu`	`you need basically ethernet for fax to work right`
`19:51.35`	`brad_mssw`	`elvisthedj: never heard of junctionnetworks`
`19:51.41`	`justinu`	`junction seems good`
`19:51.53`	`brad_mssw`	`justinu: yeah, don't do much faxing, i'd be happy with 50% success rate`
`19:51.59`	`justinu`	`lol`
`19:52.06`	`elvisthedj`	`ok.. The bounty is up to 20 bucks. I have all the bin files,so you don't have to send me anything "illegal".. just help get the files on the phone`
`19:52.07`	`brad_mssw`	`justinu: we were at around 90% ... now we're at 2%`
`19:53.01`	`justinu`	`brad_mssw: a lot of these smaller voip providers seem to have these problems`
`19:53.04`	`brad_mssw`	`elvisthedj: do you have an ip for their iax gateway I can ping to see latency ?`
`19:53.16`	`dpryo`	`When I originate an outbound call through the asterisk manager, that is connected to a local extension (AGI-script), how do I get the 'called number'?`
`19:53.18`	`brad_mssw`	`elvisthedj: or hostname`
`19:53.30`	`brad_mssw`	`justinu: how big is junctionnetworks? any clue?`
`19:53.41`	`elvisthedj`	`brad_mssw, i'm not in the right spot :( i'm looking on their site`
`19:53.53`	`justinu`	`brad: no idea`
`19:54.02`	`*** join/#asterisk kiwnix (n=egarcia@219.red-82-158-158.user.auna.net)`
`19:54.07`	`elvisthedj`	`brad_mssw, hang on.. i'll just login and get the setup stuff`
`19:54.09`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)`
`19:54.16`	`brad_mssw`	`iax.jnctn.net it looks like`
`19:54.25`	`elvisthedj`	`brad_mssw, sounds right :0`
`19:54.35`	`rue_mohr`	`[TK]D-Fender ?`
`19:54.56`	`*** join/#asterisk DrWho (n=MIKE@mike-new.tc3net.com)`
`19:55.48`	`[TK]D-Fender`	`rue_mohr : yeah? I responded in PM`
`19:56.04`	`rue_mohr`	`did you remember to register your nick or log in first?`
`19:56.10`	`[TK]D-Fender`	`yup`
`19:56.14`	`rue_mohr`	`???`
`19:56.32`	`rue_mohr`	`I didn't get the answer....`
`19:56.36`	`[TK]D-Fender`	`reconfirmed`
`19:56.47`	`*** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com)`
`19:56.53`	`rue_mohr`	`there we go`
`19:56.55`	`elvisthedj`	`brad_mssw, sip proxy is sip.jnctn.net .. trunk is sip.jnctn.net`
`19:56.55`	`[TK]D-Fender`	`Just repasted`
`19:57.19`	`brad_mssw`	`elvisthedj: about 44ms`
`19:57.57`	`*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)`
`19:59.47`	`elvisthedj`	`21 dollars now people.. Help me turn this piece of plastic into a 7940`
`20:00.22`	`*** join/#asterisk jontow (i=jontow@secure-bsd.be)`
`20:00.56`	`watchy`	`what you offering $ for?`
`20:00.59`	`tzanger`	`cool`
`20:01.08`	`watchy`	`sex?`
`20:01.10`	`tzanger`	`I have the HS850 now (had an HS810 and the treo-branded bt headset)`
`20:01.22`	`_di`	`est' russike?`
`20:01.46`	`*** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net)`
`20:02.16`	`dos000`	`anyone knows an ATA that allows forwarding an inbound call from the internet via the fxo port ?`
`20:02.56`	`[TK]D-Fender`	`dos000 : SPA-3000`
`20:03.17`	`dos000`	`[TK]D-Fender, nice .. even linksys then ?`
`20:04.16`	`*** join/#asterisk file (n=joshnet@mctnnbsa24w-142167048221.nb.aliant.net)`
`20:04.46`	`*** join/#asterisk santiago (n=santiago@208.195.215.97)`
`20:04.58`	`*** join/#asterisk rizzo (n=rizzo@gentoo/developer/rizzo)`
`20:05.15`	`[TK]D-Fender`	`dos000 : Sipura = Linksys now.`
`20:05.41`	`watchy`	`i love chickens`
`20:05.44`	`*** part/#asterisk rizzo (n=rizzo@gentoo/developer/rizzo)`
`20:06.12`	`dos000`	`[TK]D-Fender, do you know which linksys part/serial number corresponds to sipura 3000 ? this is awsome`
`20:06.31`	`[TK]D-Fender`	`dos000 : Its not listed on their site. What do you want to know?`
`20:06.49`	`[TK]D-Fender`	`Its still only on www.sipura.com under the model number.`
`20:07.00`	`*** join/#asterisk Utah_Dave (n=boucha@0-1pool139-119.nas28.salt-lake-city1.ut.us.da.qwest.net)`
`20:07.03`	`*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`20:07.30`	`dos000`	`[TK]D-Fender, i just need to know if linksys already integrated the sipura voip to fxo rounting in their product. Most probably sipura will disapear soon.`
`20:07.40`	`_di`	`d000ds who can help me with muh asterisk ? it dont want make sip proxy for me :(`
`20:08.29`	`blitzrage`	`asterisk isn't a proxy...`
`20:08.34`	`zoa`	`its not sip proxy`
`20:08.40`	`blitzrage`	`its a b2bua`
`20:08.41`	`_di`	`without difference`
`20:08.42`	`zoa`	`i might be a little drunk, but i stil now that`
`20:08.49`	`_di`	`pbx`
`20:08.51`	`blitzrage`	`back to back user agent`
`20:08.56`	`zoa`	`cheers`
`20:08.57`	`*** join/#asterisk Chert (n=email@83.221.168.199)`
`20:09.05`	`dos000`	`asterisk will soon make its own genre !`
`20:09.18`	`_di`	`i just want to have connection between 2 soft phones`
`20:09.24`	`_di`	`via sip`
`20:09.31`	`blitzrage`	`~docs`
`20:09.32`	`jbot`	`docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html`
`20:10.00`	`*** part/#asterisk Chert (n=email@83.221.168.199)`
`20:10.04`	`wunderkin`	`~seen lrizzo`
`20:10.26`	`jbot`	`lrizzo <n=luigi@81-174-21-10.f5.ngi.it> was last seen on IRC in channel #asterisk, 4d 2h 18m 29s ago, saying: 'ping... any autoconf guru around here ?'.`
`20:10.26`	`*** join/#asterisk backblue (n=moo@87-196-9-100.net.novis.pt)`
`20:10.26`	`_di`	`blitzrage, i dont need docs`
`20:10.28`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`20:10.28`	`_di`	`i have it`
`20:10.41`	`dos000`	`_di, checkout http://asteriskathome.sourceforge.net/`
`20:10.42`	`blitzrage`	`ok, then what is your specific question?`
`20:10.57`	`_di`	`asterisk dont binds to port`
`20:11.07`	`backblue`	`ei ppl, anyone working with asterisk clustering?`
`20:11.08`	`_di`	`5060`
`20:11.21`	`blitzrage`	`backblue: yah, a bit`
`20:11.28`	`_di`	`but all configs are ok`
`20:11.37`	`_di`	`and asterisk is loading normally`
`20:11.53`	`backblue`	`blitzrage: how do you merge the info bettewen 2 asterisk servers?`
`20:11.59`	`[TK]D-Fender`	`dos000 : the only Sipura/Linksys with FXO is the SPA-3000`
`20:12.11`	`blitzrage`	`backblue: DUNDi`
`20:12.17`	`backblue`	`DUNDi?`
`20:12.20`	`backblue`	`?? dundi`
`20:12.21`	`blitzrage`	`www.dundi.com`
`20:12.30`	`justinu`	`how come no one ever talks about dundi in here?`
`20:12.30`	`blitzrage`	`lol!`
`20:12.36`	`blitzrage`	`because there is a #dundi`
`20:12.40`	`justinu`	`oh`
`20:12.42`	`blitzrage`	`and not many people use it`
`20:12.45`	`justinu`	`ic`
`20:12.54`	`dos000`	`_di, do yoirsefl a favor, instal *@home to a separate machine and try copying and pasting the config files which are accessible from a web interface`
`20:13.08`	`blitzrage`	`I just came up with a new topology which will basically allow me to use asterisk a faux sip proxy :)`
`20:13.16`	`blitzrage`	`as a*`
`20:13.29`	`justinu`	`why would you want to do that?`
`20:13.31`	`*** join/#asterisk fugitivo (n=ajf@201.255.177.172)`
`20:13.42`	`justinu`	`(out of curiosity)`
`20:13.44`	`dos000`	`[TK]D-Fender, thanks for the info`
`20:14.00`	`[TK]D-Fender`	`np`
`20:14.30`	`backblue`	`blitzrage: can you have 2 asterisks servers with the same userlist (realtime,or whatever) and call from one server, to other?`
`20:14.33`	`fugitivo`	`anyone is using MachineDetect() ?`
`20:14.35`	`zoa`	`hey ho Cresl1n`
`20:14.39`	`zoa`	`hey ho BladeRunner05`
`20:14.41`	`zoa`	`euh`
`20:14.43`	`zoa`	`blitzrage`
`20:15.14`	`Cresl1n`	`zoa!!!!`
`20:15.37`	`bhickey_`	`dos000 : Grandstream 488 will do the same and can be got a bit cheaper than the Sipura 3000`
`20:15.42`	`*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)`
`20:16.04`	`*** join/#asterisk santiago (n=santiago@208.195.215.97)`
`20:16.55`	`_di`	`dos000, i dont need @home ver. , the goal is connect asterisk and mvts`
`20:17.06`	`Nivex`	`bhickey_: interesting. I'd heard that the spa-3000 had some overheat problems. Any problems with your 488?`
`20:17.26`	`_di`	`but first of all i need to now how i can do simple things, and how i can solve simple problems`
`20:18.01`	`blitzrage`	`zoa: y0!`
`20:18.08`	`blitzrage`	`backblue: basically, this is what I do`
`20:18.15`	`*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)`
`20:18.23`	`*** join/#asterisk kink0 (n=k@62.37.205.161)`
`20:18.25`	`kink0`	`hi`
`20:19.01`	`*** join/#asterisk Renacor (n=kvirc@ip21.farheap.net)`
`20:19.06`	`[TK]D-Fender`	`blitzrage : got your Poly running yet? ;)`
`20:19.26`	`bhickey_`	`Nivex: I haven't come across anything disturbing yet`
`20:19.28`	`Renacor`	`anybody ever seen "Don't know what to do if second ROSE component is of type 0x6" ??`
`20:19.30`	`blitzrage`	`backblue: I have a script which pulls all my data from a postgreSQL database. This script then creates the flatfiles and reloads asterisk. This happens every 5 mins (I don't use realtime, but you could). Then, when a call comes in on one of the Asterisk servers, it asks the other boxes where the final destination is.`
`20:19.42`	`mog_work`	`yes Renacor`
`20:20.00`	`mog_work`	`it means that some piece of information is not supported by our pri stack`
`20:20.05`	`mog_work`	`it wont cause a dropped call`
`20:20.13`	`[av]bani`	`[TK]D-Fender: only the polycom 601 has the browser right? the 501 and 301 have no xml/xhtml/etc`
`20:20.14`	`mog_work`	`just means we are getting info we dont understand`
`20:20.23`	`[TK]D-Fender`	`[av]bani : correct`
`20:20.28`	`[TK]D-Fender`	`60x`
`20:20.48`	`*** part/#asterisk santiago (n=santiago@208.195.215.97)`
`20:20.49`	`[av]bani`	`even the $149 cisco 7905g has xml support :<`
`20:20.51`	`blitzrage`	`backblue: Then a box replies with a destination, and the call gets sent there. You could also mix in some GROUP_COUNT stuff to limit the number of calls on a box, so that if it has too many, then it doesn't reply to the DUNDi lookup`
`20:20.53`	`backblue`	`blitzrage: can we talk in private? i need real help.`
`20:21.09`	`blitzrage`	`backblue: sorry, thats all the info I can give you -- I'm developing it myself and can't really tell you any more :)`
`20:21.32`	`blitzrage`	`plus I have to go to the bank so I can get someone to pay me a nice big invoice :)`
`20:21.41`	`[TK]D-Fender`	`[av]bani : Yeah, but is a 6-line phone w/ possibly side caddy, supporting 802.3af and not a PITA to get firmware for? :)`
`20:21.47`	`blitzrage`	`[TK]D-Fender: no -- I lost the power adapter for my poly :( know where I can find one cheap?`
`20:22.02`	`[av]bani`	`well, the aastra 91xx supports xml also...`
`20:22.10`	`[av]bani`	`$135`
`20:22.16`	`*** join/#asterisk elvisthedj (n=Johnny@th20.montanavision.com)`
`20:22.17`	`blitzrage`	`Renacor: I've seen them -- you can safely ignore them. Nothing bad happened when I did on my PRIs`
`20:22.22`	`[TK]D-Fender`	`blitzrage : its a common adapter. I'm sure you could grab a multi-volt one and it'll work...`
`20:22.37`	`[av]bani`	`just disappointed that the 501 doesnt support it at least`
`20:22.44`	`blitzrage`	`[TK]D-Fender: yah... I looked for one at Radioshack/The Source and they were like 30+ dollars`
`20:22.44`	`[TK]D-Fender`	`[av]bani : Thats XML for the CONFIGS. the 91xx don't have full pixel screen....`
`20:22.45`	`backblue`	`blitzrage: i dont want you to tell me your implementation, i have a big problem that is "non-fixed users" i have 2 asterisk the user exists on both asterisk, but they have together to know in which asterisk is the user registered, because user CAN/SHOULD register on both, and make calls.`
`20:22.55`	`justinu`	`the aastra's are nice phones`
`20:22.58`	`justinu`	`i have a 480i`
`20:23.16`	`[av]bani`	`fender -> no, it explicitly has xml browser support... no graphics of course, but text...`
`20:23.20`	`[av]bani`	`http://www.o2m8.com/modules.php?name=News&file=article&sid=25`
`20:23.20`	`blitzrage`	`backblue: I'll give you a hint -- regexten and regcontext -- you're trying to do the exact same thing I am, but I don't want to give any more info :)`
`20:23.29`	`[TK]D-Fender`	`aasta = bleh. Last I checked you NEEDED a provisioning server and had to buy a power supply if you weren't using PoE.`
`20:23.38`	`justinu`	`blitz: how come?`
`20:23.39`	`*** join/#asterisk NDT (n=me@cpe-24-194-166-119.nycap.res.rr.com)`
`20:23.43`	`[av]bani`	`its limiting yeah,b ut better than nothing`
`20:24.04`	`justinu`	`fender: not true anymore about provisioning, but they don't come with any power supply, yes`
`20:24.07`	`backblue`	`blitzrage: ok, tks for the clues, i will have this working, in the next week, i hope, even if i have to code asterisk to do the clustering stuff all by himself.`
`20:24.15`	`justinu`	`aastra has a nice web interface now`
`20:24.16`	`tzanger`	`hmm... any of you guys have any suggestions for minimizing realtor fees/commissions on the sale of a house?`
`20:24.16`	`[av]bani`	`and it also support BLF... polycom still doesnt?`
`20:24.18`	`Renacor`	`blitzrage: thanks`
`20:24.21`	`tzanger`	`(totally OT but I do use * at home, heh)`
`20:24.27`	`justinu`	`heh`
`20:24.28`	`blitzrage`	`when a user registers to a box (could be any of them), regexten will place a NoOp() for that extension in a context. Use that context as the advertisment context in dundi.conf, and the box with the client registered will be the one that replies in your dundi network`
`20:24.36`	`justinu`	`tzanger: negotiate`
`20:24.37`	`[av]bani`	`i couldnt find anything about BLF in polycom other than "maybe in the future, but dont hold your breath"`
`20:24.42`	`*** join/#asterisk svenna_ (n=svenna@p548D17E9.dip0.t-ipconnect.de)`
`20:24.44`	`tzanger`	`justinu: well that's what I am planning on doing`
`20:24.47`	`bhickey_`	`Hello. Anyone using SIP -> GSM gateway devices? Using SIP ATA's with FXO ports & GSM gateways seems to take forever to connect (average 14-19 seconds)`
`20:24.54`	`svenna_`	`hi all`
`20:24.58`	`blitzrage`	`backblue: do yu program in C? If you do, I can help you if you can help me`
`20:25.11`	`justinu`	`tzanger: as far as any specific negotiations points, i'm not sure... maybe there's some message boards about it?`
`20:25.13`	`kink0`	`bhickey_, yes, that takes a time`
`20:25.22`	`*** join/#asterisk zippp (n=zip@63.99.9.2)`
`20:25.22`	`dos000`	`bhickey_, what gsm GW ?`
`20:25.23`	`blitzrage`	`I need a small change to the code, but I haven't had time to figure it out -- but it should be a simple change`
`20:25.30`	`[av]bani`	`the big pixel screen in the grandstream 2000 is just screaming for xml/xhtml support`
`20:25.33`	`[TK]D-Fender`	`[av]bani : BLF? As in for presence?`
`20:25.48`	`tzanger`	`justinu: nods I was heading there but thought I'd ask those here who may know`
`20:25.51`	`kink0`	`bhickey_, I am actually ussing sound card -> mobile phone , AT commands`
`20:25.56`	`bhickey_`	`dos300: Nokia 22 on both Sipura 3000 & Grandstream 488's so far`
`20:25.56`	`svenna_`	`how can i get the phonenumber, tht calls in? i tried to mess with CALLERID but id didnt work...`
`20:26.00`	`zippp`	`?nufone problems?`
`20:26.03`	`kink0`	`and gsm network takes a while to send the ring`
`20:26.17`	`[TK]D-Fender`	`[av]bani : Poly does presence already, and I use it myself...`
`20:26.19`	`justinu`	`tzanger: i was under the impression that most of these guys were paid in redbull, so i'm not sure why you ask here ;)`
`20:26.22`	`[av]bani`	`fender, busy lamp field, eg using eg the leds next to your speed dial buttons to show status of other extensions`
`20:26.38`	`tzanger`	`justinu: :-)`
`20:26.39`	`[TK]D-Fender`	`[av]bani : The do that already....`
`20:26.41`	`bhickey_`	`kink0: now that's just showing off ;)`
`20:26.45`	`[av]bani`	`i couldnt find BLF info in the polycom documentation`
`20:26.45`	`blitzrage`	`backblue: I sent you a PM -- get back to me when you have a chance`
`20:26.46`	`tzanger`	`someone else just have stutter audio to nufone/`
`20:26.47`	`tzanger`	`?`
`20:26.57`	`zippp`	`tzanger, yea`
`20:27.05`	`tzanger`	`zippp: cool so it does happen on occassion :-)`
`20:27.09`	`[TK]D-Fender`	`[av]bani : Funny I see it in all the SIP admin guides....`
`20:27.10`	`svenna_`	`hasnt anybody ab idea?`
`20:27.15`	`zippp`	`seems they say , "yes we are working on it" on the nufone login page`
`20:27.17`	`[av]bani`	`fender, page?`
`20:27.18`	`elvisthedj`	`Current bid.. 22 dollars for help upgrading firmware on cisco 7940 :D (paypal account req'd)`
`20:27.22`	`tzanger`	`zippp: hahah really`
`20:27.32`	`tzanger`	`where`
`20:27.36`	`kink0`	`bhickey_, but I think there no manner to avoid this time, since is about the same as when I dial manually from a mobile terminal phone`
`20:27.36`	`zippp`	`https://www.nufone.net/account/`
`20:27.42`	`tzanger`	`oh yeah that is not for that though`
`20:27.45`	`[av]bani`	`i did a search for BLF and acrobat said no joy`
`20:27.48`	`tzanger`	`that's just working on the new members interface`
`20:27.51`	`[av]bani`	`also busy lamp`
`20:27.52`	`tzanger`	`I've seen it, it is sweet`
`20:27.53`	`elvisthedj`	`iaxy is fried.. firmware on my 7940 is too old to connect to chan_sccp`
`20:28.09`	`kink0`	`bhickey_, how much channels are you ussing ?`
`20:28.27`	`[TK]D-Fender`	`[av]bani : SIP 1.5 Admin Guide, section 3.4.1`
`20:28.30`	`zippp`	`so you are having stutter problems also?`
`20:28.47`	`[TK]D-Fender`	`[av]bani : BLF is not the proper term in SIP. PRESENCE`
`20:28.52`	`bhickey_`	`kink0: the GSM Gateway setup seems to be contrinuting a minimum of 6-8 seconds extra. Just testin on 1-2 channels at the moment`
`20:29.02`	`[TK]D-Fender`	`And thatnk you thats a reason I bought these :)`
`20:29.15`	`tzanger`	`wow nufone's actually having connectivity issues`
`20:29.19`	`tzanger`	`I didn't think I'd ever see the day`
`20:29.23`	`zippp`	`thought so`
`20:29.27`	`zippp`	`terrible here`
`20:29.38`	`kink0`	`bhickey_, ok, I am testing with about 2 channels also, but I was seeking for some audio USB devices to add a lot of channels`
`20:29.46`	`[av]bani`	`fender, presence might be the techincal term, everyone from aastra to zyxel use BLF because its a term coming from analogue phones i guess`
`20:30.05`	`[av]bani`	`soits a term telco people are familiar with for 40 years`
`20:30.05`	`tzanger`	`bah and a codec error with asterlink`
`20:30.20`	`[TK]D-Fender`	`justinu : You can do XML like you showed for the 480i on the 60x as well...`
`20:30.43`	`justinu`	`right`
`20:30.59`	`[TK]D-Fender`	`[av]bani : Oh well! Aastra was in the analog game long before VoIP and I guess decided to hang onto it...`
`20:31.02`	`[av]bani`	`too bad the 501 cant :<`
`20:31.02`	`justinu`	`i'm not advocating aastra over polycom, i'm just saying aastra's are nice too`
`20:31.30`	`[TK]D-Fender`	`[av]bani : But look what you get for $250 out of an IP 601!`
`20:31.49`	`[av]bani`	`well, the 501 seems like not a good deal compared to the aastra 91xx's`
`20:32.07`	`[av]bani`	`which give you xml support, and a backlit screen...`
`20:32.13`	`[av]bani`	`for the same price`
`20:32.16`	`[TK]D-Fender`	`justinu : Yes in its way, but you lost the full graphics screen, etc on it and a number of other quirks`
`20:32.30`	`bhickey_`	`kink0: Current call setup times are just too long. I was thinking I might have to try a Digium FXO card next or go for a proper SIP -> GSM device like the Voiceblue think from 2n.cz`
`20:32.41`	`justinu`	`fender: true about graphics, but not sure about quirks`
`20:32.58`	`kink0`	`bhickey_, yes, just today I have ordered a Stargate from 2N too`
`20:33.02`	`[av]bani`	`7905g does graphics xml.. heh`
`20:33.06`	`ravenpi`	`So, what's the "one true SIP soft client" for 'doze?`
`20:33.17`	`justinu`	`eyebeam`
`20:33.23`	`kink0`	`but I still my develoment ussing cheap sound cards and free mobile terminals, that at the moment are working fine.`
`20:34.21`	`[TK]D-Fender`	`[av]bani : 91xx = ugly screen, no MGCP dial-plan, not sure on audio quality for comparison but I'd bet on Poly...`
`20:34.30`	`kink0`	`anywise times I think will be about the same, ussing sound cards and AT interface to mobile, Digium FXO to a docking, or Stargate`
`20:34.36`	`*** join/#asterisk FastJack (i=fastjack@p5091FE1E.dip.t-dialin.net)`
`20:34.42`	`bhickey_`	`kink0: That Stargate looks like some beast alright.`
`20:35.04`	`[av]bani`	`from what i can tell, aastra has great audio ... comes from being in the business for decades?`
`20:35.05`	`kink0`	`yes, I prefered 2N, even others are a bit cheap, like Valiant or so`
`20:35.36`	`kink0`	`I see the time to deliver a call is there always due to the GSM network routing process`
`20:36.01`	`[av]bani`	`tis a pity the spa-941 doesnt do xml. would be a killer deal if it did`
`20:36.17`	`[av]bani`	`it seems linksys does not want to sell the spa-941 though.`
`20:36.23`	`bhickey_`	`kink0: Thanks, I'll try the FXO card route next anyway and see what effect that has on call setup times.`
`20:36.23`	`kink0`	`bhickey_, where are you from ? ( I think GSM gateways are not very ussual in USA )`
`20:36.33`	`[TK]D-Fender`	`[av]bani : Yeah the SPA-941 could do a fair bit better I find. I might rather have chosen an IP501 for home instead...`
`20:36.38`	`fugitivo`	`kink0: are you using 2n gsm gateways?`
`20:36.50`	`kink0`	`fugitivo, I just ordered, not arrives yet.`
`20:36.57`	`fugitivo`	`sip gsm gateway?`
`20:37.06`	`[TK]D-Fender`	`[av]bani : Still a nice phone, but the 20$ difference makes it hard`
`20:37.06`	`[av]bani`	`fender, seen the linksys phm1200 ?`
`20:37.11`	`*** join/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com)`
`20:37.18`	`[TK]D-Fender`	`[av]bani : Yeah... PROPRIETARY. ick`
`20:37.18`	`kink0`	`fugitivo, no, I ordered a PRI interface, since I have a digium PRI card yet.`
`20:37.21`	`[av]bani`	`?`
`20:37.24`	`zoa`	`the best looking phone for the moment is the thomson`
`20:37.28`	`[av]bani`	`proprietary?`
`20:37.33`	`SimonR`	`has anyone seen problems with zombie AGI processes?`
`20:37.33`	`kink0`	`fugitivo, and I pretend to use Asterisk begin the 2N`
`20:37.34`	`[TK]D-Fender`	`PHM isn't straigh SIP.`
`20:37.42`	`fugitivo`	`kink0: how much is that one? i know the sip gsm gateway is like 900 euros`
`20:37.43`	`[av]bani`	`o_O`
`20:37.51`	`[TK]D-Fender`	`its made ONLY for their specialized boxes`
`20:37.53`	`[av]bani`	`Operation of a PHM1200 IP phone requires the installation, configuration and operation of one Linksys One services router (such as the SVR3000) at the same site.`
`20:37.57`	`[av]bani`	`ahaha.. thats super.`
`20:38.00`	`[av]bani`	`arseholes`
`20:38.04`	`*** join/#asterisk CleanerX1idle (n=jens@nat-ph3-wh.rz.uni-karlsruhe.de)`
`20:38.08`	`[TK]D-Fender`	`I prefer the term "craptastic".`
`20:38.13`	`[av]bani`	`they are sooo screwing themselves on that`
`20:38.31`	`kink0`	`fugitivo, a complete 32 channels , including voIP card, antenna spliter, two external yagi anntenas is about 16000 Euro ( 18000 USD )`
`20:38.39`	`[av]bani`	`if its a nice enough phone i'm sure someone will RE it like they did with cisco's skinny protocol`
`20:38.44`	`[TK]D-Fender`	`[av]bani : I doubt that. They're just losing the hobbyists and gaining the turnkey people.`
`20:38.52`	`zoa`	`i can make it cheaper`
`20:38.54`	`fugitivo`	`kink0: wow`
`20:39.00`	`bhickey_`	`fugitivo: what model is that for 900 euro and where are you getting that price?`
`20:39.02`	`[av]bani`	`svr3000 doesnt strike me as exactly turnkey`
`20:39.09`	`kink0`	`fugitivo, well, the same from Teles is 22,000 Euro`
`20:39.15`	`zoa`	`i can build 32 channels for a lot less then 16000 euro`
`20:39.20`	`zoa`	`not buy, build`
`20:39.20`	`kink0`	`and the "same" from Valiant is about 12000 Euro`
`20:39.27`	`fugitivo`	`bhickey_: the regular sip gsm gateway from 2n, i met them here in argentina at expocomm, they give us that price`
`20:39.45`	`fugitivo`	`gave`
`20:39.48`	`[TK]D-Fender`	`[av]bani : Linksys is basically making a cheaper CallManager setup now. Unless its compliant I wouldn't touch it personally.`
`20:39.59`	`kink0`	`fugitivo, did you meet Mr. Michael from 2N at Argentina about two months ago ?`
`20:40.29`	`fugitivo`	`kink0: i don't remember his name... let me see if i find his card`
`20:40.53`	`[av]bani`	`well they could do ok if it doesnt suck, i mean look at mitel and their proprietary stuff. they do ok i guess`
`20:40.56`	`bhickey_`	`fugitivo : That's less than half the trade price they quoted me for a 2-channel Voiceblue Lite ???`
`20:41.24`	`fugitivo`	`kink0: Jan Matejcek and Michal Kratochvil (weird names)`
`20:41.25`	`kink0`	`bhickey_, voiceblue is a very little to do any serious traffic`
`20:41.37`	`kink0`	`yes , Michal`
`20:41.49`	`kink0`	`or Michael in english :)`
`20:42.03`	`bhickey_`	`kink0: yes I know but 4 channels will be enough to prove the concept for now`
`20:42.38`	`[av]bani`	`if grandstream puts some kind of browser support into the gxp that would be sooo awesome`
`20:42.38`	`kink0`	`bhickey_, all is depending what SIM plans you buy, or if you need to swap perididally SIMs`
`20:42.39`	`fugitivo`	`bhickey_: i think it was a single or dual channel, can't remember`
`20:42.45`	`[av]bani`	`the lcd is largely a gimmick otherwise`
`20:43.08`	`fugitivo`	`kink0: they travel around the globe, lucky guys :)`
`20:43.16`	`kink0`	`bhickey_, think this... before to buy a Blue: A 2N basic Stargate with 2 or 4 channels will cost about 2500`
`20:43.40`	`kink0`	`fugitivo, yes, when he come back from Argentina goes to SIMO at Madrid, fews days later.`
`20:44.19`	`fugitivo`	`are you in madrid/`
`20:44.38`	`[TK]D-Fender`	`[av]bani : Yeah, the GXP with some tweaking and another 50$ worth of better materials would be a serious phone.`
`20:44.43`	`kink0`	`no, I am at south Spain, not exactly Madrid now.`
`20:44.56`	`[TK]D-Fender`	`The problems on it are pretty wild though. (range)`
`20:45.08`	`bhickey_`	`kink0: 4 channels will help bring a 10k bill down by about 4.5k or so`
`20:45.26`	`fugitivo`	`that's a LOT`
`20:45.29`	`*** join/#asterisk CleanerX_idle (n=jens@nat-ph3-wh.rz.uni-karlsruhe.de)`
`20:46.01`	`*** join/#asterisk l2trace (n=_l2trace@m015f36d0.tmodns.net)`
`20:46.02`	`kink0`	`bhickey_, yes, if you use i.e. a Digium PRI and a basic Stargate, you are near the same price as ussing Blue, but you have a high scalability`
`20:46.20`	`bhickey_`	`fugitivo: yes it's replacing landline -> mobile calls which cost a lot here with mobile -> mobile calls on the same network`
`20:46.32`	`*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)`
`20:46.40`	`kink0`	`bhickey_, where are you from ?`
`20:47.15`	`bhickey_`	`kink0: thanks I must have another look at all the 2n product line. It's been a while since I checked them out.`
`20:47.28`	`bhickey_`	`kink0 : IN ireland`
`20:47.30`	`fugitivo`	`bhickey_: here we have corporate mobile plans, all calls between company's mobiles are free, imagine that`
`20:47.57`	`*** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it)`
`20:48.04`	`l2trace`	`is anyone using a fax behind a sipura 2002 ?`
`20:48.33`	`kink0`	`fugitivo, yes, here also are free mobile calls inside a group, but that is not very valid to sell minutes`
`20:48.33`	`*** join/#asterisk jets (n=jetsnoc@meowwwww.pmt.coop)`
`20:48.37`	`bhickey_`	`fugitivo : same here but you've got to pay extra per month per mobile for the privilege. When you've got 50 phones that adds up to about 1000 euro amount for all thise "free" calls`
`20:48.53`	`kink0`	`and we at the company no are many people spoken at mobile phones between us !!`
`20:49.20`	`*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)`
`20:49.32`	`*** part/#asterisk _di (n=disnider@papados.sferos.com)`
`20:49.34`	`chiardon`	`zt spanconfig failed on span1 no such device or address (6) . . .whats the problem`
`20:49.52`	`chiardon`	`txns`
`20:50.27`	`*** join/#asterisk santiago (n=santiago@208.195.215.97)`
`20:51.17`	`kink0`	`I am planning to connect a lot of audio Alsa-OSS interfaces to the PC, but last try ussing Creative did not susscess`
`20:52.02`	`*** join/#asterisk evilrabbi (i=evilrabb@71.144.78.244)`
`20:52.16`	`Katty`	`anyone framilier with the windows registry?`
`20:52.29`	`evilrabbi`	`where would I look for information on routing calls by what interface the call was received on?`
`20:52.30`	`chiardon`	`anybody?`
`20:52.54`	`Katty`	`[TK]D-Fender: mew?`
`20:53.43`	`*** join/#asterisk rm (n=rm@66.193.229.254)`
`20:53.57`	`[TK]D-Fender`	`Katty: mew.`
`20:54.14`	`rm`	`does anyone know if asterisk supports TDMoIP?`
`20:54.16`	`Katty`	`[TK]D-Fender: are you any good at windowsy snooping?`
`20:54.35`	`*** join/#asterisk Jzalae (n=sk@bb-205-209-93-139.gwi.net)`
`20:54.36`	`[TK]D-Fender`	`Katty: marginally (and a small one at that). Ask away....`
`20:54.42`	`detatch`	`yes it does`
`20:54.43`	`rm`	`i see stuff about TDMoE but not IP`
`20:54.51`	`detatch`	`er`
`20:54.51`	`detatch`	`yea`
`20:54.52`	`rm`	`detatch: can you point me to more info?`
`20:54.53`	`detatch`	`tdmoe`
`20:54.58`	`rm`	`but not ip?`
`20:55.00`	`detatch`	`dunno about overip`
`20:55.03`	`generalhan`	`When i try to compile asterisk 1.2.1 (after compiling libpri and zaptel) im getting this error :: collect2: ld returned 1 exit status make: *** [asterisk] Error 1 :: anyone know whats going on ?`
`20:55.07`	`rm`	`ok`
`20:55.11`	`Katty`	`[TK]D-Fender: i'm writing an auditing batch script. copying the index.dat file...along with reg query information about typedurls and recently searched keywords, etc.`
`20:55.18`	`Katty`	`[TK]D-Fender: have any more neat things to make a note of?`
`20:55.18`	`rm`	`detatch: thanks`
`20:55.21`	`fugitivo`	`generalhan: that's not the error`
`20:55.27`	`detatch`	`and this is not pastebin`
`20:55.47`	`[TK]D-Fender`	`Katty: Sorry, out of my league there...`
`20:55.55`	`generalhan`	`fugitivo: what do you mean thats not the error ?`
`20:55.57`	`Katty`	`[TK]D-Fender: k'then (=`
`20:56.16`	`fugitivo`	`generalhan: that's the message that there's an error, not the error itself`
`20:56.39`	`*** join/#asterisk javar (n=javar@Dynamic-IP-cr20011863171.cable.net.co)`
`20:56.42`	`generalhan`	`i see ... and where is the actual error ?`
`20:56.56`	`generalhan`	`cause i need to get this working !`
`20:57.00`	`fugitivo`	`before that`
`20:57.05`	`generalhan`	`ohh`
`20:57.16`	`fugitivo`	`pastebin`
`20:57.21`	`generalhan`	`check`
`20:58.43`	`generalhan`	`fugitivo: do you have a second to look at the error im getting? im sure its something simple and im just being retarded: http://generalhan.pastebin.ca/36621`
`20:59.16`	`*** join/#asterisk _mistral (i=mistral@jstevenson.plus.com)`
`20:59.25`	`fugitivo`	`<PROTECTED>`
`20:59.30`	`fugitivo`	`that;s the error`
`20:59.48`	`generalhan`	`lol i saw that ! once you said it was before it, i just dont know what exactly lssl is let a lone where to find it`
`20:59.51`	`zoa`	`apt-get install libssl-dev`
`21:00.21`	`fugitivo`	`ssl libs and headers`
`21:00.23`	`fugitivo`	`you need that`
`21:01.36`	`*** part/#asterisk bhickey_ (n=chatzill@212.2.174.21)`
`21:01.52`	`generalhan`	`thanks guys`
`21:02.08`	`generalhan`	`what is the apt-get, the only one that i have used is yum`
`21:02.17`	`fugitivo`	`use yum`
`21:02.20`	`fugitivo`	`apt-get is for debian`
`21:02.24`	`*** part/#asterisk jets (n=jetsnoc@meowwwww.pmt.coop)`
`21:02.40`	`generalhan`	`i did yum ssl libssl-dev ssl-dev. what is it ?`
`21:02.55`	`fugitivo`	`i think libssl-dev will be enough`
`21:03.33`	`generalhan`	`fugitivo: it says no match for argument: libssl-dev`
`21:04.20`	`fugitivo`	`generalhan: i don't know yum, maybe you need to do "yum install packagename" ?`
`21:04.29`	`*** join/#asterisk r0d3nt\|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)`
`21:04.34`	`fugitivo`	`yum search package ?`
`21:04.43`	`generalhan`	`i did "yum install libssl-dev" and it came back with "no match"`
`21:04.54`	`fugitivo`	`do you have any way to search the package name?`
`21:05.11`	`generalhan`	`yum search`
`21:05.15`	`*** join/#asterisk Seldon19751 (n=someone@199.243.101.131)`
`21:05.23`	`fugitivo`	`yum search ssl`
`21:05.24`	`mog_work`	`hey anoyone know how to add files to bootable iso`
`21:05.27`	`mog_work`	`and reburn it?`
`21:05.50`	`Seldon19751`	`Hello; can someone tell me what I need to do with Asterisk to enable users to enter Speed Dial entries from their Polycom 501s`
`21:06.02`	`fugitivo`	`mog_work: mounting it?`
`21:06.17`	`mog_work`	`can you remount rw and just write and reburn?`
`21:06.24`	`fugitivo`	`no idea :)`
`21:06.29`	`mog_work`	`lol`
`21:06.37`	`fugitivo`	`maybe`
`21:06.39`	`fugitivo`	`try it`
`21:07.14`	`*** join/#asterisk Dark_ (n=t7DS@200.206.141.40)`
`21:07.18`	`fugitivo`	`is any way to specify the filename for automon?`
`21:07.19`	`*** part/#asterisk Dark_ (n=t7DS@200.206.141.40)`
`21:08.24`	`dos000`	`mog_work, mount -o loop -t iso9660 filename.iso /mnt/iso`
`21:08.28`	`[TK]D-Fender`	`Seldon1975 : Speed dial entries on Polycom phones have nothing to do with *`
`21:08.57`	`Seldon19751`	`D-Fender: oh`
`21:08.59`	`[TK]D-Fender`	`Seldon1975 : A phone dials what a phone wants to dial.`
`21:09.14`	`chiardon`	`<PROTECTED>`
`21:09.28`	`generalhan`	`fugitivo: thanks for the help ... it turned out to be "yum install openssl-devel"`
`21:09.37`	`Seldon19751`	`D-Fender: when users try to enter speed dials on their PC501s the phone flashes up a message 'Busy! Try Again Later' and does nothing`
`21:09.37`	`fugitivo`	`generalhan: ok :)`
`21:09.42`	`*** join/#asterisk ManxPower (i=ewieling@210.sub-70-197-97.myvzw.com)`
`21:10.00`	`Seldon19751`	`Anyone here using Polycom 501s?`
`21:10.22`	`Katty`	`i'm using 500s`
`21:10.43`	`Katty`	`what seems to be your major malfuntion, Seldon19751?`
`21:10.44`	`rue_mohr`	`Jan 12 13:09:22 WARNING[19687]: chan_iax2.c:7487 socket_read: Received mini frame before first full voice frame`
`21:10.48`	`[TK]D-Fender`	`Seldon1975 : Sounds like the dial-plan on your phone isn't what you want it to be then.`
`21:10.52`	`Seldon19751`	`when users try to enter speed dials on their PC501s the phone flashes up a message 'Busy! Try Again Later' and does nothing`
`21:10.57`	`rue_mohr`	`could that cause missing audio data?`
`21:11.08`	`Seldon19751`	`Katty: have you entered Speed Dials into your Contact List?`
`21:11.14`	`*** join/#asterisk krustyclown (n=hmm@202.153.246.14)`
`21:11.15`	`Katty`	`Seldon19751: yes.`
`21:11.39`	`Seldon19751`	`D-Fender: hang on - dialplan? I thought you said it had nothing to do with Asterisk`
`21:11.49`	`[TK]D-Fender`	`Seldon1975 : The Polycom'`
`21:11.51`	`*** join/#asterisk redman (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)`
`21:11.55`	`[TK]D-Fender`	`s MGCP dial-plan.`
`21:11.57`	`[TK]D-Fender`	`not *`
`21:12.03`	`Seldon19751`	`D-Fender oooh`
`21:12.08`	`Seldon19751`	`D-Fender: thanks`
`21:12.33`	`[TK]D-Fender`	`Seldon1975 : What exactly are you trying to do? Who's speed-dial? A contact in your IP501 directory? Or an * extensions.conf speedial?`
`21:13.31`	`Seldon19751`	`D-Fender, Katty: well (speed dials aside for the moment) I get the error when I go to Menu > Features > Contact Directory > Add`
`21:13.33`	`Katty`	`Seldon19751: can i see your speed dial thingy in extenions?`
`21:13.33`	`mog_work`	`that wont mount it rw`
`21:14.00`	`Seldon19751`	`Katty: extensions.conf?`
`21:14.02`	`Katty`	`Seldon19751: what's the cli say, exactly, when you dial..pastebin?`
`21:14.36`	`Seldon19751`	`'Busy! Please Try Again..'`
`21:14.41`	`Katty`	`Seldon19751: i need the whole thing (=`
`21:15.23`	`[TK]D-Fender`	`Seldon1975 : WHATS THE ERROR?`
`21:15.26`	`Seldon19751`	`Katty: Sorry for being dense; I'm not really sure what you're after`
`21:15.31`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`21:15.40`	`Katty`	`Seldon19751: i want to see the exact steps asterisk is taking when it dials.`
`21:15.49`	`Seldon19751`	`I'm not dialling yet`
`21:15.56`	`Seldon19751`	`im just on the Polycom Handset`
`21:15.56`	`[TK]D-Fender`	`Katty : I'm thinking the problem is all polycom setup related...`
`21:16.00`	`Katty`	`oh.`
`21:16.12`	`Seldon19751`	`On the handset itself I'm trying to add a Contact`
`21:16.13`	`[TK]D-Fender`	`Seldon1975 : SO you go to add a contact, THEN what?`
`21:16.33`	`Seldon19751`	`When I press 'Add' it brings up a message saying 'Busy! Please Try Again..'`
`21:16.52`	`Seldon19751`	`and goes back to the Directory listing (which is empty)`
`21:16.56`	`[TK]D-Fender`	`Seldon1975 : What SIP/ Bootrom you running on it?`
`21:17.15`	`[TK]D-Fender`	`Seldon1975 : And your problem definately has nothing to do with *`
`21:17.23`	`Seldon19751`	`ok`
`21:17.43`	`*** join/#asterisk bkw__ (n=bkw_@adsl-70-142-59-48.dsl.tul2ok.sbcglobal.net)`
`21:17.45`	`[TK]D-Fender`	`I'd bet on "out of memory" or a provisioning problem.`
`21:17.55`	`Katty`	`Seldon19751: http://lists.digium.com/pipermail/asterisk-users/2005-September/123907.html`
`21:18.10`	`Seldon19751`	`BootBlock: 2.5.0; Bootrom 2.6.2.0032; SIP.ld Version: 1.5.2.0054`
`21:18.40`	`[TK]D-Fender`	`Seldon1975 : You provisioning the phones?`
`21:18.50`	`Seldon19751`	`D-Fender: yes`
`21:18.54`	`Seldon19751`	`D-Fender: using TFTP`
`21:19.24`	`Seldon19751`	`Katty: thanks - that looks relevent`
`21:19.32`	`Seldon19751`	`Katty: I'll read the whole thing`
`21:20.30`	`*** join/#asterisk delox99 (n=delox@modemcable246.108-203-24.mc.videotron.ca)`
`21:21.06`	`delox99`	`hi all`
`21:21.28`	`Seldon19751`	`So it looks like I have to roll back the PC501's firmware`
`21:21.29`	`*** join/#asterisk r0d3nt_m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)`
`21:21.32`	`delox99`	`anyone knows whats an aproximate ratio on calling cards usage`
`21:21.34`	`Seldon19751`	`that hurts`
`21:22.00`	`delox99`	`like if i sell 1800 cards, how many phone lines will i be using?`
`21:22.01`	`*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)`
`21:22.42`	`Seldon19751`	`D-Fender, Katty: if I roll back the sip.ld to Sip 1.4.1.0040, I guess I have to simultaneously roll back the Bootrom?`
`21:22.50`	`[TK]D-Fender`	`Seldon1975 : Roll forward to 1.6.2`
`21:22.58`	`[TK]D-Fender`	`you don't want to go backwards...`
`21:23.01`	`Katty`	`Seldon19751: what happens if you disconnect the phone from the server?`
`21:23.08`	`Katty`	`Seldon19751: and then try to add the contact?`
`21:23.23`	`Katty`	`Seldon19751: does it still screw up?`
`21:25.52`	`Seldon19751`	`ill try it`
`21:26.39`	`brad_mssw`	`any recommendations for sip or iax providers, other than teliax or junction networks ?`
`21:26.55`	`fugitivo`	`voicepulse`
`21:27.21`	`*** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com)`
`21:27.30`	`brad_mssw`	`fugitivo: any experience with them ?`
`21:27.43`	`fugitivo`	`minimal`
`21:27.51`	`fugitivo`	`it works`
`21:27.56`	`Seldon19751`	`D-Fender, Katty: also is there a specific way to set up speed dials on Asterisk too? I searched for sdpeed dial at voip-info but all I got was http://www.voip-info.org/wiki/index.php?page=PBX+Speed+Dial`
`21:28.22`	`brad_mssw`	`fugitivo: have a proxy address I can do a traceroute on for them by any chance ?`
`21:28.42`	`fugitivo`	`hold on`
`21:28.43`	`Katty`	`Seldon19751: sure, but it's a global speed dial thing.`
`21:28.44`	`[TK]D-Fender`	`Seldon1975 : Its basic dial-plan stuff... c'mon...`
`21:28.51`	`Katty`	`Seldon19751: like anyone can dial 5490 and it will dial a number.`
`21:29.03`	`Katty`	`Seldon19751: works just like an extension.`
`21:29.08`	`Seldon19751`	`right`
`21:29.11`	`Seldon19751`	`I was just checking`
`21:29.15`	`Katty`	`(=`
`21:29.24`	`[TK]D-Fender`	`exten => 11,1,Dial(ZAP/G1/areallylongnumber,20) <- theres a speed-dial...`
`21:29.38`	`[TK]D-Fender`	`11 = a reallylongnumber!`
`21:29.41`	`Seldon19751`	`also, I tried adding a contact when the phone was off the network and it still fails`
`21:30.02`	`Seldon19751`	`ok D-Fender thanks - I was just wondering if there was any specific functionality`
`21:30.05`	`fugitivo`	`brad_mssw: gw5.voicepulse.com (sip) gwiax01.voicepulse.com gwiax02.voicepulse.com (iax)`
`21:30.21`	`brad_mssw`	`fugitivo: thanks`
`21:30.50`	`[TK]D-Fender`	`ok, off home, later all`
`21:31.09`	`Katty`	`byebye fender.`
`21:31.09`	`brad_mssw`	`fugitivo: iax gateways don't resolve`
`21:31.52`	`fugitivo`	`brad_mssw: my mistake`
`21:32.06`	`fugitivo`	`brad_mssw: gwiaxt01.voicepulse.com and gwiaxt02.voicepulse.com`
`21:32.12`	`brad_mssw`	`ahh, thanks`
`21:32.40`	`[av]bani`	`http://bani.anime.net/phonez/`
`21:33.24`	`RoyK`	`zoa: PING`
`21:33.53`	`rue_mohr`	`gee, we have 10 digits here to our neighbour`
`21:34.08`	`rue_mohr`	`11 if its long distance to them`
`21:34.21`	`RoyK`	`8 digits to all of norway.......`
`21:34.24`	`*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)`
`21:35.01`	`fugitivo`	`8 here too`
`21:35.08`	`RoyK`	`fugitivo: where?`
`21:35.12`	`fugitivo`	`Argentina`
`21:35.26`	`RoyK`	`i thought you had loads of people there....`
`21:35.51`	`fugitivo`	`well, not 8 for all of Argentina :)`
`21:36.06`	`fugitivo`	`if you call long distance it's 3 more`
`21:36.10`	`rue_mohr`	`we used to be 7 for local calls, I dont think telus tried hard when splitting area codes`
`21:36.42`	`RoyK`	`in .no it's 8`
`21:36.48`	`RoyK`	`anywhere....`
`21:37.22`	`fugitivo`	`how do you know when it's a long distance call?`
`21:37.55`	`RoyK`	`zoa: ?`
`21:38.08`	`*** join/#asterisk _zigo__ (n=ogiz@m14.net81-64-48.noos.fr)`
`21:41.45`	`*** join/#asterisk JunK-Y_ (n=junky@69.156.216.128)`
`21:43.35`	`*** join/#asterisk george435 (n=tyu@194.154.204.68.cfl.res.rr.com)`
`21:43.45`	`docelm0`	`FLORIDA!`
`21:43.46`	`l2trace`	`can anyone offer me some advice on getting asterisk to do realtime from a mysql db ?`
`21:43.53`	`docelm0`	`Check the wiki`
`21:43.57`	`docelm0`	`take to long to explain`
`21:43.57`	`l2trace`	`i have`
`21:44.00`	`Seldon19751`	`D-Fender, Katty: sorry just one more small question; can the PC501's, PC601s and PC301s use the same sip.ld?`
`21:44.06`	`docelm0`	`what kinda realtime?`
`21:44.09`	`l2trace`	`it connects`
`21:44.17`	`docelm0`	`SIP/IAX/Voicemail/Dialplan?`
`21:44.32`	`l2trace`	`sippeers is not querying the table`
`21:44.43`	`docelm0`	`have you turned on Debug?`
`21:44.43`	`fugitivo`	`l2trace: my advice is not to use mysql`
`21:44.46`	`fugitivo`	`it's evil`
`21:44.46`	`l2trace`	`yes`
`21:44.51`	`l2trace`	`really`
`21:44.51`	`docelm0`	`What does it say?`
`21:44.57`	`docelm0`	`is there an issue with the query..`
`21:45.02`	`docelm0`	`PostGres blows..`
`21:45.03`	`l2trace`	`there is no query`
`21:45.12`	`l2trace`	`there is a connection`
`21:45.13`	`rue_mohr`	`WHAT!`
`21:45.14`	`l2trace`	`on a reload`
`21:45.15`	`docelm0`	`What version are you using of asterisk?`
`21:45.16`	`rue_mohr`	`does not!`
`21:45.33`	`fugitivo`	`PosgreSQL rocks`
`21:45.39`	`docelm0`	`blows..`
`21:45.42`	`fugitivo`	`rocks`
`21:45.45`	`docelm0`	`I will stick with MySQL 5`
`21:45.46`	`l2trace`	`Ssterisk SVN-trunk-r7230 built by root`
`21:45.47`	`rob0`	`boys!!`
`21:45.48`	`l2trace`	`Asterisk SVN-trunk-r7230 built by root`
`21:45.52`	`fugitivo`	`blows your mysql 5`
`21:45.58`	`docelm0`	`ohh baby`
`21:46.02`	`docelm0`	`Is that head?`
`21:46.04`	`docelm0`	`or 1.2?`
`21:46.16`	`docelm0`	`If its head it may be broke.. I am using 1.2`
`21:46.25`	`docelm0`	`Check bugs.digium.com for information`
`21:46.30`	`l2trace`	`ew`
`21:46.38`	`l2trace`	`that may be`
`21:47.49`	`l2trace`	`thanks`
`21:49.54`	`george435`	`What are the sipura 3000 pstn(tab) settings to make the Systm episode with Todd Long work. I can get the line 1 to work through , but cannot get the telco line to answer with anything execept a fast busy?`
`21:53.17`	`george435`	`Sorry, that should have been the Systm episode with John Todd`
`21:55.11`	`evilrabbi`	`Position 6 - Spanning-Tree Protocol Posture`
`21:55.17`	`evilrabbi`	`http://www.collegesexadvice.com/sex.shtml`
`21:58.40`	`*** join/#asterisk copantl (n=galel@63.245.93.138)`
`21:59.00`	`copantl`	`help`
`21:59.58`	`copantl`	`line 49: Unable to open master device '/dev/zap/ctl' zaptel.`
`21:59.58`	`rue_mohr`	`heh, I said the same thing`
`22:00.04`	`*** join/#asterisk derka (n=derka@crn93-1-82-237-178-115.fbx.proxad.net)`
`22:00.45`	`l2trace`	`nice everyone at borders likes it`
`22:01.52`	`copantl`	`i got this erron when i tried to run ztcfg -vv`
`22:02.06`	`copantl`	`line 49: Unable to open master device '/dev/zap/ctl'`
`22:02.07`	`copantl`	`zaptel.`
`22:02.15`	`rue_mohr`	`are you root when you run it?`
`22:02.19`	`copantl`	`yes`
`22:02.34`	`rue_mohr`	`if you do ls /dev/zap/* what do you get?`
`22:02.41`	`evilrabbi`	`is an agi required for me to route traffic based on what channel it came in on?`
`22:02.53`	`rue_mohr`	`whats an agi?`
`22:03.07`	`evilrabbi`	`a script for asterisk`
`22:03.17`	`rue_mohr`	`ah`
`22:03.19`	`copantl`	`a lot of /dev/zap/1............190`
`22:03.22`	`rue_mohr`	`then I think the answer is no`
`22:03.31`	`rue_mohr`	`copantl good, is ther a ctrl?`
`22:03.36`	`rue_mohr`	`er ctl?`
`22:03.37`	`evilrabbi`	`agi means asterisk gateway interface`
`22:03.48`	`copantl`	`/dev/zap/ctl yes`
`22:04.04`	`copantl`	`sorry`
`22:04.05`	`copantl`	`no`
`22:04.12`	`copantl`	`ctrl no`
`22:04.14`	`rue_mohr`	`copantl try cat /dev/zap/ctl see if it gives you an error`
`22:04.20`	`copantl`	`ok`
`22:04.42`	`*** join/#asterisk Burgwork (n=corey@S010600131016cf6f.gv.shawcable.net)`
`22:04.59`	`copantl`	`cat: /dev/zap/ctl: No such device or address`
`22:05.13`	`copantl`	`i got a te110p`
`22:05.15`	`Burgwork`	`can I make an asterisk<-->asterisk call ring a different ring tone to an inbound call?`
`22:06.20`	`copantl`	`question.... if i upgrade my kernel i have to upgrade my asterisk or only the zaptel?`
`22:06.30`	`fugitivo`	`* glibc detected * free(): invalid pointer: 0x00007fffff932945 ***`
`22:06.41`	`tzanger`	`fugitivo: nice`
`22:06.46`	`evilrabbi`	`haha`
`22:06.47`	`fugitivo`	`:\`
`22:07.55`	`copantl`	`rue_mohr: cat: /dev/zap/ctl: No such device or address`
`22:08.26`	`rue_mohr`	`copantl there's your problem driver`
`22:08.30`	`rue_mohr`	`copantl lsmod ?`
`22:08.37`	`copantl`	`nothing`
`22:09.37`	`ManxPower`	`copantl, if you upgrade your kernel you must REINSTALL zaptel`
`22:09.39`	`l2trace`	`with realtime config will sip show peers execute a select on the sippeers table without anyone registered ?`
`22:09.44`	`*** join/#asterisk Iva1 (n=ivan@host234-246.pool8254.interbusiness.it)`
`22:09.46`	`copantl`	`ok`
`22:09.54`	`copantl`	`i did it`
`22:10.08`	`*** part/#asterisk Iva1 (n=ivan@host234-246.pool8254.interbusiness.it)`
`22:11.26`	`rue_mohr`	`copantl did you need to insert your module?`
`22:11.36`	`copantl`	`i did modprobe wcte11xp`
`22:12.05`	`copantl`	`FATAL: Module wcte11xp not found.`
`22:12.34`	`rue_mohr`	`modprobe foo`
`22:12.38`	`rue_mohr`	`not insmod`
`22:13.12`	`copantl`	`right i did that`
`22:13.28`	`*** join/#asterisk zotz (n=zotz@24.231.47.175)`
`22:15.13`	`copantl`	`what can be wrong?`
`22:15.53`	`copantl`	`i use the debian packages for zaptel and asterisk`
`22:16.31`	`*** join/#asterisk badboyz (n=bbz@adsl-70-128-78-22.dsl.stlsmo.swbell.net)`
`22:18.48`	`*** join/#asterisk razu_ (n=razu@213-35-173-39-dsl.prn.estpak.ee)`
`22:19.52`	`rkioko`	`hi`
`22:19.55`	`rkioko`	`need advice`
`22:19.55`	`rue_mohr`	`also look in /lib/modules/{kernel-version}/* for your driver`
`22:20.07`	`rkioko`	`need to terminate to pstn`
`22:20.14`	`rue_mohr`	`sorry, I'm really just here to ask * questions`
`22:20.23`	`rkioko`	`and considering sangoma + asterisk`
`22:20.31`	`rkioko`	`or as5300`
`22:20.55`	`rkioko`	`we need to around 100 concurrent calls`
`22:21.07`	`rkioko`	`to pstn`
`22:21.09`	`*** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)`
`22:21.09`	`rue_mohr`	`remember that a T1 only carries 24`
`22:21.15`	`rkioko`	`will asterisk support that`
`22:21.17`	`rkioko`	`on E1`
`22:21.44`	`rkioko`	`or should i go with the as5300`
`22:21.48`	`rue_mohr`	`no idea, we onyl have enough T1's going in to generate 48 concurrent calls`
`22:21.54`	`rkioko`	`reliability is a key issue`
`22:22.16`	`rue_mohr`	`know what your doing, build carefully, and maintain`
`22:22.28`	`rue_mohr`	`unlike us who implemented it in 2 days`
`22:22.40`	`rue_mohr`	`and have had no time to maintiain it`
`22:22.45`	`rue_mohr`	`or test it properly`
`22:22.50`	`rkioko`	`ok`
`22:22.55`	`rue_mohr`	`we have ANGRY users`
`22:23.19`	`rkioko`	`service delivery is of utmost importance`
`22:23.21`	`rue_mohr`	`to the point where they have stated that asterisk is cr** and demand that the whole thing be replaced`
`22:23.30`	`rkioko`	`reliable service`
`22:23.34`	`msw`	`rue_mohr: oh no`
`22:23.41`	`rue_mohr`	`its already too late`
`22:23.43`	`*** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net)`
`22:23.48`	`rkioko`	`sorry about that`
`22:23.49`	`msw`	`rue_mohr: :-(`
`22:24.04`	`rue_mohr`	`everytime a user has ANY kind of call problem (be it our system or not) they get irate`
`22:24.37`	`tainted-`	`well they expect POTS reliability`
`22:24.54`	`rue_mohr`	`right now I'm being beat on for voicemail problems, but I'm having a lot of trouble getting help`
`22:25.22`	`*** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk)`
`22:25.45`	`tainted-`	`what kind of problems?`
`22:26.05`	`rkioko`	`i need feedback from someone who has done stress tests on second generation digium`
`22:26.07`	`rue_mohr`	`well, some of the voicemails left by external callers have no audio`
`22:26.15`	`rkioko`	`or sango a104d`
`22:26.20`	`rkioko`	`sorry sangoma`
`22:26.35`	`tainted-`	`interesting`
`22:26.36`	`tainted-`	`what codec`
`22:26.37`	`rkioko`	`with all 4 e1/t1`
`22:26.57`	`rue_mohr`	`tainted_ g723`
`22:27.13`	`gammacoder`	`anyone had an asterisk install doesn't produce any sound for VM prompts or Digial Receptionist?`
`22:27.35`	`rue_mohr`	`gammacoder do you have .gsm files in....`
`22:27.49`	`rue_mohr`	`var/lib/asterisk/sounds/`
`22:27.50`	`rue_mohr`	`??`
`22:28.06`	`*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)`
`22:28.16`	`rue_mohr`	`I suspect your have a permissions problem or need http://ftp.digium.com/pub/asterisk/asterisk-sounds-1.2.1.tar.gz`
`22:29.08`	`[av]bani`	`can you set different iptos for rtp and sip?`
`22:29.09`	`gammacoder`	`rue_mohr: I have /usr/src/asterisk/codecs/gsm`
`22:29.09`	`rue_mohr`	`please tell me if I'm right, cause I dont know if I am...`
`22:29.25`	`[av]bani`	`sip.conf has tos= but what about rtp.conf? same?`
`22:29.48`	`rue_mohr`	`gammacoder no, I'm pretty sure you need a while bunch (about 778) of files in /var/lib/asterisk/sounds/`
`22:30.12`	`gammacoder`	`rue_molu: and I have /var/lib/asterisk/sounds`
`22:30.24`	`gammacoder`	`sorry for the confusion`
`22:30.28`	`rue_mohr`	`tainted- might you be able to help me diagnose my problems?`
`22:30.39`	`rue_mohr`	`gammacoder are there files in it? what are the permissions?`
`22:31.09`	`gammacoder`	`tons of files 644 permissions for asterisk:asterisk`
`22:31.11`	`copantl`	`rue_mohr: any idea?`
`22:31.34`	`*** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker)`
`22:31.41`	`*** join/#asterisk beebz (i=bbz@adsl-70-128-78-21.dsl.stlsmo.swbell.net)`
`22:32.15`	`gammacoder`	`this is on an Asterisk@Home 2.2 install by the way`
`22:32.22`	`rue_mohr`	`.gsm files?`
`22:32.26`	`rue_mohr`	`copantl which?`
`22:32.39`	`*** join/#asterisk saftsack (n=saftsack@p54A7E1B1.dip.t-dialin.net)`
`22:32.43`	`wwalker`	`I've found online (asterisk mailing list archive) that a Polycom IP500 will accet the MAC adddress as the admin password. I've tried this multiple times. anyone know how to do this? how many digits?`
`22:32.47`	`gammacoder`	`yes - gsm files`
`22:32.59`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`22:33.40`	`gammacoder`	`hmmm - no on hold music either`
`22:33.47`	`tainted-`	`could be audio codec issues`
`22:33.54`	`tainted-`	`have u tried different codecs? rue_mohr`
`22:34.16`	`rue_mohr`	`hmmm`
`22:34.24`	`gammacoder`	`i did recently upgrade processors - do the codecs/timing sources get bothered by processor changes?`
`22:34.34`	`rue_mohr`	`if it was a codec issue, do you know what errors or messages I might se in teh logs?`
`22:34.47`	`azzie`	`good evening. can a guru tell me whether "Trying" SIP message should contain Max-Forwards header or not ?`
`22:35.19`	`gammacoder`	`i see no errors in the /var/log/asterisk/full - and see the normal progression of attempting to play throught the voicemail greetings`
`22:35.52`	`rue_mohr`	`WARNING[784]: file.c:583 ast_readaudio_callback: Failed to write frame`
`22:35.54`	`rue_mohr`	`WARNING[16560]: app_voicemail.c:4946 vm_authenticate: Couldn't read username`
`22:35.54`	`rue_mohr`	`NOTICE[19687]: chan_iax2.c:7150 socket_read: Rejected connect attempt from 10.255.40.41, request '6882@process-routing' does not exist`
`22:35.58`	`rue_mohr`	`anything like those?`
`22:36.52`	`l2trace`	`does Binding sipusers to mysql/Asterisk/SipPeers`
`22:37.10`	`l2trace`	`mean that it is using the table SipPeers`
`22:37.11`	`l2trace`	`>`
`22:37.13`	`l2trace`	`?`
`22:37.24`	`gammacoder`	`no signs of 784, 16560, or 19687 anywhere`
`22:38.28`	`*** join/#asterisk Lee619 (n=Lee@netblock-66-245-227-194.dslextreme.com)`
`22:38.48`	`Lee619`	`hello`
`22:39.12`	`Lee619`	`anybody home?`
`22:39.22`	`gammacoder`	`here`
`22:40.51`	`Lee619`	`i figured out my FWD problem...`
`22:41.03`	`Lee619`	`turns out the reason that it wouldn't register was an FWD problem....`
`22:41.09`	`*** part/#asterisk george435 (n=tyu@194.154.204.68.cfl.res.rr.com)`
`22:41.30`	`Lee619`	`it's supposed to take 20 minutes for FWD to become IAX enabled once selected on the FWD website...`
`22:41.36`	`Lee619`	`but it now takes more then 24 hours....`
`22:41.53`	`Lee619`	`took almost 2 days for my FWD number to become IAX enabled... geesh`
`22:42.06`	`Lee619`	`anyhow... it now registers...`
`22:42.19`	`Lee619`	`thanks to all who provided suggestions yesterday... :)`
`22:44.10`	`*** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM)`
`22:44.14`	`*** part/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM)`
`22:44.23`	`*** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM)`
`22:50.37`	`*** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net)`
`22:51.20`	`*** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net)`
`22:52.14`	`rue_mohr`	`is there a GOOD site for debugging * ?`
`22:52.28`	`rue_mohr`	`I'm finding more questions than answers`
`22:52.49`	`rue_mohr`	`and every few min...`
`22:52.50`	`*** join/#asterisk dcoulson (n=dcoulson@207.166.203.178)`
`22:53.44`	`rue_mohr`	`minutes, I have users calling me with more voicemails that have no audio`
`22:53.52`	`rue_mohr`	`(that was one jsut now)`
`22:54.23`	`gammacoder`	`i haven't found anything better than www.voip-info.org and lists.digium.com - but they don't really qualify as good`
`22:55.09`	`rue_mohr`	`:( I was hoping people would be a little better here, but I'm not feeling the love`
`22:56.08`	`*** join/#asterisk bjohnson (n=bjohnson@i216-58-48-168.cybersurf.com)`
`22:56.36`	`kink0`	`there anyway to send something like answer(channel) ?`
`22:57.16`	`rue_mohr`	`sorry, I have no idea`
`22:57.53`	`rue_mohr`	`hmm, what I need is a utility to reprocess logs into some sort of call history chart`
`22:58.03`	`rue_mohr`	`perl...`
`22:58.09`	`*** join/#asterisk fifer (n=sirfifer@207.202.227.161)`
`22:58.11`	`rue_mohr`	`and some sort of state engine...`
`22:58.44`	`rue_mohr`	`question that I'm sure one fo you knows the answser on`
`22:58.56`	`fifer`	`I've been going through a much more dificult than it shoudl be installation of AMP (not something I blame on AMP)`
`22:59.07`	`Burgwork`	`can I make an asterisk<-->asterisk call ring a different ring tone to an inbound call?`
`22:59.08`	`rue_mohr`	`in the full log, does Jan 12 14:58:01 DEBUG[6999]: the [#] represent a uniq call thread?`
`22:59.28`	`fifer`	`I have finaly taken care of all the dependancies and most of the installation steps, but I have run into a problem with the actual amp install script`
`22:59.45`	`fifer`	`Anyone here able to tackle taht?`
`22:59.48`	`rue_mohr`	`fifer sorry, I have no idea`
`23:00.09`	`fifer`	`actual error: /usr/bin/php: symbol lookup error: /usr/lib/libgssapi_krb5.so.2: undefined symbo`
`23:00.22`	`fifer`	`krb5 is one of the packages I updated/added`
`23:00.32`	`fifer`	`installed fine (apeard to)`
`23:00.34`	`gammacoder`	`fifer: not sure`
`23:00.39`	`rue_mohr`	`fifer tried #php?`
`23:00.43`	`fifer`	`Hm......`
`23:00.47`	`rue_mohr`	`or #apache?`
`23:00.51`	`fifer`	`rue_mohr: good idea`
`23:01.08`	`rue_mohr`	`they might not know asteresk, but may be able to give you generic help on those`
`23:01.37`	`rue_mohr`	`in the full log, does Jan 12 14:58:01 DEBUG[6999]: the [#] represent a uniq call thread?`
`23:01.56`	`gammacoder`	`rue_mohr: I'm a novice - not sure`
`23:02.22`	`rue_mohr`	`heh, how long you been working with it?`
`23:03.13`	`gammacoder`	`about 3 months`
`23:03.24`	`rue_mohr`	`stead or on and off?`
`23:04.02`	`*** join/#asterisk outofjungle (n=outofjun@61.247.249.151)`
`23:04.52`	`enemy^x`	`can someone tell me how to get the presence working within xten phone? Do I need to put ser into my system to make it work or can it be done within asterisk=?`
`23:04.56`	`gammacoder`	`i have 1 smallbiz with ~10 users in production, another with ~25 users close to rollout, and a thrid with ~10 users about to start`
`23:05.38`	`gammacoder`	`but my time has been roughly 10 hours a week`
`23:06.05`	`rue_mohr`	`sounds like your still more experianced than me...`
`23:06.20`	`*** join/#asterisk cripito (n=ncripito@63.161.160.195)`
`23:06.34`	`gammacoder`	`i'm no guru`
`23:06.37`	`cripito`	`hi!`
`23:07.07`	`rob0`	`rue_mohr: probably the PID, I would guess`
`23:07.14`	`rue_mohr`	`I'm the one working on fixing up some problems in a system the great kb1canobe built, something about water and over my head, I know linux, which is a start`
`23:07.37`	`rue_mohr`	`rob0 can you give me any hints to help me trace the life of a call in the logs?`
`23:08.28`	`rob0`	`I could look at my own logs and see ...`
`23:08.57`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`23:09.24`	`gammacoder`	`rue_mohr: I've had the same issue decyphering the logs to find an individual call thread`
`23:10.33`	`rob0`	`... no, I don't know much about it. Does the wiki have anything?`
`23:10.50`	`rue_mohr`	`it would be nice if there were state machine 'thread' identities in the logs, maybe its time for a patch`
`23:10.57`	`*** join/#asterisk zotz (n=zotz@24.231.47.175)`
`23:11.19`	`rue_mohr`	`which is funny, cause it would probably be easier for me to do that than the missing audio probelm I'm working on`
`23:12.49`	`detatch`	`hey can someone tell me what the extensions.ael is for in 1.2.1?`
`23:14.25`	`fifer`	`new language for dialplans`
`23:14.39`	`fifer`	`AEL is more like a simple programing language`
`23:15.00`	`fifer`	`extensions.ael is the AEL replacement/augmentation of the old extensions.conf`
`23:15.14`	`detatch`	`hmm`
`23:15.16`	`fifer`	`you can have both (hav'nt tried myself)`
`23:15.21`	`gammacoder`	`is there any easy way to read / write to the asterisk db from the command line?`
`23:15.36`	`detatch`	`fifer: is there a way to have asterisk not load it?`
`23:15.54`	`detatch`	`otherwise i get warning errors in the cli when its not there, of course`
`23:16.07`	`fifer`	`Just leave it alone.`
`23:16.39`	`fifer`	`If you have a new Asterisk install and did a make samples, you WILL have a extensions.ael, but you do not need to do anything with it unless you need/want to.`
`23:16.51`	`detatch`	`hmm`
`23:16.51`	`detatch`	`ok`
`23:20.28`	`*** join/#asterisk pembajak_sejati (i=budi@142.179.115.60)`
`23:21.09`	`wwalker`	`What would the terms to look for be for this. I want to be able to "announce" a call that is being forwarded in house via that phones (IP 500) speaker.`
`23:22.43`	`ObsidianX`	`how does asterisk get its MD5 hash?`
`23:23.02`	`gammacoder`	`wwalker: search about auto-answer`
`23:23.19`	`ObsidianX`	`i put one thing into the config, and the error in the console "hash != hash" neither hashes are what i've made`
`23:23.28`	`wwalker`	`gammacoder thx!`
`23:24.12`	`ObsidianX`	`wow yeah, and the hash changes every time i try and login`
`23:24.13`	`*** join/#asterisk Slackuser_ (n=FullT@200.195.76.25)`
`23:24.31`	`gammacoder`	`wwalker: to enable the speaker on a Grandstream GXP-2000 w/ firmware 1.0.1.13 or greater (for instance) you'd do a SIPAddHeader(Call-Info: answer-after=0)`
`23:25.08`	`wwalker`	`gammacoder thx`
`23:25.27`	`gammacoder`	`np`
`23:25.48`	`*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)`
`23:27.46`	`shmaltz`	`anybody here has the latest cisco firmware for 7960?`
`23:28.04`	`pembajak_sejati`	`hi is it possible to do like this:`
`23:28.06`	`*** join/#asterisk Skkip (n=Skipper@216.160.91.91)`
`23:28.10`	`pembajak_sejati`	`exten => 55317,1,dial(SIP/hakan)`
`23:28.10`	`pembajak_sejati`	`exten => 317,1,dial(SIP/hakan)`
`23:28.10`	`pembajak_sejati`	`exten => 317,2,dial(SIP/55317)`
`23:29.09`	`pembajak_sejati`	`maybe dial(SIP/55317) is not valid, how to dial 55317?`
`23:29.18`	`*** join/#asterisk tetsuzan (n=rider@201.2.206.4)`
`23:31.07`	`*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)`
`23:31.33`	`*** join/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net)`
`23:31.47`	`*** join/#asterisk pembajak_sejati (n=budi@202.161.95.42)`
`23:32.07`	`FarrisG`	`Can anyone help me figure out why we're no longer sending caller ID info correctly? I'm kind of a noob, and SBC tells me it's all setup correctly on their end.`
`23:32.49`	`gammacoder`	`is SBC specifying the Caller ID, or are they passing through whatever you specify?`
`23:33.12`	`FarrisG`	`gammacoder: SBC is specifying`
`23:33.21`	`gammacoder`	`then it appears to be their issue`
`23:33.45`	`gammacoder`	`my telco passes whatever I specify`
`23:33.56`	`FarrisG`	`gammacoder: Where do you specify it? Maybe they're lying`
`23:34.57`	`gammacoder`	`in AMP it is in each extension configuration, under "outbound callerid"`
`23:35.11`	`FarrisG`	`gammacoder: Ahh... I don't use AMP`
`23:36.54`	`gammacoder`	`OK - make sure you've got entries in one of the extensions.conf files like this:`
`23:37.07`	`gammacoder`	`ECIDxxxx = xxx-xxx-xxxx`
`23:37.27`	`gammacoder`	`where ECID[extenstion] = caller id number`
`23:37.33`	`*** join/#asterisk viLeR (i=1000@66.128.47.232)`
`23:37.50`	`FarrisG`	`I have no ECID entries`
`23:38.12`	`gammacoder`	`perhaps you should wink`
`23:39.06`	`*** part/#asterisk Burgwork (n=corey@S010600131016cf6f.gv.shawcable.net)`
`23:40.27`	`*** join/#asterisk chiardon (n=chiardon@200.71.58.39)`
`23:40.45`	`chiardon`	`hello`
`23:41.03`	`FarrisG`	`gammacoder: Do I need one of those for each extension, or can I somehow set it globally?`
`23:42.00`	`gammacoder`	`i've got them for each extension - not sure if you can set it globally`
`23:43.26`	`gammacoder`	`FarrisG: apparently you can set them per trunk. Like: OUTCID_1 = 123-456-7890`
`23:48.16`	`*** join/#asterisk brookshire (n=nubb@gateway.digium.com)`
`23:49.50`	`*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)`
`23:51.38`	`FarrisG`	`gammacoder: is that set in extensions.conf, too?`
`23:52.05`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`23:52.05`	`*** mode/#asterisk [+o anthm] by ChanServ`
`23:52.09`	`gammacoder`	`FarrisG: yep - for me its in extensions_additional.conf`
`23:52.41`	`[TK]D-Fender`	`Oh joy... I hear an AMP in the air!`
`23:54.23`	`copantl`	`ok, i reinstalled my zaptel and the module is recognice, i did a lsmod and it's there ..cte11xp`
`23:54.58`	`copantl`	`but i saw the E1 led and is off!!`
`23:55.33`	`copantl`	`what can be possible wrong?'`
`23:57.07`	`gammacoder`	`copantl: is you physical wiring correct - I was bitten by the need for a T1 crossover cable at that point in my install`
`23:57.49`	`copantl`	`it was working, i just made a reintallation`
`23:57.50`	`*** join/#asterisk rue_work (n=not@h24-207-96-51.cst.dccnet.com)`
`23:58.04`	`gammacoder`	`gotcha`
`23:58.38`	`rue_work`	`:/ something int eh system kills dialtones from the audio steams, this dosn't help me use the only modem I can access to debug incomming voicemail problems`
`23:58.56`	`copantl`	`can be a fisical problem?`
`23:58.59`	`rue_work`	`makes me wish I was able to find a modem that voice worked on`
`23:59.40`	`rue_work`	`have to find a external one though cause they are all winmodems with no linux support for vioce functions`
`23:59.48`	`*** join/#asterisk Pzykotic (n=pzykotic@vnnyca-cuda1-cablebdl-70-34-198-51.vnnyca.adelphia.net)`

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