irclog2html for #asterisk on 20070228

`00:00.35`	`*** join/#asterisk robby____ (n=robby@203.63.126.9)`
`00:01.15`	`*** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)`
`00:01.42`	`robby____`	`hi all, in the manager interface, when you do queuestatus, whats the time frame around calls taken etc, like, does it reset daily or something like that?`
`00:02.36`	`notoriousrab`	`hey, anyone know if WaitExten application has changed in version 1.4 - trying to set up an auto attendant`
`00:03.44`	`infinity1`	`hwo come asterisk 1.4.0 is still the latest? usually 1.x.0 comes out and about a week later 1.x.1 is out.`
`00:06.16`	`*** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com)`
`00:06.19`	`*** join/#asterisk trogs (n=dwarf@thor.jedi.net.nz)`
`00:07.57`	`jaxxan`	`i figured it out, the sound file just isn't detailed enough`
`00:08.29`	`jaxxan`	`and if you press * while it's waiting for an extension, it reads the * as the extension, but while it's playing the pbx-invalid sound file you can press * to back out to the main menu`
`00:10.34`	`eald`	`Hi, In order to test performance/stability for asterisk 1.2 and 1.4 (along with a etch vs sarge) in the same hardware I'm thinking in a chroot system, for this, I need to make some consideration apart from ports for mysql, and asterisk (sip and rtp)? maybe I have to compile one ztdummy with another name?`
`00:10.57`	`jaxxan`	`the 'press3 for adv options' in the main voicemail menu doesn't need to be there. it's called when listening to a message. it shouldn't be in the main menu`
`00:11.15`	`*** join/#asterisk apardo (n=apardo@87.217.145.41)`
`00:11.29`	`jaxxan`	`if you enter it from the main menu, it just tells you to press * to go back to the main menu`
`00:13.55`	`*** part/#asterisk trogs (n=dwarf@thor.jedi.net.nz)`
`00:15.25`	`*** join/#asterisk LanceSnyder (n=LanceSny@adsl-10-1-235.mia.bellsouth.net)`
`00:16.27`	`*** part/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net)`
`00:20.28`	`jaxxan`	`there is no ability to listen to your recorded unavailable or busy message before recording a new one.`
`00:21.16`	`errr`	`when I run pri show span 1 from asterisk cli I get no PRI running on span 1, but Im not sure how to trouble shoot the issue. Can anyone help?`
`00:21.18`	`generalhan`	`jaxxan: you can go to /var/spool/asterisk/voicemail and listen to the messages there manually before recording a new one !`
`00:22.01`	`creature_`	`jaxxan: what version are you running?`
`00:22.02`	`jaxxan`	`that's not the point`
`00:22.08`	`jaxxan`	`1.2.12.1`
`00:22.16`	`jaxxan`	`my voicemail users dont have console access to my server ya know`
`00:22.17`	`creature_`	`jaxxan: i'm able to listen to the recorded message before accepting it`
`00:22.22`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`00:22.30`	`jaxxan`	`what version are you running ?`
`00:22.38`	`creature_`	`jaxxan: and then i can decide to delete it or save it`
`00:22.46`	`generalhan`	`jaxxan: why not create an exten => that plays their message`
`00:22.56`	`jaxxan`	`i can't`
`00:23.00`	`jaxxan`	`these are mobile phone users`
`00:23.07`	`generalhan`	`...`
`00:23.20`	`jaxxan`	`creature_: what version are you running?`
`00:23.20`	`creature_`	`jaxxan: currently 1.4.0 but it worked great with 1.2.0 aswell`
`00:23.32`	`creature_`	`one moment, i'll try it again`
`00:23.32`	`generalhan`	`do they not still work off the dialplan of your main asterisk server ?`
`00:23.56`	`jaxxan`	`generalhan: they do, but assigning a number just so a user can listen to their greetings is out of the question`
`00:24.13`	`jaxxan`	`these are PBX users, they are voicemail customers`
`00:24.18`	`jaxxan`	`erm.. these aren't pbx users`
`00:24.29`	`generalhan`	`i see ...`
`00:25.06`	`creature_`	`jaxxan: trying right now`
`00:29.02`	`creature_`	`jaxxan: works just fine, need to sleep now so i cant help you to investigate :/`
`00:29.29`	`jaxxan`	`that's 1.2.x ?`
`00:29.57`	`jaxxan`	`think it's my voicemail.conf ?`
`00:30.53`	`*** part/#asterisk errr (n=errr@fedora/errr)`
`00:34.18`	`k-man_`	`do i just have to modprobe the ztdummy to get timing to work with 2.6 kernels? or i need to also configure something in asterisk?`
`00:36.09`	`jaxxan`	`mine totally makes you re-record your message every time when pressing 0 -> 1`
`00:36.50`	`*** join/#asterisk Mentifisto (i=Loki@unaffiliated/mentifisto)`
`00:37.04`	`*** part/#asterisk Mentifisto (i=Loki@unaffiliated/mentifisto)`
`00:38.56`	`*** join/#asterisk LanceSnyder (n=LanceSny@adsl-10-1-235.mia.bellsouth.net)`
`00:39.14`	`*** join/#asterisk topping (n=topping@dsl093-079-162.sfo1.dsl.speakeasy.net)`
`00:39.20`	`jaxxan`	`k-man: i have no clue, i get my timing from a dms100`
`00:41.19`	`k-man_`	`ok`
`00:41.20`	`k-man_`	`thanks`
`00:43.52`	`SplasPood`	`heh, digium's phone system is broken`
`00:43.56`	`*** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com)`
`00:43.57`	`SplasPood`	`thats... comforting ;)`
`00:46.23`	`*** join/#asterisk EvilRick (n=bob@41.207.226.26)`
`00:46.38`	`tzanger`	`ok`
`00:46.41`	`EvilRick`	`I have 2 isdn (BRI) cards in my system as well as a E1 PRI attached to a cell router. the PRI seems to be all green and configured but when I try use it, it reports line congestion .. ie cant dial wight now`
`00:46.44`	`EvilRick`	`any ideas?`
`00:46.46`	`tzanger`	`chan_bluetooth or chan_cellphone here I come`
`00:46.58`	`EvilRick`	`-- Called g1/0874561392`
`00:47.07`	`EvilRick`	`<PROTECTED>`
`00:47.13`	`EvilRick`	`<PROTECTED>`
`00:47.19`	`EvilRick`	`<PROTECTED>`
`00:47.25`	`EvilRick`	`<PROTECTED>`
`00:47.35`	`EvilRick`	`g0 are my 4 isdn lines and they seem to work just fine (1-2 and 4-5 3,6 are for signaling)`
`00:48.01`	`k-man_`	`well.. it seems i have entered into the compiling hell of asterisk and zaptel drivers`
`00:48.24`	`k-man_`	`as i need the zaptel dummy driver for the 1khz clock`
`00:48.28`	`k-man_`	`i get this error:`
`00:49.24`	`k-man_`	`http://pastebin.ca/375120`
`00:49.29`	`k-man_`	`any idea why i would get that?`
`00:49.34`	`LanceSnyder`	`that's the error?`
`00:49.43`	`k-man_`	`see pastebin`
`00:49.43`	`LanceSnyder`	`just a link ?`
`00:49.45`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`00:49.45`	`*** mode/#asterisk [+o mog] by ChanServ`
`00:50.31`	`Strom_M`	`it's teh mog`
`00:50.38`	`k-man_`	`mog?`
`00:50.44`	`k-man_`	`oh`
`00:50.45`	`k-man_`	`mog`
`00:51.25`	`*** join/#asterisk rene- (n=rene-@200.34.66.137)`
`00:51.32`	`rene-`	`ok\`
`00:51.52`	`rene-`	`using irc from vim is REALLY GEEKY`
`00:52.04`	`robby____`	`anyone got a link for the syntax of sip hints in 1.4.* ?`
`00:52.06`	`rene-`	`havent done it but i was looking at the plugin`
`00:52.56`	`robby____`	`hint(SIP/9000) 9000 => &ael-std-exten-ael(${EXTEN},SIP);`
`00:53.01`	`robby____`	`is what i have, but doesnt appear to be working`
`00:56.51`	`*** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)`
`00:57.08`	`*** join/#asterisk topping (n=topping@dsl093-079-162.sfo1.dsl.speakeasy.net)`
`00:59.07`	`k-man_`	`hmm... looks like this is a reported issue, but no visible fix for it`
`00:59.09`	`k-man_`	`http://bugs.digium.com/view.php?id=9091&nbn=2`
`00:59.57`	`*** join/#asterisk zpertee (n=chatzill@cpe-65-25-121-5.neo.res.rr.com)`
`01:00.21`	`zpertee`	`I need help with a .call file. can someone please help me`
`01:01.23`	`*** join/#asterisk ZaVoid (n=colin@c-67-165-25-195.hsd1.pa.comcast.net)`
`01:01.50`	`ZaVoid`	`hello... quick stupid question.. if i reload from the console to allow another port in sip.conf for sip to listen on will it drop all active calls/`
`01:03.31`	`*** join/#asterisk shodan (n=shodan@ip109.96-113-216.pppoe1.joliette.intermonde.net)`
`01:04.04`	`*** join/#asterisk sfbosch (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com)`
`01:04.22`	`sfbosch`	`Greetings - I have a voice mail question`
`01:04.39`	`sfbosch`	`I am not getting the "unavailable" message when a user calls voice mail`
`01:04.47`	`tuan_modulis`	`queues are giving me big headaches... for some reason, I get multiple hangups from one call`
`01:05.34`	`sfbosch`	`CLI output doesn't offer much`
`01:05.34`	`tuan_modulis`	`it's frustrating since the hangup clears out a bunch of data i needed`
`01:08.08`	`tuan_modulis`	`anyone else experience the multiple hangup problem from queues?`
`01:08.23`	`tuan_modulis`	`cuz when using DeadAGI... uuuuuggghh`
`01:09.42`	`tuan_modulis`	`maybe there's another way to keep track of dead channels...`
`01:10.47`	`sfbosch`	`this is one chatty channel.`
`01:10.52`	`ZaVoid`	`lol`
`01:10.59`	`sfbosch`	`Sorry, tuan_modulis, I can't help you :(`
`01:11.25`	`sfbosch`	`Voice mail? Anyone? Why aren't my saved prompts being played to callers?`
`01:11.40`	`tuan_modulis`	`it sure is, all our problems tend to be ambiguous`
`01:11.42`	`tuan_modulis`	`heh`
`01:12.19`	`zpertee`	`anyone know anything about .call files`
`01:13.35`	`*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)`
`01:13.38`	`ZaVoid`	`nope`
`01:13.55`	`Strom_M`	`zpertee: ask a specific question and you may get an answer`
`01:16.18`	`*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)`
`01:20.51`	`tuan_modulis`	`it's time to hack... the true secret behind quickly developped enterprise solutions`
`01:21.03`	`ZaVoid`	`lol`
`01:22.13`	`robby____`	`does anyone have a link to detailed documentation about action queuestatus?`
`01:22.25`	`robby____`	`were using trial and error atm to try and work out its actual behaviour`
`01:22.35`	`robby____`	`timeframes etc`
`01:22.56`	`robby____`	`have looked on the net and cant find anything much`
`01:27.44`	`tuan_modulis`	`well, dunno much about that, but i know some of those softwares probably look at /var/log/asterisk/queue_log`
`01:28.37`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-2-16.red.bezeqint.net)`
`01:40.17`	`*** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net)`
`01:41.39`	`JT`	`k-man_: what zap and ast version?`
`01:42.24`	`wunderkin`	`is it necessary to have some kind of baseline contact directory on a polycom phone? i just want to make sure it is ok not to... this time i wont make a zero length file... :P`
`01:42.38`	`*** join/#asterisk dj-fu (n=ajc@203-211-107-82.ue.woosh.co.nz)`
`01:43.14`	ez`	`i m looking to buy a really good dual fxs adapter, which one do you recommend ?`
`01:46.24`	`SplasPood`	`Qwell[]: so its looking like I got a bad TDM800P`
`01:50.45`	`rene-`	`whats a tdm800p?`
`01:50.59`	`rene-`	`wow`
`01:51.05`	`rene-`	`never seen those`
`01:53.11`	`SplasPood`	`its a bigger TDM400`
`01:53.13`	`SplasPood`	`I suppose`
`01:53.24`	`SplasPood`	`2x as big ;)`
`01:53.46`	`*** join/#asterisk Kumbang (n=kumbang@167.205.24.67)`
`01:54.17`	`ZaVoid`	`lol`
`01:54.26`	`ZaVoid`	`sounds like a POS regardless`
`01:55.17`	`SplasPood`	`haha, how's that?`
`01:55.59`	`ZaVoid`	`its got tdm in the name`
`01:57.41`	`robby____`	`whats the svn command to grab down trunk?`
`01:58.00`	`robby____`	`svn export svn://svn.digium.com/svn/asterisk/trunk asterisk isnt working`
`01:59.35`	`LanceSnyder`	`google is your friend.`
`01:59.43`	`robby____`	`trued that`
`01:59.45`	`robby____`	`tried*`
`01:59.58`	`*** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net)`
`02:00.59`	`LanceSnyder`	`man svn`
`02:05.26`	`*** join/#asterisk wunderkin- (i=kev@ip72-208-3-221.ph.ph.cox.net)`
`02:06.17`	`SplasPood`	`LanceSnyder: well unless his URL is wrong... thats proper syntax`
`02:08.09`	`*** join/#asterisk topping (n=topping@dsl093-079-162.sfo1.dsl.speakeasy.net)`
`02:08.18`	`SplasPood`	`LanceSnyder: svn.digium.com is simply down.`
`02:08.28`	`SplasPood`	`as well as their support line for installations`
`02:08.32`	`SplasPood`	`or it was an hour ago`
`02:09.40`	`LanceSnyder`	`SplasPood, ahh ok. well at least that is right syntax`
`02:10.00`	`SplasPood`	`see`
`02:10.22`	`SplasPood`	`the quick jump to shouting RTFM at someone only works if they aren't already correct.`
`02:11.24`	`orlock`	`damn, the new google maps images of sydney are cool`
`02:13.06`	`elriah`	`lol, how the heck do you logoff an agent in 1.2?`
`02:14.55`	`elriah`	`Is there an agentlogoff command or have I just missed something entirely?`
`02:16.55`	`JT`	`SplasPood: the TDM800P is technically a small TDM2400P not a big TDM400P :)`
`02:19.14`	`SplasPood`	`JT: ahh.. a bad guess :)`
`02:19.16`	`*** join/#asterisk test34 (n=test34@unaffiliated/test34)`
`02:19.35`	`wunderkin`	`elriah, on the cli.. agent logoff..`
`02:19.38`	`SplasPood`	`elriah: AgentLogoff`
`02:19.43`	`SplasPood`	`yea`
`02:20.01`	`SplasPood`	`wasn't there a dialplan function?`
`02:20.10`	`SplasPood`	`an app, rather`
`02:20.27`	`elriah`	`AgentLogoff? I swear I can't find it in 1.2??`
`02:20.39`	`SplasPood`	`well it might not exist`
`02:20.39`	`SplasPood`	`heh`
`02:20.47`	`SplasPood`	`but agent logoff from the cli works`
`02:20.52`	`SplasPood`	`you could use System() to call it`
`02:21.16`	`SplasPood`	`asterisk -r -x 'agent logoff Agent/10011'`
`02:21.24`	`elriah`	`I think I found a way.. brb`
`02:21.48`	`SplasPood`	`yea I think you can pass no callback or something`
`02:21.57`	`CrashHD`	`I had call-limit = 10 but calls were hitting that limit for phones even though they had no other active calls....why is that?`
`02:21.57`	`wunderkin`	`there should be an app and ami too`
`02:23.18`	`*** join/#asterisk kikoafonso (i=Kiko@201.37.194.100)`
`02:23.19`	`*** join/#asterisk mrc3__ (n=mrc3@189.157.107.61)`
`02:25.49`	`*** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net)`
`02:26.05`	`mrc3__`	`hey, hello! i have a sip user named jsmith, registered in asterisk. if i do "dial SIP/jsmith", is it supposed to work from the console just like that?`
`02:28.05`	`[TK]D-Fender`	`mrc3_ : if you're intending on using chan_oss as a local "soft-phone", sure`
`02:28.28`	`[TK]D-Fender`	`mrc3_ : Though normally you'd test between devices other than the console.`
`02:28.52`	`mrc3__`	`right, right. i finally got my pap2 AND a softphone to work, and now i'm on to my dial plan`
`02:29.14`	`elriah`	`SplasPood: Yea, but using a system call to logoff agents somehow diesn't seem right. It's like running Apache as root or something.`
`02:29.16`	`mrc3__`	`ah, wait, it worked`
`02:29.29`	`*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)`
`02:29.52`	`mrc3__`	`i don't know why i can't call the pap2, though. i'll bug again in a jiffy`
`02:30.03`	`*** part/#asterisk rene- (n=rene-@200.34.66.137)`
`02:30.34`	`SplasPood`	`elriah: heh.. well... As much as I love it... I'd say ... Yea that's asterisk`
`02:30.56`	`elriah`	`So how do other folks logoff agents?`
`02:31.05`	`SplasPood`	`I think I'm doing it the way I described`
`02:31.07`	`elriah`	`Just don't, wait for them to timeout in a queue?`
`02:31.09`	`elriah`	`Ahh.`
`02:31.28`	`elriah`	`SplasPood: Thanks for the help, btw.`
`02:31.31`	`SplasPood`	`np`
`02:31.35`	`SplasPood`	`I don't time agents out at all`
`02:31.42`	`SplasPood`	`and I do callbacks`
`02:32.02`	`elriah`	`How do your agents "get out" of the queue? Are calling agent logout with System()?`
`02:32.15`	`SplasPood`	`yea`
`02:32.18`	`elriah`	`Well, when in Rome ...`
`02:32.20`	`SplasPood`	`123 logs em in`
`02:32.21`	`elriah`	`:p`
`02:32.23`	`SplasPood`	`456 logs em out`
`02:32.25`	`SplasPood`	`that deal`
`02:32.31`	`SplasPood`	`there might be a better way`
`02:32.43`	`SplasPood`	`I presume I would have used it at the time if it existed or I was aware of it`
`02:32.46`	`SplasPood`	`however..`
`02:34.16`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`02:35.34`	`elriah`	`[TK]D-Fender: Hey! What gives? ;}`
`02:35.54`	`[TK]D-Fender`	`elriah : Not me :) No quarter!`
`02:36.16`	`[TK]D-Fender`	`whee`
`02:36.32`	`elriah`	`lol`
`02:36.36`	`mrc3__`	`this is what asterisk complains about: "WARNING[881]: channel.c:2752 ast_channel_make_compatible: No path to translate from SIP/ddiaz-0819a7c0(262144) to SIP/arimba-0818cf10(4)"`
`02:37.15`	`mrc3__`	`it says that it's got sip response 410 "gone", and that SIP/ddiaz is circuit-busy`
`02:37.22`	`elriah`	`SplasPood: What distro are you runnng? My Asterisk installs are in sudo environments so I don't think a shell command is going to work right...`
`02:37.42`	`[TK]D-Fender`	`mrc3_ : * can't transcode the call and the endpoint have no compatible codecs. Not sure but it looks like one side is G.729`
`02:38.04`	`mrc3__`	`[TK]D-Fender, let me check for that, thanks`
`02:38.17`	`SplasPood`	`elriah: System will run it with the same privs as asterisk, thus it should work fine... might need to change the path to asterisk, of course..`
`02:38.34`	`[TK]D-Fender`	`elriah : Should work.... after all, it's the Asterisk daemon user calling "System", no?`
`02:38.35`	`SplasPood`	`elriah: I used debian`
`02:39.27`	`SplasPood`	`s/use`
`02:39.30`	`SplasPood`	`erm`
`02:39.32`	`SplasPood`	`yea.`
`02:44.27`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`02:46.55`	`*** join/#asterisk shodan (n=shodan@ip109.96-113-216.pppoe1.joliette.intermonde.net)`
`02:48.11`	`elriah`	`Wow. No AgentLogoff command. Guess I fincally found something to contribute to Asterisk.. now where did I put that C++ reference..`
`02:49.39`	`elriah`	`When I use autoagentlogoff in agents.conf, will it logoff everybody that doesn't answer within my timeout period?`
`02:49.41`	`*** join/#asterisk Strom_M (n=strom@63.110.13.126)`
`02:51.12`	`*** join/#asterisk bigred (n=ian@75-1-209-228.lightspeed.snantx.sbcglobal.net)`
`02:51.42`	`bigred`	`I just setup my vitelity settings, and now when I enter in the console, there is some sort of loop and I get the "CLI>" prompt over and over until I kill the process`
`02:51.58`	`*** join/#asterisk arcanine (n=arcanine@203.82.44.179)`
`03:11.19`	`LanceSnyder`	`bigred, check asterisk_safe`
`03:11.25`	`LanceSnyder`	`oh he's gone`
`03:11.34`	`LanceSnyder`	`well.. guess he didn't want help`
`03:12.09`	`elriah`	`When using autologoff in agents.conf, if strategy is ringall, and there are 10 agents logged in, and nobody answers, and the autologff time lapses, will everybody be logged out?`
`03:13.20`	`LanceSnyder`	`i dont believe so`
`03:13.43`	`LanceSnyder`	`they should be logged out until they want to be logged out`
`03:13.47`	`LanceSnyder`	`shouldn't*`
`03:14.02`	`elriah`	`I'm using AgentCallBackLogin.`
`03:14.12`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`03:14.55`	`k-man_`	`[TK]D-Fender, i'm trying to compile the zaptel dummy driver to get the 1khz clock thing happening but i get a compile error`
`03:15.23`	`LanceSnyder`	`elriah, i believe im wrong`
`03:16.06`	`LanceSnyder`	`elriah, after the agent's max ring time is up, the agent is deemed unavailable`
`03:16.08`	`LanceSnyder`	`and logged off`
`03:16.20`	`LanceSnyder`	`so i was wrong.`
`03:16.48`	`elriah`	`But if I'm using ringall, and all agents are being called, and the timeout is exceeded, will they all be logged off?`
`03:16.57`	`LanceSnyder`	`yes`
`03:17.04`	`LanceSnyder`	`they will all be deemed unavailable`
`03:17.13`	`JT`	`k-man_: i already asked, but... what versions?`
`03:17.53`	`k-man_`	`jt, actually, it was kernel 2.6.16.20 but i just upgraded and i realised its still finding the old kernel source`
`03:18.02`	`k-man_`	`jt, just reconfiguring.. .standby`
`03:18.07`	`MooingLemur`	`darn electrostatic cat resetting my GPX-2000 with his tail.`
`03:18.14`	`k-man_`	`or just go about your business until i get stuck again ;)`
`03:19.04`	`MooingLemur`	`GXP, rather`
`03:19.20`	`elriah`	`Thanks.`
`03:19.29`	`*** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net)`
`03:19.43`	`LanceSnyder`	`elriah, there may be ways to stop it`
`03:19.47`	`LanceSnyder`	`but i've not ever used it`
`03:19.48`	`LanceSnyder`	`so i dont know`
`03:20.03`	`elriah`	`I'm going to write an AgentLogoff() app.`
`03:20.18`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`03:20.18`	`*** mode/#asterisk [+o mog] by ChanServ`
`03:34.28`	`nop45`	`tried to enable testfeature with: DYNAMIC_FEATURES=hangup#blindxfer#testfeature#atxfer#automon I am still missing something to get it to go.`
`03:36.23`	`*** join/#asterisk foxxtrot (n=craig@c-67-185-241-244.hsd1.wa.comcast.net)`
`03:45.56`	`*** join/#asterisk Deeewayne (n=dwayne@68.62.166.235)`
`03:46.50`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`03:46.50`	`*** mode/#asterisk [+o russellb] by ChanServ`
`03:48.51`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`03:48.51`	`*** mode/#asterisk [+o russellb] by ChanServ`
`03:50.32`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`03:50.50`	`*** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com)`
`03:53.42`	`*** join/#asterisk bmg505 (n=leon@c1-227-8.rndf.isadsl.co.za)`
`04:03.39`	`*** join/#asterisk topping (n=topping@204.152.96.238)`
`04:07.26`	`mrc3__`	`LanceSnyder, JT, eald_home, yesterday you were helping me to get ekiga/twinkle to talk to asterisk. well, they now work: i had a stun server configured from previous attempts. removing it did the trick!`
`04:07.58`	`JT`	`yeah i thought so, it clearly stated the STUN server wasn't working :P`
`04:08.10`	`sfbosch`	`I have a voice mail question. Anybody game?`
`04:08.30`	`mrc3__`	`thanks for the help!`
`04:11.03`	`nop45`	`WHEN do I use testfeature (#9)? during a conversation or after "flashing" ?`
`04:11.53`	`k-man_`	`jt, fyi, once i fixed the link the compile of zaptel worked`
`04:12.01`	`JT`	`ok`
`04:12.08`	`JT`	`1.2.x?`
`04:12.09`	`*** join/#asterisk foxxtrot (n=craig@c-67-185-241-244.hsd1.wa.comcast.net)`
`04:12.20`	`k-man_`	`jt, also fyi, apparantly the jitter problem can be because if no 1khz clock.. hence the need for the zaptel dummy driver`
`04:12.24`	`k-man_`	`jt, 1.4`
`04:13.13`	`JT`	`right, and if you want to get real technical, you could compile your kernel with HZ of 1000Hz`
`04:13.21`	`JT`	`haven't heard of sip needing zaptel`
`04:13.25`	`JT`	`sounds strange`
`04:13.32`	`JT`	`why aren't you going 1.2?`
`04:14.04`	`jpalmer`	`JT: asterisk depends on zaptel for timing. why they don't just move to POSIX timers.. I don't know.. but thats another question`
`04:14.19`	`k-man_`	`jt... dunno`
`04:14.33`	`mosty`	`sfbosch: don't ask to ask. just ask.`
`04:14.37`	`JT`	`jpalmer: zaptel interfaces, IAX trunking, and MeetMe relies on zaptel`
`04:14.44`	`k-man_`	`jt, i guess someone has to test 1.4 or it will never proceed.. and i'm not in production so its not hard for me to test`
`04:15.20`	`JT`	`jpalmer: nothing else should require zap`
`04:15.20`	`JT`	`oh, i thought you wanted to fix the problems, k-man_ :P`
`04:18.16`	`k-man_`	`jt... well.. at least help in the debugging of them`
`04:21.18`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`04:21.50`	`sfbosch`	`Okay -- voice mail problem:`
`04:21.56`	`*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)`
`04:22.01`	`sfbosch`	`Callers do not hear the "unavailable" message.`
`04:22.08`	`sfbosch`	`any idea why?`
`04:22.35`	`mosty`	`did you specify the "play unavailable" message option to the voicemail command?`
`04:23.00`	`sfbosch`	`As in: exten => 2211,1,Dial(SIP/211,10)`
`04:23.00`	`sfbosch`	`exten => 2211,2,VoiceMail(u211@default)`
`04:23.01`	`sfbosch`	`exten => 2211,3,Hangup`
`04:23.20`	`sfbosch`	`This looks right: exten => 2211,2,VoiceMail(u211@default)`
`04:23.46`	`sfbosch`	`CLI output shows that Asterisk is trying to play something`
`04:24.08`	`sfbosch`	`<PROTECTED>`
`04:24.17`	`sfbosch`	`but caller hears only a loud click`
`04:24.19`	`sfbosch`	`or scraping sound`
`04:24.41`	`mosty`	`does that file exist? is it readable by asterisk? can you test the sound with a sound playing app?`
`04:25.06`	`sfbosch`	`the sound file is unavail.wav, and it is in /var/spool/asterisk/voicemail/default/211`
`04:25.24`	`sfbosch`	`the directory "unavail" does not exist until the first time you call voice mail -- then something is creating it`
`04:25.40`	`sfbosch`	`and the sound was created using VoiceMailMain()`
`04:25.45`	`*** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net)`
`04:25.47`	`xo8ox`	`hey guys`
`04:26.02`	`mosty`	`sfbosch: what codec is the call using?`
`04:26.14`	`LanceSnyder`	`sfbosch, ls /var/spool/asterisk/voicemail/default/211 \| grep unavail`
`04:26.49`	`xo8ox`	`guys howcome everytime I call any extention in the network it says that exten is not available ?`
`04:26.58`	`sfbosch`	`asterisk1 211 # ls \| grep unavail`
`04:26.59`	`sfbosch`	`unavail`
`04:26.59`	`sfbosch`	`unavail.wav`
`04:27.21`	`sfbosch`	`I don't actually know what codec the call is using -- it's defaults`
`04:27.41`	`mosty`	`sfbosch: do you have g729 licences, and is the caller using g729?`
`04:27.58`	`JerJer`	`xo8ox: maybe because the extension is not available ???`
`04:27.59`	`sfbosch`	`caller is a SIP extension, a Polycom 501`
`04:28.27`	`LanceSnyder`	`sip show peer extension@context`
`04:28.28`	`LanceSnyder`	`try that`
`04:28.44`	`LanceSnyder`	`err... sip show peer extension`
`04:28.52`	`orlock`	`Is there echo cancellation in asterisk?`
`04:28.54`	`LanceSnyder`	`replace extension with the extension`
`04:29.26`	`*** join/#asterisk x86 (n=x86@p3m/member/x86)`
`04:29.27`	`xo8ox`	`it is`
`04:29.41`	`mosty`	`orlock: there is for zaptel and misdn devices, but not for voip channels as far as i know`
`04:30.06`	`xo8ox`	`from that extens phone I can call outside or go to voice mail etc but from exten to exten the system keeps saying its not available`
`04:30.14`	`orlock`	`goddamn`
`04:30.15`	`sfbosch`	`LanceSnyder, mosty: Codecs are gsm, ulaw, alaw, h263`
`04:30.33`	`mosty`	`sfbosch: can you confirm that unavail.wav is ok? play it in audacity or something`
`04:30.35`	`LanceSnyder`	`in that order?`
`04:31.16`	`LanceSnyder`	`asterisk -rx 'sip show peer $EXTENSION'`
`04:31.20`	`LanceSnyder`	`tell me what you see there`
`04:31.29`	`LanceSnyder`	`this will tell you if the device is even registered with the server`
`04:31.31`	`sfbosch`	`LanceSnyder: hang on`
`04:31.45`	`xo8ox`	`I get this`
`04:31.46`	`sfbosch`	`LanceSnyder: I am seeing registration messages`
`04:31.48`	`xo8ox`	`[Feb 27 21:30:18] WARNING[22117]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)`
`04:31.59`	`LanceSnyder`	`it's in your dialplan`
`04:32.01`	`LanceSnyder`	`extensions.conf`
`04:32.21`	`LanceSnyder`	`check your spelling and syntax`
`04:32.29`	`xo8ox`	`me ?`
`04:34.32`	`xo8ox`	`guys whats this warning .. WARNING[22138]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)`
`04:34.43`	`*** join/#asterisk bigred (n=ian@75-1-209-228.lightspeed.snantx.sbcglobal.net)`
`04:36.03`	`JunK-Y`	`the message isnt pretty clear?`
`04:36.57`	`k-man_`	`is there some way i can get asterisk to report the frequency of the clock it is getting?`
`04:37.05`	`k-man_`	`or to test that the clock i have set up is working?`
`04:37.32`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`04:39.20`	`*** join/#asterisk bkw_ (n=brian@dsl093-079-130.sfo1.dsl.speakeasy.net)`
`04:42.48`	`k-man_`	`hmm`
`04:42.49`	`xo8ox`	`anybody here can help me debug this issue ?`
`04:43.07`	`k-man_`	`zttest returns very variable results... is that normal for ztdummy?`
`04:43.42`	`*** join/#asterisk angom_h (n=Angel@red-corp-201.143.196.183.telnor.net)`
`04:44.17`	`pigpen`	`xo8ox, sounds like your peer is not associated or setup correctly...check out your sip.conf`
`04:44.37`	`pigpen`	`k-man_, depends....what variable results are you referring to.`
`04:44.55`	`k-man_`	`Best: 99.963379 -- Worst: -136.853027 -- Average: -52.220962`
`04:45.05`	`bigred`	`I got a fresh install of asterisk, followed the vitelity instructions, but when i start up with the -vvvc, the CLI prompt continually loops and I have to ctrl-c the process to kill it`
`04:45.19`	`bigred`	`anyone know why?`
`04:45.43`	`pigpen`	`bigred, check out your log files for answers.`
`04:46.40`	`pigpen`	`k-man_, my thought would be this. It is a dummy interface...who cares?`
`04:46.53`	`k-man_`	`pigpen, because i need it for the 1khz timer`
`04:47.01`	`k-man_`	`to solve this jitter problem`
`04:47.28`	`xo8ox`	`pigpen: when I dial from my cisco phone to my soft phone it works.. but when I call from or to the polycom phones that we have it says extenssion is not available`
`04:47.31`	`k-man_`	`is there a decent searchable archive of the asterisk-users mailing list somewhere?`
`04:47.32`	`xo8ox`	`I don't get it`
`04:47.48`	`pigpen`	`Well, I have cards in all mine, from one port to 4 pri's.....I have no need for ztdummy.`
`04:47.51`	`sfbosch`	`mosty: The file plays as the scraping sound`
`04:48.22`	`*** join/#asterisk foobar778 (i=johhny@ip68-100-210-15.dc.dc.cox.net)`
`04:48.26`	`pigpen`	`xo8ox, my guess is your polycom's are not setup right.`
`04:48.32`	`sfbosch`	`mosty: so I guess my question is -- why is the format wrong?`
`04:48.49`	`xo8ox`	`damn polycoms hehe`
`04:48.51`	`foobar778`	`[TK]D-Fender: are u there`
`04:49.02`	`*** join/#asterisk Strom_M (n=strom@209.19.56.4)`
`04:49.05`	`sfbosch`	`even weirder is that I can record the message using VoiceMailMail(), then replay them`
`04:49.09`	`*** join/#asterisk pardove (n=chatzill@195.146.47.143)`
`04:49.31`	`pardove`	`is soft-switch.org UP?`
`04:49.35`	`sfbosch`	`but when somebody actually calls the voicemail, scraping noise instead`
`04:49.38`	`*** join/#asterisk bkw_ (n=brian@dsl093-079-162.sfo1.dsl.speakeasy.net)`
`04:50.58`	`*** join/#asterisk LanceMSnyder (n=LanceSny@adsl-10-1-235.mia.bellsouth.net)`
`04:52.01`	`pardove`	`what's happened to soft-switch.org?`
`04:52.03`	`*** join/#asterisk angom_w (n=Angel@red-corp-201.143.196.183.telnor.net)`
`04:52.03`	`foobar778`	`I have a question I have a did numvber and after a certain time and is unanswerd it goes to DISA allowing the outside user to make voip calls bridging the inbound call to the outside provider but when the inbound call jangsup the longdistance outbound calls continues How to make that call terminate aswell`
`04:52.25`	`nop45`	`anyone working with testfeature command ?`
`04:52.40`	`orlock`	`What exactly does zttest test?`
`04:53.09`	`foobar778`	`I have a question I have a did numvber and after a certain time and is unanswerd it goes to DISA allowing the outside user to make voip calls bridging the inbound call to the outside provider but when the inbound call hangsup the longdistance outbound calls continues How to make that call terminate aswell Spelling corrected`
`04:53.16`	`mosty`	`sfbosch: try watching the asterisk console when you record the file, see if there are any errors`
`04:54.59`	`sfbosch`	`mosty: I set "format=wav" in the general context of voicemail.conf: it works now!`
`04:57.32`	`robby____`	`quit`
`04:59.10`	`pigpen`	`I am trying to replace a IAX trunk (don't ask why...well...audio problems). I am attempting to setup a sip peer/user in it's place.`
`04:59.19`	`pigpen`	`I can call from the remote asterisk just fine.`
`04:59.33`	`pigpen`	`but inbound it bitches...that the remote doesn't exist.`
`04:59.34`	`pigpen`	`http://pastebin.ca/375295`
`04:59.53`	`pigpen`	`Now, the first thing you would think is "set up the peer/user in reverse..."`
`05:00.21`	`pigpen`	`yeah...no. Neither side will auth. Fails pathetically.`
`05:00.37`	`Juggie`	`pigpen, did you consider setting up a iax trunk w/o trunking first`
`05:00.59`	`Juggie`	`i think solving your 'audio problems' would be much better`
`05:01.02`	`pigpen`	`iax trunk with out trunking`
`05:01.11`	`pigpen`	`Juggie, I have been fighting it for 6 months.`
`05:01.21`	`pigpen`	`But if you want to take a shot I am game.`
`05:01.42`	`Juggie`	`between what versions of *?`
`05:02.49`	`pigpen`	`1.2.11 > 1.2.12.1`
`05:02.57`	`pigpen`	`1.2.9.1 > 1.2.12.1`
`05:03.05`	`pigpen`	`1.4.0 > 1.2.12.1`
`05:03.13`	`pigpen`	`1.2.14 > 1.2.12.1`
`05:03.23`	`Juggie`	`have you tried w/o 1.2.12.1 in the mix?`
`05:03.34`	`pigpen`	`Well, 1.2.12 is the head.`
`05:03.48`	`Juggie`	`have you considered that version may have the problem? :)`
`05:04.36`	`pigpen`	`Well, I have spoken to many that have noted that iax can have voice chopping....`
`05:04.39`	`*** join/#asterisk bigred (n=ian@75-1-209-228.lightspeed.snantx.sbcglobal.net)`
`05:04.48`	`pigpen`	`moving to sip resolved the issue.`
`05:04.56`	`Juggie`	`under how much load?`
`05:05.15`	`pigpen`	`Varies...1 - 2 calls up to 10-15`
`05:05.25`	`Juggie`	`that doesnt make sense`
`05:05.25`	`pigpen`	`Calls over a 40MB fiber connection.`
`05:05.29`	`pigpen`	`yeah.`
`05:05.34`	`pigpen`	`welcome to my world.`
`05:05.39`	`Strom_M`	`what kind of latency on the fiber?`
`05:05.43`	`Juggie`	`so, the first thing i would do is change the version of * on the head.`
`05:05.45`	`pigpen`	`1ms.`
`05:05.52`	`Juggie`	`or try iax w/ trunk=yes/no etc.`
`05:05.57`	`LanceSnyder`	`ohhhhhhhhhh fiber`
`05:05.58`	`bigred`	`is 1.4 the latest stable version of asterisk?`
`05:06.02`	`LanceSnyder`	`<drooling>`
`05:06.30`	`pigpen`	`that fiber line is the skinny side...the other side we have 400mb....(uplink)`
`05:06.38`	`*** join/#asterisk dfsexor (n=ircap8@72-140-231-201.fibertel.com.ar)`
`05:06.41`	`pigpen`	`So what do you think of 1.4 at the head?`
`05:06.48`	`Juggie`	`eugh, no, not yet`
`05:06.54`	`Juggie`	`not unless you are ok w/ 1.4svn`
`05:06.55`	`LanceSnyder`	`<even more drooling>`
`05:07.02`	`pigpen`	`I have been working on moving to 1.4 to take advatage of RTA with postgres (which I have working I might add)`
`05:07.16`	`Juggie`	`if you use 1.4svn more power to you, i would not use 1.4.0 however`
`05:07.31`	`bigred`	`what's wrong with 1.4?`
`05:07.32`	`pigpen`	`Yeah..that is why I haven't deployed yet.`
`05:07.59`	`Juggie`	`nothing in perticular just that the point release of 1.4.0 has a million different little bugs that you'll complain about, only to discover they are already fixed in svn`
`05:08.01`	`pigpen`	`bigred, well, some things are not quite...there, documented, just right, etc...`
`05:08.19`	`bigred`	`pigpen: what are you getting to work w/postgres then?`
`05:08.33`	`pigpen`	`So, 1.2.14, get all up to 1.2.14 try again.`
`05:08.44`	`dfsexor`	`hello everybody I just installed AsteriskWin32 and I continous hear a woman telling me the 1000 box speach. How I can disable this? (pls excuse my english)`
`05:08.48`	`pigpen`	`if it still borks the voice...then I will try sip.`
`05:08.48`	`Juggie`	`yes`
`05:08.57`	`Juggie`	`upgrade both your boxes to latest and try.`
`05:08.59`	`pigpen`	`but why change trunk=yes/no?`
`05:09.09`	`Juggie`	`because it could be a bug in trunking`
`05:09.13`	`Strom_M`	`latest is 1.2.15 :)`
`05:09.16`	`antlers`	`not an Asterisk question per-se, but anyone have an opinion on OS X softphones?`
`05:09.28`	`pigpen`	`will it still work as a trunk?`
`05:09.37`	`Juggie`	`yes, it will just not conserve bandwidth`
`05:09.38`	`pigpen`	`antlers, idefisk.`
`05:09.47`	`pigpen`	`Juggie, got plenty of that.`
`05:10.04`	`Juggie`	`pigpen, all trunk=yes does is share the packet`
`05:10.06`	`bigred`	`so should i run trunk or 1.2.15?`
`05:10.07`	`pigpen`	`bigred, RTA (real time exten, voicemail, cdr, etc...`
`05:10.09`	`antlers`	`pigpen, danke`
`05:10.34`	`Juggie`	`so if i have 10 calls up, instead of sending 10 headers w/ 20ms of audio it will send 1 header w/ 20x10ms of audio`
`05:10.38`	`pigpen`	`Juggie, good idea....`
`05:10.39`	`Juggie`	`or whatever, something like that`
`05:11.11`	`Juggie`	`but try it on or off`
`05:11.20`	`Juggie`	`whatever the oposite is of the way you have it now`
`05:11.28`	`bigred`	`pigpen: voicemail stored in the db?`
`05:11.34`	`SplasPood`	`Qwell[]: ah! I seem to be having a Dell 2950 irq sharing + zaptel problem`
`05:11.48`	`pigpen`	`bigred, voicemail.`
`05:11.57`	`pigpen`	`well..the settings.`
`05:12.01`	`bigred`	`pigpen: but the actual sound files?`
`05:12.02`	`bigred`	`ah`
`05:12.06`	`pigpen`	`vm is still in the file system.`
`05:12.15`	`pigpen`	`yeah..sorry.`
`05:12.27`	`bigred`	`you dont happen to have vitelity sip setup, do you?`
`05:12.30`	`pigpen`	`but I love having my dialplan in postgres...much nicer.`
`05:12.42`	`pigpen`	`that too.`
`05:12.56`	`*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)`
`05:13.02`	`pigpen`	`working nicely with my polycom 500,501,600,650`
`05:13.14`	`antlers`	`pigpen, what about a SIP softphone... any suggestions there?`
`05:13.18`	`pigpen`	`well...define vitelity...`
`05:13.30`	`pigpen`	`antlers, I think xten has one...`
`05:13.31`	`dfsexor`	`hello everybody I just installed AsteriskWin32 and I continous hear a woman telling me the 1000 box speach. How I can disable this? (pls excuse my english)`
`05:13.33`	`bigred`	`uhm. vitelity pay as you go?`
`05:13.39`	`pigpen`	`no.`
`05:13.46`	`bigred`	`what do you use then/`
`05:14.08`	`pigpen`	`use what? pstn?`
`05:14.38`	`wunderkin`	`so.. is it ok not to have any kind of directory file on a polycom?`
`05:14.39`	`pigpen`	`Juggie, thanks bty...`
`05:14.59`	`pigpen`	`wunderkin, sure..you just won't have a directory file.`
`05:15.48`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`05:15.53`	`blitzrage`	`xheliox: I think your bug got fixed`
`05:15.55`	`wunderkin`	`not too sure if this is going to fix anything... i dont think that was the problem... i bet we will be back to having phone reboots.. again .. :/`
`05:16.07`	`foobar778`	`I have a question I have a did numvber and after a certain time and is unanswerd it goes to DISA allowing the outside user to make voip calls bridging the inbound call to the outside provider but when the inbound call jangsup the longdistance outbound calls continues How to make that call terminate aswell`
`05:17.32`	`pigpen`	`foobar778, not seeing your dialplan, you may want to insert a hangup in your dialplan...somewhere...`
`05:17.50`	`blitzrage`	`it should really drop the channel automatically`
`05:17.52`	`foobar778`	`pigpen I have placed a hangup`
`05:18.06`	`blitzrage`	`I'd be interested in seeing your dialplan to see how you exactly accomplished that happening`
`05:18.11`	`pigpen`	`true...but who knows...`
`05:18.14`	`pigpen`	`yeah.`
`05:18.15`	`foobar778`	`is there a delay`
`05:18.47`	`foobar778`	`<PROTECTED>`
`05:18.47`	`blitzrage`	`the whole situation sounds a bit odd to me`
`05:18.59`	`*** part/#asterisk dfsexor (n=ircap8@72-140-231-201.fibertel.com.ar)`
`05:18.59`	`foobar778`	`how so blitz?`
`05:19.13`	`blitzrage`	`I'm confused as to what you're actually trying to do... (it is late here too)`
`05:19.16`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`05:19.27`	`foobar778`	`Ill get dial plan`
`05:19.34`	`blitzrage`	`it goes to DISA after not being answered? odd`
`05:19.46`	`blitzrage`	`foobar778: yah -- dialplan will help a lot. PUt in a pastebin`
`05:20.13`	`foobar778`	`pastebin link`
`05:20.43`	`blitzrage`	`~pb`
`05:20.48`	`jbot`	`[pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`05:23.42`	`foobar778`	`http://channels.debian.net/paste/5528.`
`05:25.29`	`foobar778`	`did the pastebin work?/`
`05:26.44`	`*** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com)`
`05:27.45`	`pigpen`	`I would say no..use the first one.`
`05:28.01`	`foobar778`	`me pigpen?/`
`05:28.37`	`foobar778`	`http://channels.debian.net/paste/5528`
`05:28.57`	`foobar778`	`works here`
`05:31.10`	`pigpen`	`ah..now it is there.`
`05:31.10`	`foobar778`	`????`
`05:31.20`	`foobar778`	`ol`
`05:31.24`	`foobar778`	`ok`
`05:32.26`	`pigpen`	`which did are you having issues with?`
`05:33.27`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`05:34.18`	`*** join/#asterisk sharp (n=sharp@2001:470:1f01:ffff:0:0:0:1c23)`
`05:34.44`	`foobar778`	`1703incoming`
`05:34.59`	`foobar778`	`when it goes to disa`
`05:35.10`	`foobar778`	`Il dial out_99`
`05:35.17`	`foobar778`	`then hangup`
`05:35.25`	`foobar778`	`seems after disda`
`05:35.28`	`foobar778`	`disa`
`05:35.34`	`foobar778`	`call continues`
`05:36.48`	`pigpen`	`pretty straight forward...it should hang up.`
`05:37.04`	`foobar778`	`well u would think`
`05:37.37`	`foobar778`	`but it doesnt thie issue is this it will increse charges`
`05:38.18`	`pigpen`	`Well, first, I would clean up your dialplan...`
`05:38.25`	`pigpen`	`get rid of all the extra crap..`
`05:38.33`	`pigpen`	`I am still trying to find out_99`
`05:39.26`	`foobar778`	`outgoing`
`05:39.40`	`foobar778`	`-99`
`05:39.58`	`foobar778`	`exten => _99.,1,Dial(SIP/${EXTEN:2}@tom88940,60)`
`05:40.20`	`pigpen`	`you may try putting a exten => _99.,Hangup`
`05:40.23`	`pigpen`	`after that line.`
`05:40.46`	`pigpen`	`that's what I would try anyway.`
`05:41.00`	`foobar778`	`yea`
`05:41.10`	`foobar778`	`will fdo`
`05:41.15`	`foobar778`	`will do`
`05:41.17`	`pigpen`	`otherwise it is left open ended...`
`05:42.50`	`foobar778`	`actually I had done that`
`05:43.16`	`foobar778`	`I pasted a previous extensions.conf`
`05:43.34`	`pigpen`	`submit a bug?`
`05:43.37`	`foobar778`	`<PROTECTED>`
`05:43.54`	`pigpen`	`you may want to ask tomorrow as many are sleeping...as I should...`
`05:44.48`	`foobar778`	`yes seems once it gets into Disa and u hangup the line u caled on the Disa line is still going`
`05:45.08`	`foobar778`	`and the calls fron disa continue`
`05:46.15`	`pigpen`	`you may try setting up "#" as a hangup in the features.conf...`
`05:46.15`	`pigpen`	`<PROTECTED>`
`05:52.36`	`*** join/#asterisk zeeesh (i=zeeesh@202.38.55.125)`
`05:52.37`	`zeeesh`	`hi`
`05:53.10`	`*** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)`
`06:01.26`	`k-man_`	`jt, have you seen openvoice.com.au?`
`06:01.31`	`k-man_`	`jt, free sounds for asterisk`
`06:02.41`	`JT`	`yes, i thought we already discussed them`
`06:02.46`	`JT`	`i didn't like them`
`06:04.22`	`k-man_`	`jt, oh... i don't recall discussing them`
`06:04.22`	`k-man_`	`jt, what don't you like about them? isn't it better than the american voice`
`06:05.53`	`k-man_`	`jt, what about the british sounds? are any of them any good?`
`06:06.51`	`JT`	`it sounds crap, it doesn't sound neutral, it's male, and the guy is a radio announcer and it sounds like he's doing a radio announcement`
`06:07.21`	`JT`	`pigpen: so let me get this straight... you tried 10 billion versions of asterisk but didn't try trunk=no?`
`06:08.41`	`k-man_`	`jt, yeah it does sound kind of commercial radio station ish`
`06:09.06`	`JT`	`they're not very good`
`06:09.17`	`JT`	`i can make better ones for a phone system, given a recording studio`
`06:09.37`	`k-man_`	`jt, i think we should start a fund`
`06:09.43`	`k-man_`	`to get some good sounds recorded`
`06:09.53`	`JT`	`try the other ones first`
`06:09.53`	`k-man_`	`we could hire a studio or something`
`06:10.00`	`k-man_`	`which other ones?`
`06:10.16`	`JT`	`there's at least 2 sets of australian asterisk prompts`
`06:10.22`	`k-man_`	`is there? hmm`
`06:10.28`	`JT`	`dunno if the other ones are at all available for free`
`06:11.37`	`JT`	`http://www.voipshop.com.au/product_info.php?products_id=72`
`06:12.10`	`k-man_`	`jt, ah yes... free for non commercial use`
`06:12.17`	`xo8ox`	`guys when I dial a que I do go on hold with music on hold but none of the que lines ring ?`
`06:12.23`	`xo8ox`	`do agents have to logg in ?`
`06:12.44`	`xo8ox`	`for their phone to ring ?`
`06:14.08`	`xo8ox`	`anyone ?`
`06:14.30`	`JT`	`wow, waiting a whole 2 minutes, bravo`
`06:20.45`	`k-man_`	`is there some way i can listen to gsm files under windows?`
`06:20.49`	`*** join/#asterisk masked (i=masked@shell.iinet.net.au)`
`06:20.54`	`masked`	`hi ho`
`06:21.29`	`jpalmer`	`s/me/my/`
`06:21.31`	`xo8ox`	`change the .gsm to wav`
`06:21.32`	`xo8ox`	`:P`
`06:21.36`	`masked`	`i get these errors when loading ztdummy, http://www.pastebin.ca/375339 i think maybe it doesn't like my openMosix kernel, can someone confirm please?`
`06:22.14`	`*** join/#asterisk bkw_ (n=brian@72-254-46-103.client.stsn.net)`
`06:23.06`	`*** join/#asterisk phpboy (n=shane@196.211.17.202)`
`06:23.36`	`JT`	`might be compiled against the wrong kernel tree`
`06:23.42`	`JT`	`masked: what do you need ztdummy for?`
`06:24.14`	`phpboy`	`hey all, I changed my recordings log DIR in agents.conf... but it doesn't seem to be set correctly, can anybody give me any advice on this?`
`06:24.14`	`masked`	`JT: meetme`
`06:24.30`	`JT`	`you could use app_conference instead`
`06:24.44`	`masked`	`JT: i do have an one of those motorola cards around that do real timing`
`06:25.31`	`JT`	`motorola cards?...`
`06:25.35`	`masked`	`JT: i actually just did a fresh install with the asterisk-gui, and make a conference, i only guessed that used meetme`
`06:25.38`	`masked`	`uhm`
`06:25.43`	`masked`	`iono, they are old dialup modems`
`06:25.44`	`JT`	`haha asterisk-gui`
`06:25.55`	`JT`	`why the hell would that provide zap timing?`
`06:25.56`	`masked`	`with zaptel drivers`
`06:26.06`	`JT`	`it's not motorola`
`06:26.07`	`masked`	`and you can use them for pstn`
`06:26.21`	`masked`	`but they are crap here cos of the impedence on the lines`
`06:26.28`	`masked`	`umm... thought it was..`
`06:26.34`	`JT`	`intel chipset`
`06:26.38`	`JT`	`x100p`
`06:26.47`	`JT`	`it's no longer made at all`
`06:26.53`	`masked`	`yeah x100p, has a motorola chip on it though.`
`06:27.08`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`06:27.12`	`JT`	`yeah they're pretty crap`
`06:27.52`	`masked`	`hmm, i just want a conference app for the car club`
`06:28.01`	`JT`	`anyway, use real asterisk and you can make your own conferences with app_conference`
`06:28.28`	`masked`	`yeah, i've done that in the past, i thought the gui might be mature enough to do it for me`
`06:28.37`	`JT`	`it's super alpha`
`06:29.23`	`masked`	`k`
`06:29.42`	`masked`	`hmm`
`06:30.00`	`masked`	`well i'll guess that it's the gui that is giving me problems then`
`06:30.13`	`JT`	`well meetme still needs zap timing`
`06:30.16`	`JT`	`no idea why`
`06:30.17`	`JT`	`but it does`
`06:30.39`	`masked`	`if i call the box via a voip gw, the itsp tells me i'm on the phone`
`06:30.50`	`masked`	`asterisk says the call has been rejected.`
`06:31.03`	`masked`	`i thought it was for syncrony`
`06:31.07`	`masked`	`+h`
`06:31.32`	`JT`	`sure, there's no technical reason why it's required though`
`06:31.38`	`JT`	`just one of those strange quirks`
`06:31.39`	`*** join/#asterisk dennisharrison (n=dennisha@68-114-106-133.dhcp.slid.la.charter.com)`
`06:31.44`	`masked`	`beyond me.`
`06:33.17`	`dennisharrison`	`murderer ....`
`06:33.54`	`masked`	`ok, fresh install`
`06:33.59`	`masked`	`i'll try again`
`06:34.08`	`dennisharrison`	`well hope you get it right this time`
`06:37.15`	`*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)`
`06:40.56`	`xo8ox`	`guys someone plz help`
`06:41.22`	`xo8ox`	`its midnight and I'm still at work trying to fix this problem`
`06:45.10`	`k-man_`	`how do i tell asterisk to use the sound prompts in au instead of the default?`
`06:48.36`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`06:48.45`	`phpboy`	`hey all, I changed my recordings log DIR in agents.conf... but it doesn't seem to be set correctly, can anybody give me any advice on this?`
`06:56.13`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`07:05.25`	`yonahw-work`	`i am trying to convert a number so that it is in the format that my provider will accept it and am having some trouble here`
`07:05.36`	`yonahw-work`	`can someone take a look at http://pastebin.ca/375355 and tell me what i am doing wrong here`
`07:13.44`	`*** part/#asterisk sfbosch (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com)`
`07:24.14`	`*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)`
`07:29.48`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`07:43.53`	`*** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud)`
`07:44.38`	`*** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr)`
`07:57.54`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`07:58.47`	`*** join/#asterisk topping (n=topping@204.152.96.238)`
`08:01.06`	`*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)`
`08:02.16`	`Mahmoud`	`regarding dial plans, are periorities just steps of execution, which one comes 1st?`
`08:03.04`	`Mahmoud`	`similar to firewall rules, we assign them with numbers, which means which one is looked up 1st`
`08:03.12`	`EmleyMoor`	`Priorities are steps of execution`
`08:03.24`	`Mahmoud`	`so my understanding is right =) thanks`
`08:03.30`	`EmleyMoor`	`1 comes first, and there must always be a 1`
`08:03.52`	`EmleyMoor`	`You can number, or use n, thereafter, and use labels and offsets from labels`
`08:03.56`	`Mahmoud`	`hmmm why didn't they just do it without priorities being mentioned, just execution the 1st line 1st`
`08:04.05`	`Mahmoud`	`EmleyMoor, yeah, n makes it easier`
`08:04.36`	`Mahmoud`	`what would happen if two lines of commands regarding same extention had the same priority`
`08:04.39`	`EmleyMoor`	`Because some apps jump to n+100 automatically in certain conditions`
`08:04.49`	`Mahmoud`	`i see`
`08:05.06`	`EmleyMoor`	`Probably, it would stop`
`08:05.17`	`yonahw-work`	`is there a function to append variable2 to variable1?`
`08:06.02`	`EmleyMoor`	`yonahw-work: Set(variable1=${variable1}${variable2})`
`08:07.04`	`yonahw-work`	`emelymoor: can i do set(variable3=${variable1}${variable2})?`
`08:07.12`	`EmleyMoor`	`Yes`
`08:08.17`	`yonahw-work`	`hmm thought i tried that but apparently not since it now works`
`08:11.36`	`Mavvie`	`Is there a way to get more output from the RTP streams? For example, the amount of traffic pushed through them, jitter etc?`
`08:11.49`	`SheriF_SpacE`	`Mahmoud: oh god at last someone else from arabic country :-d`
`08:12.12`	`Mahmoud`	`from egypt?`
`08:16.29`	`Makenshi`	`Tenders for Tier-1 Registry for 4.4.e164.arpa are due in just under 4 hours`
`08:18.17`	`tzafrir_laptop`	`yonahw-work, please pastebin the relevant snippet from your dialplan and fron a CLI trace`
`08:18.57`	`tzafrir_laptop`	`Mahmoud, ask whois`
`08:19.17`	`Mahmoud`	`tzafrir_laptop, whois can't tell nationality`
`08:19.31`	`Makenshi`	`Whois is useless`
`08:20.19`	`*** join/#asterisk dorel__ (n=liran@80.179.31.43.static.012.net.il)`
`08:21.04`	`tzafrir_laptop`	`I actually mean a whois on the IP address, but then again, not helpful for both of you, for different reasons`
`08:21.21`	`Makenshi`	`oh well on ip it's fairly useless`
`08:21.39`	`Makenshi`	`i've never had a reply to an email i've sent to an contact listed on ripe/arin/apnic`
`08:22.22`	`Makenshi`	`at least you can find out the upstream provider`
`08:22.28`	`Makenshi`	`sorry, i'm being an arse`
`08:22.42`	`Makenshi`	`it's very early here`
`08:29.59`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`08:34.25`	`Mahmoud`	`do i need to load certain modules in order for asterisk to listen on a UDP port`
`08:34.46`	`Mahmoud`	`freebsd's sockstat shows "root asterisk 783 3 stream /var/run/asterisk.ctl`
`08:35.10`	`*** join/#asterisk x86 (n=x86@p3m/member/x86)`
`08:35.12`	`Mahmoud`	`not listening to any specific port, and when i try to connect to it, it replays with an ICMP port not available packet`
`08:40.55`	`*** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com)`
`08:43.44`	`*** join/#asterisk yxa (n=lonari@58.185.90.101)`
`08:44.20`	`yxa`	`are there any other service provider like IPKall?`
`08:45.40`	`Mahmoud`	`hmmm using asterisk we can make a systematic prostitute phone`
`08:45.52`	`Mahmoud`	`"pres more.. more"`
`08:46.14`	`Mahmoud`	`s/pres/press/`
`08:46.33`	`Mahmoud`	`amazing bot o.O`
`08:47.02`	`*** join/#asterisk [shodan] (n=shodan@ip083.96-113-216.pppoe1.joliette.intermonde.net)`
`08:47.51`	`Mahmoud`	`s/$am/da/`
`08:48.21`	`Mahmoud`	`doesn't fully support regexp =]`
`08:54.25`	`*** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com)`
`08:57.15`	`*** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com)`
`08:58.30`	`*** join/#asterisk sudhir492 (n=sudhir@c-71-63-59-45.hsd1.va.comcast.net)`
`08:58.36`	`sudhir492`	`Hi All.`
`09:00.04`	`*** join/#asterisk shadebob (n=chatzill@adll-219-245-206-196.marocconnect.net.ma)`
`09:08.09`	`*** join/#asterisk RoyK (n=roy@217-175-35.100710.adsl.tele2.no)`
`09:09.23`	`*** join/#asterisk vlt (n=dm@p54B334B0.dip0.t-ipconnect.de)`
`09:12.58`	`*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)`
`09:13.52`	`*** join/#asterisk Ahrimanes (n=ma@81.7.159.2)`
`09:15.24`	`*** join/#asterisk RoyKa (n=roy@217-175-152.100710.adsl.tele2.no)`
`09:29.26`	`*** join/#asterisk Bazy (n=bazy@89.137.178.124)`
`09:33.03`	`*** join/#asterisk penguinFunk (n=penguin@87.224.86.46)`
`09:38.30`	`*** join/#asterisk shadebob (n=chatzill@adll-219-245-206-196.marocconnect.net.ma)`
`09:39.17`	`*** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net)`
`09:40.46`	`*** join/#asterisk Dibbler (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com)`
`09:49.44`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53)`
`09:52.06`	`*** join/#asterisk shadebob (n=chatzill@adll-219-245-206-196.marocconnect.net.ma)`
`09:56.40`	`*** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) [NETSPLIT VICTIM]`
`09:56.40`	`*** join/#asterisk vlt (n=dm@p54B334B0.dip0.t-ipconnect.de) [NETSPLIT VICTIM]`
`09:56.40`	`*** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) [NETSPLIT VICTIM]`
`09:56.40`	`*** join/#asterisk bkw_ (n=brian@72-254-46-103.client.stsn.net) [NETSPLIT VICTIM]`
`09:56.40`	`*** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) [NETSPLIT VICTIM]`
`09:56.41`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`09:56.41`	`*** join/#asterisk foxxtrot (n=craig@c-67-185-241-244.hsd1.wa.comcast.net) [NETSPLIT VICTIM]`
`09:56.41`	`*** join/#asterisk Zaw (i=zaw@unaffiliated/zaw) [NETSPLIT VICTIM]`
`09:56.41`	`*** join/#asterisk lorinc (n=ang@pool-6007.adsl.interware.hu) [NETSPLIT VICTIM]`
`09:56.41`	`*** join/#asterisk TommyBJ (n=JensenTB@80.89.35.2) [NETSPLIT VICTIM]`
`09:56.41`	`*** join/#asterisk mrc3_ (n=mrc3@189.157.107.61) [NETSPLIT VICTIM]`
`09:56.41`	`*** join/#asterisk osiris (n=osiris@71.205.27.131) [NETSPLIT VICTIM]`
`09:56.42`	`*** join/#asterisk kgx (n=kgx@60.234.20.178) [NETSPLIT VICTIM]`
`09:56.42`	`*** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) [NETSPLIT VICTIM]`
`09:56.42`	`*** join/#asterisk Nivex (n=kjotte@user-0ce2nsk.cable.mindspring.com) [NETSPLIT VICTIM]`
`09:56.42`	`*** join/#asterisk Az_au (n=az@216.127.73.119) [NETSPLIT VICTIM]`
`09:56.42`	`*** mode/#asterisk [+o Qwell] by irc.freenode.net`
`09:56.42`	`*** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) [NETSPLIT VICTIM]`
`09:57.18`	`*** join/#asterisk fr33bi3 (n=fr33bi3@122.164.139.137)`
`10:07.07`	`*** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) [NETSPLIT VICTIM]`
`10:07.10`	`*** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) [NETSPLIT VICTIM]`
`10:07.10`	`*** join/#asterisk fr33bi3 (n=fr33bi3@122.164.139.137) [NETSPLIT VICTIM]`
`10:07.11`	`*** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) [NETSPLIT VICTIM]`
`10:07.11`	`*** join/#asterisk bkw_ (n=brian@72-254-46-103.client.stsn.net) [NETSPLIT VICTIM]`
`10:07.11`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`10:07.11`	`*** join/#asterisk foxxtrot (n=craig@c-67-185-241-244.hsd1.wa.comcast.net) [NETSPLIT VICTIM]`
`10:07.11`	`*** join/#asterisk Zaw (i=zaw@unaffiliated/zaw) [NETSPLIT VICTIM]`
`10:07.11`	`*** join/#asterisk lorinc (n=ang@pool-6007.adsl.interware.hu) [NETSPLIT VICTIM]`
`10:07.11`	`*** join/#asterisk TommyBJ (n=JensenTB@80.89.35.2) [NETSPLIT VICTIM]`
`10:07.12`	`*** join/#asterisk mrc3_ (n=mrc3@189.157.107.61) [NETSPLIT VICTIM]`
`10:07.12`	`*** join/#asterisk osiris (n=osiris@71.205.27.131) [NETSPLIT VICTIM]`
`10:07.12`	`*** join/#asterisk kgx (n=kgx@60.234.20.178) [NETSPLIT VICTIM]`
`10:07.12`	`*** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) [NETSPLIT VICTIM]`
`10:07.12`	`*** join/#asterisk Nivex (n=kjotte@user-0ce2nsk.cable.mindspring.com) [NETSPLIT VICTIM]`
`10:07.12`	`*** join/#asterisk Az_au (n=az@216.127.73.119) [NETSPLIT VICTIM]`
`10:07.12`	`*** mode/#asterisk [+o Qwell] by irc.freenode.net`
`10:11.16`	`*** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr)`
`10:11.41`	`angryuser`	`good day everybody`
`10:11.44`	`*** join/#asterisk orkid (n=orkid@dataq2.utias.utoronto.ca)`
`10:15.41`	`AF-Slash`	`god day angryuser`
`10:15.44`	`AF-Slash`	`*good`
`10:16.43`	`angryuser`	`AF-Slash: ypu can call me god, no problem:)`
`10:17.16`	`AF-Slash`	`lol`
`10:17.40`	`*** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro)`
`10:18.45`	`AF-Slash`	`whats up angryuser`
`10:19.20`	`Mahmoud`	`any one here running asterisk on freebsd?`
`10:19.37`	`angryuser`	`AF-Slash: nothing * is working fine(finally) going to run some new tests in 40 min's`
`10:19.59`	`Ahrimanes`	`Mahmoud: i used to run lots of Asterisk on FreeBSD`
`10:20.21`	`AF-Slash`	`zaptel hardware doesnt work great on freebsd`
`10:20.28`	`Mahmoud`	`Ahrimanes, when i type "sockstat \| grep asterisk" i don't see it listening on UDP 5090, instead, it just says "stream"`
`10:20.50`	`*** part/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`10:20.53`	`Mahmoud`	`Ahrimanes, when my SIP phones connect to it, they recieve UDP messages "port not available" or similar`
`10:21.00`	`Ahrimanes`	`Mahmoud: hm 5090? sip is 5060`
`10:21.07`	`Mahmoud`	`sorry, i meant 60`
`10:21.10`	`Ahrimanes`	`ok`
`10:21.22`	`Ahrimanes`	`Mahmoud: you sure that chan_sip.so is loaded right in asterisk ?`
`10:21.54`	`Mahmoud`	`Ahrimanes, heh, 0 modules loaded`
`10:21.59`	`Mahmoud`	`"module show"`
`10:22.08`	`Ahrimanes`	`Mahmoud: thaaat might be a problem :)`
`10:22.09`	`Mahmoud`	`Ahrimanes, this is my 1st touch with *`
`10:22.26`	`Ahrimanes`	`Mahmoud: :)`
`10:24.42`	`Mahmoud`	`so, i'll create modules.conf and add some modules`
`10:25.05`	`Ahrimanes`	`yup`
`10:25.08`	`Ahrimanes`	`or autoload`
`10:25.50`	`*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)`
`10:26.49`	`Mahmoud`	`is ODBC used to store user accounts and dial plans in databases?`
`10:30.44`	`Ahrimanes`	`it can be used for that`
`10:33.52`	`Mahmoud`	`hmm still getting port unreachable icmp messages`
`10:34.47`	`Mahmoud`	`i have only one config file, called "modules.conf" with [modules] \n autoload \n load=chan_sip.so`
`10:34.58`	`Ahrimanes`	`asterisk -rx "load chan_sip.so" && sockstat -4 \| grep 5060`
`10:35.27`	`Ahrimanes`	`Mahmoud: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+modules.conf`
`10:36.30`	`Mahmoud`	`[Feb 28 14:33:36] WARNING[1612]: loader.c:362 load_dynamic_module: Error loading module 'chan_sip.so': Cannot open "/usr/local/asteresk/lib/asterisk/modules/chan_sip.so"`
`10:36.45`	`Ahrimanes`	`hm`
`10:36.54`	`Mahmoud`	`no sip.conf`
`10:37.00`	`Mahmoud`	`is it required for the module to run?`
`10:37.21`	`Ahrimanes`	`hm probably, but chan_sip.so doesnt exist.. that is required for sure`
`10:37.40`	`Mahmoud`	`is is there`
`10:37.44`	`Mahmoud`	`ohhh`
`10:37.47`	`Mahmoud`	`i am stupid`
`10:38.08`	`Mahmoud`	`typo /usr/local/asteresk/lib/asterisk/modules/chan_sip.so`
`10:38.14`	`Ahrimanes`	`seems your install might not be good`
`10:38.16`	`Mahmoud`	`i renamed it manually`
`10:38.26`	`Mahmoud`	`i renamed asteresk to asterisk`
`10:38.29`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net)`
`10:38.33`	`Mahmoud`	`by "mv" command :D`
`10:38.37`	`Ahrimanes`	`oh`
`10:38.39`	`Mahmoud`	`a very dirty way`
`10:38.52`	`Ahrimanes`	`/usr/local/asterisk/etc is where the configs are then ?`
`10:39.07`	`Mahmoud`	`my prefix is /usr/local/asterisk`
`10:39.21`	`Mahmoud`	`so under there, i have etc, share, var, lib, bin, sbin..etc`
`10:39.48`	`Ahrimanes`	`ok, well if you renamed a folder or file, you need to change asterisk.conf to reflect that change`
`10:39.50`	`Mahmoud`	`i just wanted to see all files in one place, since i'm totally new to *`
`10:39.59`	`Mahmoud`	`yeah.. which i didn't`
`10:40.15`	`Mahmoud`	`actually i have no asterisk.conf, but it seems the default is what it was told in the ./configure process`
`10:40.51`	`Ahrimanes`	`ah you didnt do make samples ?`
`10:41.05`	`Mahmoud`	`nope, didn't work as it was written in the PDF guide`
`10:41.13`	`Ahrimanes`	`hm ok`
`10:41.13`	`Mahmoud`	`the PDF says, no need for ./configure either`
`10:41.19`	`Ahrimanes`	`tried installing from ports?`
`10:41.22`	`Mahmoud`	`but the version i'm using is new, and changed`
`10:41.29`	`*** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it)`
`10:41.41`	`Mahmoud`	`i want to learn it.. if i used ports it will swamp my system without knowing what happen`
`10:41.41`	`Ahrimanes`	`ok`
`10:41.54`	`Mahmoud`	`i think you installed it without ./configure right?`
`10:43.46`	`Ahrimanes`	`hm ./configure && make`
`10:43.54`	`Ahrimanes`	`is basically what i usually do`
`10:44.10`	`Mahmoud`	`and then "make samples" ?`
`10:44.42`	`Ahrimanes`	`make install`
`10:44.45`	`Ahrimanes`	`then make samples`
`10:44.57`	`Ahrimanes`	`then you'll get sample config files to play with`
`10:45.11`	`Mahmoud`	`yeah, but it didn't work for me.. could be typos`
`10:45.32`	`Mahmoud`	`i was doing it ad midnight`
`10:45.36`	`Mahmoud`	`s/ad/at/`
`10:48.02`	`Mahmoud`	`erased everything and started from scratch =]`
`10:51.48`	`giasai68`	`hello`
`10:52.18`	`giasai68`	`I want generate a call and forward it to IP address using asterisk`
`10:52.23`	`giasai68`	`how I can do?`
`10:55.35`	`Mahmoud`	`Ahrimanes, listening on 5060 now, thanks to make samples`
`10:55.53`	`Ahrimanes`	`Mahmoud: glad to help`
`10:56.02`	`Mahmoud`	`Ahrimanes, but i'll erase all files in /usr/local/asterisk/etc/asterisk to do the config from scratch by my self.. is this enough to undo everything made by "make samples"`
`10:56.08`	`*** join/#asterisk key2 (n=key2@81.52.138.22)`
`10:57.03`	`Ahrimanes`	`Mahmoud: yeah, make samples just installs configs`
`10:57.10`	`Mahmoud`	`how neat`
`11:03.50`	`*** join/#asterisk saftsack (n=oliver@pD9E07AD7.dip.t-dialin.net)`
`11:16.07`	`*** join/#asterisk kieran491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au)`
`11:20.22`	`kieran491`	`to recive calls on asterisk what ports do you need open?`
`11:22.27`	`Ahrimanes`	`5060 and 10000 to 20000 udp`
`11:22.53`	`kieran491`	`k thanks`
`11:23.14`	`Ahrimanes`	`if you're doing sip anyways.. if you're doing iax more ports would be needed`
`11:26.45`	`tzafrir_laptop`	`I just wasted a while on a stupid problem: if after reloading zaptel (and/or xpp) things "don't work" and /proc/zaptel (or /proc/xpp) is empty when it shouldn't be, try lsof /proc/zaptel or lsof /proc/xpp`
`11:27.17`	`tzafrir_laptop`	`and see if any shell whose cwd is there`
`11:27.45`	`penguinFunk`	`how can i find out about what music i can use for my music on hold without breaching any copyirghts ?`
`11:27.54`	`penguinFunk`	`copyrights*`
`11:28.23`	`Ahrimanes`	`any royalty free music will do`
`11:28.29`	`tzafrir_laptop`	`penguinFunk, classical`
`11:28.34`	`kieran491`	`Ahrimanes: what other ports sorry? i am using iax`
`11:28.42`	`tzafrir_laptop`	`or look in creative-common's site`
`11:28.44`	`penguinFunk`	`thanks`
`11:29.32`	`Ahrimanes`	`kieran491: 4569 udp`
`11:29.58`	`kieran491`	`no range?`
`11:30.02`	`Gido-E`	`tzafrir_laptop check for creative commons`
`11:30.55`	`Ahrimanes`	`kieran491: uhm afair iax sends rtp over the same port`
`11:31.17`	`kieran491`	`ohh k`
`11:32.30`	`Ahrimanes`	`kieran491: http://www.voip-info.org/wiki-IAX has lots of info`
`11:34.30`	`penguinFunk`	`cant believe that you still have to pay for royalty free music`
`11:34.42`	`kieran491`	`thanks`
`11:34.58`	`Ahrimanes`	`penguinFunk: you can find free, royalty free music on google`
`11:35.53`	`*** join/#asterisk coppice (n=chatzill@106.206.17.210.dyn.pacific.net.hk)`
`11:38.01`	`*** join/#asterisk fr33bi3 (n=fr33bi3@122.164.139.137)`
`11:40.49`	`*** join/#asterisk Strom_M (n=strom@209.19.56.4)`
`11:41.07`	`*** join/#asterisk shinux__ (n=shinux@41.222.64.83)`
`11:41.46`	`*** join/#asterisk fourcheeze (n=rich@85-189-96-153.rcg-global.managedbroadband.co.uk)`
`11:42.23`	`*** join/#asterisk tparcina (n=tparcina@cisco16.fesb.hr)`
`11:42.31`	`*** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net)`
`11:45.51`	`fourcheeze`	`our outbound sip provider in their reporting gives me lots of stats like packet loss during calls`
`11:46.01`	`fourcheeze`	`can I get stats like that out of * ?`
`11:48.25`	`*** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au)`
`11:48.30`	`kezza491`	`hmm`
`11:48.31`	`kezza491`	`i have set my asterisk box up and i cant seem to recive any calls...`
`11:56.35`	`fourcheeze`	`kezza491: something in the logs?`
`11:59.00`	`Makenshi`	`T-00:01:00 until the tenders are due`
`11:59.52`	`kezza491`	`fourcheeze: Nop`
`12:00.09`	`fourcheeze`	`kezza491: so nothing arrived then`
`12:00.17`	`fourcheeze`	`try debugging the ip that things are coming from`
`12:00.26`	`fourcheeze`	`kezza491: are we talking incoming sip?`
`12:00.43`	`kezza491`	`IAX`
`12:00.48`	`fourcheeze`	`ok`
`12:00.53`	`fourcheeze`	`and it's in debug mode?`
`12:01.14`	`kezza491`	`ehh nop`
`12:01.21`	`fourcheeze`	`you might want that`
`12:01.27`	`fourcheeze`	`then put a call through and see what appears on the console`
`12:01.33`	`fourcheeze`	`run the console with asterisk -rvvvvvvvvvvvvvvc`
`12:02.15`	`fourcheeze`	`the exact number of vs isn't important ;-)`
`12:02.32`	`kezza491`	`8-)`
`12:03.02`	`kezza491`	`there gose me...`
`12:03.19`	`kezza491`	`i am geting nothing`
`12:10.16`	`*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53)`
`12:12.52`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`12:13.30`	`Mahmoud`	`why sould asterisk need chan_oss.so or chan_alsa.co ?`
`12:13.46`	`tzafrir`	`to use a local sound card`
`12:13.52`	`Mahmoud`	`for?`
`12:14.09`	`Mahmoud`	`to play things to the local * server speaker?`
`12:14.14`	`tzafrir`	`sending messages to a speaker, or even as a poor-man's phone`
`12:14.40`	`Mahmoud`	`is it all about playing things in the local * server?`
`12:14.44`	`tzafrir`	`take a look at yate's gtk interface for some inspiration`
`12:21.18`	`*** join/#asterisk Dibbler (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com)`
`12:25.26`	`Mahmoud`	`any idea what modules are chan_sip.co dependencies?`
`12:25.44`	`Mahmoud`	`i have no modules loaded, but chan_sip.so, and sure doesn't work`
`12:25.53`	`Mahmoud`	`it works when i load all modules`
`12:26.56`	`tzafrir`	`try loading half the modules, then`
`12:27.12`	`Mahmoud`	`they are 135 modules`
`12:28.59`	`Mahmoud`	`when i try to load chan_sip.so, it says undified symbol "ast_park_call"`
`12:29.36`	`Mahmoud`	`google doesn't seem helpful`
`12:39.19`	`*** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au)`
`12:41.40`	`*** part/#asterisk Bazy (n=bazy@89.137.178.124)`
`12:41.40`	`*** join/#asterisk Bazy (n=bazy@89.137.178.124)`
`12:41.48`	`fourcheeze`	`Mahmoud: I think that might be res_features.so`
`12:41.59`	`fourcheeze`	`can't remember exactly OTTOMH`
`12:42.00`	`*** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com)`
`12:42.15`	`Mahmoud`	`you are probably right, i tried loading all res_ ones and it worked`
`12:43.36`	`Mahmoud`	`that's it, res_features.so!`
`12:44.13`	`*** join/#asterisk mkl1525 (n=qwertz@217.83.2.66)`
`12:44.50`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)`
`12:44.56`	`mkl1525`	`Hi, when a caller leaves a calling queue I can get the call with the h extension, but is there any way to get which agent handled this call?`
`12:56.04`	`*** join/#asterisk taishi (n=juanma@zion.dattaweb.com)`
`12:57.55`	`HarryR`	`mkl1525, yeah, you get an AgentConnect event through Manager`
`12:58.08`	`taishi`	`I can't figure how to make an user in sip.conf/users.conf to receive calls and direct him to a context`
`12:58.18`	`taishi`	`The main thing is I dont want that user to log in`
`12:58.24`	`taishi`	`It's more likely an 'operator'`
`13:03.53`	`*** join/#asterisk qdk (n=qdk@90.184.3.249)`
`13:09.57`	`mkl1525`	`HarryR, thanks for the hint but can I use this in the dial plan?`
`13:11.35`	`*** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net)`
`13:11.36`	`*** join/#asterisk drako (n=ljd@unaffiliated/luisjose)`
`13:11.42`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`13:14.51`	`*** join/#asterisk friedrich\| (n=friedric@e177242109.adsl.alicedsl.de)`
`13:15.15`	`*** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br)`
`13:16.48`	`emiquelito`	`hello! I'm writing a softphone with G.729A codec and I would like to know what is the correct number of bytes each rtp packet must have to talk properly with an Asterisk server. Is it 20 bytes?`
`13:17.39`	`fourcheeze`	`emiquelito: isn't that measured in ms worth of sound?`
`13:17.58`	`fourcheeze`	`i.e. 20ms per packet?`
`13:19.58`	`emiquelito`	`fourcheeze thats true, unfortunately I'm not an expert in VoIP.`
`13:20.09`	`emiquelito`	`fourcheeze so in this case, 20ms would be 20 bytes?`
`13:20.29`	`emiquelito`	`at least that's what I'm having in the client side when talking to Asterisk`
`13:22.00`	`*** join/#asterisk Strom_M (n=strom@209.19.56.4)`
`13:22.34`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`13:23.34`	`*** join/#asterisk drako (n=ljd@unaffiliated/luisjose)`
`13:24.01`	`fourcheeze`	`I've no idea how many bytes in 1 ms`
`13:24.33`	`Zefk`	`I have tested asterisk 1.4.0 with softphones and VoIP hardphones (Avaya 4610 and 4620) in our call center. I'm looking for a solution with hardphones because all the softphones that I tested (X-lite, SJPhone, IDEFisk, Diax) does not provide the voice quality requested in a call center. Is anyone know a quality hardphone that have soft comands (from PC) in order to integrate the phone in our applications? ... Thx`
`13:24.49`	`*** join/#asterisk nfi\|ermes (n=ermsewrk@217.220.121.62)`
`13:25.17`	`nfi\|ermes`	`hi all`
`13:25.37`	`*** join/#asterisk \|Vulture\| (n=_Vulture@101.222.121.70.cfl.res.rr.com)`
`13:25.44`	`Strom_M`	`Zefk: which codec are you using?`
`13:26.09`	`Zefk`	`Strom_M: a-law, u-law, and gsm`
`13:26.19`	`Strom_M`	`don't use gsm`
`13:26.24`	`RoyK`	`emiquelito: 8000 / 50 = 160 - that's the average number of bits per package with g.729, meaning an average of 20 data bytes per package @20ms packetization, but keep in mind that g.729 is not a linear codec, so packet sizes will vary`
`13:26.35`	`fourcheeze`	`what's wrong with gsm?`
`13:26.36`	`TaiSHi`	`I'd like to make a 'virtual' operator inside a LAN (for testing purposes)`
`13:26.38`	`Strom_M`	`especially not if you're going for quality`
`13:27.00`	`TaiSHi`	`But I can't seem to make up an user to direct it to an extension`
`13:27.06`	`Zefk`	`Strom_M: The main test was done with a-law and u-law with no transcoding`
`13:27.17`	`Strom_M`	`fair enough`
`13:27.19`	`\|Vulture\|`	`HAs anyone ever seen this issue: http://www.pastebin.ca/375621`
`13:27.44`	`Strom_M`	`Zefk: good, inexpensive, quality hardphones that I like are the polycom ip430`
`13:27.46`	`\|Vulture\|`	`I am wonder what your resolutions were, I have a PRI that keeps cycling up/down because of errors on the D-Chan`
`13:27.51`	`emiquelito`	`RoyK, I see... sorry for the basic question but, why 50? (800/50)`
`13:28.09`	`RoyK`	`1000ms / 20ms = 50 packets per second`
`13:28.22`	`emiquelito`	`ouch, now I understand`
`13:28.27`	`emiquelito`	`thanks a lot`
`13:28.29`	`Zefk`	`Strom_M: I need a phone with soft interface. I have to integrate the phone in our applications`
`13:28.47`	`Strom_M`	`Zefk: what do you mean "soft interface"`
`13:29.14`	`TaiSHi`	`== Auto fallthrough, channel 'SIP/600-08aa22c8' status is 'CHANUNAVAIL'`
`13:29.15`	`Zefk`	`Strom_M: An app that commands the hardphone.`
`13:29.24`	`Strom_M`	`erm yeah...I need to wake up`
`13:29.56`	`Strom_M`	`shrug I have no experience with that...`
`13:31.01`	`TaiSHi`	`I'd like to make a 'virtual' operator inside a LAN (for testing purposes), so when an intern receives a call (without an user logged in), it would do Answer(), Playback() and Hangup()`
`13:31.16`	`nfi\|ermes`	`when i logon in asterisk-gui, asterisk goes to segmentation fault: dbg of core dump ---> http://pastebin.com/890614`
`13:32.07`	`nfi\|ermes`	`anyone can help me ?`
`13:32.07`	`TaiSHi`	`nfi\|ermes: what motherboard do you have ?`
`13:33.37`	`nfi\|ermes`	`why TaiSHi ?`
`13:33.59`	`TaiSHi`	`There is explicit doc on the source that says`
`13:34.15`	`TaiSHi`	`If you have a VIA mobo, and it detects the proc as i686`
`13:34.20`	`TaiSHi`	`It will cause random core dumps`
`13:34.49`	`nfi\|ermes`	`it will cause segmentation fault ?`
`13:34.51`	`kippi`	`Help!!`
`13:35.01`	`kippi`	`I am getting a busy tone when I ring in`
`13:35.07`	`kippi`	`I am getting this message`
`13:35.07`	`kippi`	`chan_zap.c:8383 pri_dchannel: Ring requested on channel 0/1 already in use on span 1.`
`13:35.33`	`TaiSHi`	`nfi\|ermes: random core dumps, I don't know much more :P`
`13:35.43`	`Strom_M`	`kippi: are your outbound calls hunting from the high-numbered channel, or from the low-numbered channel?`
`13:35.55`	`kippi`	`not sire`
`13:36.05`	`\|Vulture\|`	`Anyone here use XO Communications for PRI providing?`
`13:36.25`	`Strom_M`	`kippi: well, now would certainly be a good time to find out :)`
`13:36.48`	`kippi`	`how can I find out`
`13:37.20`	`nfi\|ermes`	`TaiSHi, where can i find this documentation ?`
`13:37.27`	`penguinFunk`	`http://img427.imageshack.us/img427/6991/abaddayinofficeob5.gif`
`13:37.27`	`TaiSHi`	`sec`
`13:37.33`	`Strom_M`	`when you place an outbound call, what channel does the call go out over?`
`13:38.59`	`TaiSHi`	`I'd like to make a 'virtual' operator inside a LAN (for testing purposes), so when an intern receives a call (without an user logged in), it would do Answer(), Playback() and Hangup()`
`13:40.37`	`TaiSHi`	`I cant seem to find it nfi\|ermes, still looking @ it`
`13:40.58`	`nfi\|ermes`	`ok thx`
`13:41.44`	`creativx`	`how can i get the configured CID name of a SIP user via the manager interface?`
`13:42.45`	`*** join/#asterisk AlfaScorpii (n=alfascor@64-12-16-190.fibertel.com.ar)`
`13:42.51`	`AlfaScorpii`	`morning people`
`13:43.01`	`creativx`	`nevermind, sipshowpeer was it`
`13:43.14`	`TaiSHi`	`morning Alfa`
`13:43.23`	`AlfaScorpii`	`TaiSHi: how r u?`
`13:43.31`	`TaiSHi`	`With problems :D`
`13:43.48`	`*** join/#asterisk Dibbler (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com)`
`13:44.00`	`TaiSHi`	`Trying to make a virtual operator with a sexy voice`
`13:44.04`	`TaiSHi`	`But I can't get it to answer :P`
`13:44.06`	`AlfaScorpii`	`people i have a proble with outbaund calls (pstn) the calls cuts at 40 sec,`
`13:44.15`	`*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`13:45.14`	`AlfaScorpii`	`codec problem?`
`13:45.19`	`AlfaScorpii`	`any idea?`
`13:47.33`	`TaiSHi`	`Personally, no, but the forums had some info about calls cutting @ 40secs`
`13:49.10`	`AlfaScorpii`	`:(`
`13:49.17`	`TaiSHi`	`Ok, nfi\|ermes`
`13:49.24`	`TaiSHi`	`Check TFOT manual`
`13:49.38`	`TaiSHi`	`Compiling Asterisk -manual page- 45`
`13:51.16`	`\|Vulture\|`	`Anyone here use XO Communications for PRI providing?`
`13:52.30`	`TaiSHi`	`Not me`
`13:52.31`	`TaiSHi`	`Still`
`13:52.45`	`AlfaScorpii`	`not me`
`13:53.13`	`TaiSHi`	`\|Vulture\|: Know how to direct a call (to an user in sip.conf - users.conf who is logged out) to an extension ?`
`13:53.29`	`TaiSHi`	`Right now, I just made the user, and set him up w/o voicemal and context = incoming`
`13:57.48`	`\|Vulture\|`	`I don't think I understand the question`
`13:58.01`	`*** part/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br)`
`13:58.07`	`e-ddie`	`i dont think i understand the answer`
`13:58.09`	`\|Vulture\|`	`you are trying to call a sip user who is offline?`
`13:59.01`	`TaiSHi`	`Actually, my idea is to`
`13:59.21`	`TaiSHi`	`Call $(something/someone/UFO)`
`13:59.29`	`TaiSHi`	`And get an answer from an extension`
`13:59.40`	`*** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br)`
`13:59.56`	`flujan`	`hi guys... I am placing some calls using call files...`
`13:59.57`	`TaiSHi`	`(The extension I refer to is: [incoming] // answer, playback, hangup)`
`14:00.09`	`flujan`	`It works fine, but sometimes I got error messages like this:`
`14:00.12`	`flujan`	`Call failed to go through, reason 8`
`14:00.33`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`14:00.35`	`flujan`	`How can I debug this information, for instance, reason 8 sounds nothing to me... What this really means?`
`14:02.50`	`TaiSHi`	`Understood now \|Vulture\|?`
`14:02.57`	`*** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr)`
`14:04.56`	`*** join/#asterisk TJBraza (n=tj@200.203.32.201)`
`14:05.33`	`TJBraza`	`Hello...`
`14:05.48`	`TJBraza`	`I'm in need of some help.. can I shoot a question, or is this not that kind of forum?`
`14:06.00`	`TJBraza`	`channel, i mean`
`14:06.13`	`*** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il)`
`14:06.35`	`Dovid`	`morning it all`
`14:06.46`	`flujan`	`morning Dovid.`
`14:06.50`	`TJBraza`	`hello Dovid`
`14:07.40`	`*** join/#asterisk Zoid_99 (n=chris@24.214.206.254)`
`14:08.22`	`TJBraza`	`anyway, is there a way to force echo cancellation on sip to sip channels?`
`14:08.26`	`tzafrir_laptop`	`TJBraza, we tend to shoot troubles here`
`14:08.29`	`TaiSHi`	`mornin'`
`14:08.44`	`TaiSHi`	`Yeah, they usually shoot me u_U`
`14:10.05`	`TaiSHi`	`I'd like to make a 'virtual' operator inside a LAN (for testing purposes), so when an intern receives a call (without an user logged in), it would do Answer(), Playback() and Hangup()`
`14:10.06`	`TJBraza`	`hello TaiShi`
`14:10.10`	`Chris-NB`	`hi`
`14:10.20`	`Chris-NB`	`anyone using a Thomson ST2030S phone?`
`14:10.21`	`TaiSHi`	`I was just saying mornin' to Dovid`
`14:10.23`	`TaiSHi`	`But Hello`
`14:11.11`	`Dovid`	`back at ya`
`14:13.20`	`TaiSHi`	`Okey`
`14:13.50`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`14:13.50`	`TJBraza`	`i'm getting echo on a strict sip-to-sip channel.. all references on echo cancellation i can find are for zap channels`
`14:14.09`	`Strom_M`	`TJBraza: what kind of station equipment are you using?`
`14:14.10`	`TJBraza`	`i dont even have a card plugged in`
`14:14.10`	`[TK]D-Fender`	`TJBraza: What are you using on the endpoints?`
`14:14.25`	`TJBraza`	`linksys pap2`
`14:14.36`	`*** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net)`
`14:14.42`	`TJBraza`	`i think it's due to the latency`
`14:14.48`	`[TK]D-Fender`	`TJBraza: on both sides fo the call?`
`14:14.52`	`TJBraza`	`it's kind of huge... about 300ms`
`14:15.07`	`TJBraza`	`only the one making the call gets echo`
`14:15.18`	`[TK]D-Fender`	`TJBraza: Whats on the other side of the call?`
`14:15.27`	`TJBraza`	`PSTN line`
`14:15.33`	`TJBraza`	`they dont get echo, ever`
`14:15.36`	`[TK]D-Fender`	`TJBraza: Then its not SIP-SIP.`
`14:15.49`	`Dovid`	`whats the command to limit the amount of channels a SIP acount can have ?`
`14:15.50`	`*** join/#asterisk vlt (n=dm@p54B334B0.dip0.t-ipconnect.de)`
`14:15.53`	`[TK]D-Fender`	`TJBraza: And they might very well be the point of echo.`
`14:16.19`	`*** join/#asterisk ximwork (n=ximwork@adsl-4-233-20.mem.bellsouth.net)`
`14:16.20`	`TJBraza`	`ah, yes, but if i put another pap2 on the other side (300ms latency), i get echo anyway`
`14:16.24`	`TJBraza`	`always for the person calling`
`14:16.27`	`TaiSHi`	`AHh`
`14:16.29`	`TaiSHi`	`I think I found out`
`14:16.30`	`TaiSHi`	`!!`
`14:16.30`	`Ahrimanes`	`Dovid: set call-limit in sip.conf on the pere`
`14:16.33`	`TaiSHi`	`brb`
`14:16.33`	`Ahrimanes`	`peer`
`14:16.38`	`TJBraza`	`i think the echo suppressionon the pap2 is very poor`
`14:16.52`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`14:17.10`	`Dovid`	`TJBraza: which pap2 r u using ?`
`14:17.14`	`[TK]D-Fender`	`TJBraza: It is a lower model than the old SPA-2000 base....`
`14:17.54`	`vlt`	`Hello. I'm having problems with (or understanding) IAX2 connection between 2 * servers. In A's iax.conf I defined a peer B of type friend. Whose username is defined there? A's on B or B's on A?`
`14:18.03`	`TaiSHi`	`In an extension, how do I make it to go to another extension ?`
`14:18.11`	`vlt`	`TaiSHi: GoTo()`
`14:18.26`	`TaiSHi`	`Thank you :)`
`14:19.32`	`vlt`	`errm ... who's*`
`14:20.06`	`TaiSHi`	`Mmm`
`14:20.07`	`TaiSHi`	`vlt`
`14:20.17`	`TaiSHi`	`It didnt send me to other context`
`14:20.24`	`TaiSHi`	`(I think I mis made my question before)`
`14:20.27`	`[TK]D-Fender`	`TaiSHi: "show application goto"`
`14:20.35`	`[TK]D-Fender`	`TaiSHi: Read the full instructions.`
`14:20.41`	`*** part/#asterisk tparcina (n=tparcina@cisco16.fesb.hr)`
`14:20.53`	`TaiSHi`	`Command not found`
`14:21.09`	`TaiSHi`	`No such command 'show application goto' (type 'help' for help)`
`14:21.13`	`[TK]D-Fender`	`TaiSHi: you did that at the * CLI?`
`14:21.22`	`TaiSHi`	`Yes`
`14:21.23`	`TJBraza`	`[TK]D-Fender`
`14:21.36`	`TJBraza`	`do you know of a way to force echo suppression in asterisk in this case?`
`14:21.48`	`[TK]D-Fender`	`TaiSHi : wait... try "core show application goto"`
`14:22.06`	`TaiSHi`	`Great, worked :)`
`14:22.09`	`TaiSHi`	`Thanks, I will read up now`
`14:22.12`	`TJBraza`	i`m using th pap2t, latest firmware.. the one with green leds
`14:22.18`	`[TK]D-Fender`	`TJBraza: there is no EC on Sip unfortunately. The endpoints are expected to do their jobsproperly`
`14:22.43`	`[TK]D-Fender`	`TJBraza: Your firmware comes with LEDS? :)`
`14:23.27`	`TJBraza`	`lol`
`14:23.38`	`TJBraza`	`so basically i'm screwed?`
`14:23.59`	`elriah`	`Greets. Are there any user-doc templates for asterisk out there?`
`14:24.21`	`*** join/#asterisk stkn\|work (n=stkn\|wor@gentoo/developer/pdpc.active.stkn)`
`14:24.22`	`*** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net)`
`14:24.48`	`vlt`	`hmmmm, to connect two Asterisks I have to create _two_ peers on each machine, right? An [A-outbound] and an [A-inbound]?`
`14:26.02`	`Strom_M`	`vit: you can also create a single "friend" entry on both machines`
`14:27.12`	`vlt`	`Strom_M: But who's username do I need in iax.conf's [A] section? A's on B or B's on A?`
`14:27.53`	`Strom_M`	`your question is more confusing than it needs to be`
`14:27.59`	`Strom_M`	`on box A:`
`14:28.38`	`*** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net)`
`14:28.52`	`Strom_M`	`on box A, the name in brackets will be used as the user= entry on box B`
`14:28.56`	`Strom_M`	`and vice versa`
`14:30.33`	`vlt`	`Strom_M: Can we go private?`
`14:33.48`	`kippi`	`I am getting this error, I have added the information to sip.conf and reloaded, but getting this error http://pastebin.ca/375664`
`14:33.59`	`*** join/#asterisk Vulture- (n=\|Vulture@101.222.121.70.cfl.res.rr.com)`
`14:34.29`	`penguinFunk`	`username in the phone config doesnt match the user= part of the sip.conf`
`14:34.49`	`Vulture-`	`Anyone here ever install a PRI from XO Communications?`
`14:35.14`	`penguinFunk`	`kippi: try using type=friend, host=dynamic`
`14:35.22`	`kippi`	`ok`
`14:35.29`	`Strom_M`	`I think we should explicitly change all error message text to read "Go to #asterisk" -- since no one reads the text of the message, no one will come to #asterisk to inquire about the error :)`
`14:36.29`	`[TK]D-Fender`	`Strom_M: "Perverse Psychology 101"`
`14:36.39`	`*** join/#asterisk lorinc (n=ang@pool-7449.adsl.interware.hu)`
`14:36.42`	`kippi`	`penguinFunk: already got that there`
`14:36.45`	`Strom_M`	`hehe`
`14:37.34`	`*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`14:37.34`	`*** mode/#asterisk [+o anthm] by ChanServ`
`14:37.36`	`*** join/#asterisk hellojoe (n=hijoe@c-67-160-249-95.hsd1.ca.comcast.net)`
`14:37.43`	`[TK]D-Fender`	`kippi: Then read the BIG PRINT.`
`14:38.21`	`kippi`	`but it is correct, could it be because i am coming from another subnet?`
`14:38.40`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`14:38.40`	`*** mode/#asterisk [+o mog] by ChanServ`
`14:40.04`	`hellojoe`	`is there a way to dial into another context withing extensions.conf. For example, let's say there is an incoming call received on [inbound] context which, after processing needs to go to [processed] context with a Dial(LOCAL/5555${EXTEN}) command?`
`14:40.18`	`[TK]D-Fender`	`kippi: * is not lying. you have configured an end wrong.`
`14:40.38`	`hellojoe`	`i thought LOCAL translates to [default] context within extensions.conf`
`14:40.39`	`[TK]D-Fender`	`hellojoe: "show application goto"`
`14:40.50`	`hellojoe`	`goto doesn't do multiple calls`
`14:40.58`	`aydiosmio`	`I want to recieve a call, play a message and send MOH to the channel, this initiates a call to a representative and when the representitive picks up, I want to play a message to the rep and then move the caller from MOH to that rep... how would I do that?`
`14:41.10`	`[TK]D-Fender`	`hellojoe: No, "Local/" is just a way to nest an entire sub-channel which I doubt you need.`
`14:41.13`	`hellojoe`	`i want to be able to throw Dial(LOCAL/${Call1}&LOCAL/${Call2})`
`14:41.17`	`[TK]D-Fender`	`hellojoe: Or maybe you do.`
`14:41.31`	`hellojoe`	`you are right, I don't want to nest the channel`
`14:41.38`	`[TK]D-Fender`	`hellojoe: Dial(Local123@contextyouwant)`
`14:41.49`	`[TK]D-Fender`	`hellojoe: Dial(Local/123@contextyouwant)`
`14:41.50`	`hellojoe`	`aaah!`
`14:41.57`	`[TK]D-Fender`	`hellojoe: But barring that... GOTO <-`
`14:41.58`	`vlt`	`Strom_M: I pasted a config here: http://rafb.net/p/Y6MmSO21.html -- Can you have a look?`
`14:42.07`	`hellojoe`	`thanks a lot TK`
`14:42.12`	`[TK]D-Fender`	`hellojoe: np.`
`14:42.16`	`TJBraza`	`hey, [TK]-D-Fender does Fender have anything to do with the guitar?`
`14:42.21`	`Vulture-`	`Anyone here ever install a PRI from XO Communications?`
`14:42.24`	`TJBraza`	`maybe a D-tuned one? ;)`
`14:42.32`	`[TK]D-Fender`	`TJBraza: Nope, though I do play guitar`
`14:42.47`	`[TK]D-Fender`	`TJBraza: Its an old CTF FPS gaming nick.`
`14:43.09`	`TJBraza`	`cool.. i play too.. i have a les paul clone ..hehehe`
`14:45.09`	`TJBraza`	`anyway, on the echo matter, i'm pretty much left in the cold, right?`
`14:45.18`	`TJBraza`	`no way to create a dummy zap channel to handle echo or anything?`
`14:45.33`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`14:47.08`	`[TK]D-Fender`	`TJBraza: Pretty much if there aren't any more AEC options available.`
`14:47.38`	`TaiSHi`	`I did it!!`
`14:47.42`	`[TK]D-Fender`	`TJBraza: On SIP its not considered "echo" so I'm not sure how you could even really filter it.`
`14:48.03`	`*** join/#asterisk JoNate (n=noone@mail.wmelec.com)`
`14:49.07`	`*** join/#asterisk e-ddie (n=oal@62.61.133.90.generic-hostname.arrownet.dk)`
`14:49.12`	`[TK]D-Fender`	`TJBraza: See if there are gain settings you can play with.`
`14:49.30`	`*** join/#asterisk Moobius (i=Moobius@www2.techcavalry.com)`
`14:50.10`	`TJBraza`	`it boggles the mind`
`14:50.15`	`TJBraza`	`:-P`
`14:50.37`	`*** join/#asterisk ManxPower (n=manxpowe@68.113.119.116)`
`14:50.49`	`TJBraza`	`have you used * Realtime?`
`14:51.04`	`Vulture-`	`Anyone ever seen a PRI not turnup, goes green, but then says Up/Up/Up/Up/Down and errors with "pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway!"`
`14:51.13`	`ManxPower`	`Vulture-: Yes.`
`14:51.21`	`Vulture-`	`Manx: what was the issue?`
`14:51.45`	`Vulture-`	`I sent 2 new cables to make sure thats not it`
`14:51.47`	`vlt`	`Can anyone tell me how to configure this --> http://rafb.net/p/Y6MmSO21.html ?`
`14:52.03`	`ManxPower`	`I've seen it several times. It is either a mismatch between Asterisk/Zaptel/Libpri or the telco does not have the line correctly set up.`
`14:52.13`	`JoNate`	`hey guys, is there a canned application launcher that works well with asterisk?`
`14:52.14`	`*** join/#asterisk drako (n=ljd@unaffiliated/luisjose)`
`14:52.18`	`*** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud)`
`14:52.24`	`ManxPower`	`Vulture-: you are in the USA, right?`
`14:52.26`	`Vulture-`	`Manx: when you say mismatch, do you mean that the software is not correctly installed?`
`14:52.36`	`Vulture-`	`Manx: correct, it is an XO Communications PRI`
`14:53.03`	`ManxPower`	`Vulture-: I mean something like 1.0 zaptel with 1.2 libpri or something like that.`
`14:53.06`	`Vulture-`	`I was hoping it was the cable, Ill find out in an hr or two`
`14:53.30`	`ManxPower`	`Vulture-: I assume the system is running 1.2?`
`14:53.45`	`Vulture-`	`Manx: I am running the latest 1.2 release`
`14:53.52`	`[TK]D-Fender`	`vlt: http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers`
`14:54.04`	`anonymouz666`	`I am nothing without voip-info`
`14:54.04`	`Vulture-`	`Manx: using a Sangoma A101u`
`14:54.04`	`ManxPower`	`Vulture-: reinstall zaptel and libpri just to be sure.`
`14:54.06`	`anonymouz666`	`the best site ever`
`14:54.15`	`vlt`	`[TK]D-Fender: Thank you.`
`14:54.31`	`ManxPower`	`Vulture-: A101u? I thought that was the unchannelized version of their card.`
`14:54.36`	`[TK]D-Fender`	`Vulture-: And what does "wanrouter status" tell you?`
`14:54.52`	`[TK]D-Fender`	`ManxPower: Single port non-ec (equiv to TE110P)`
`14:55.00`	`ManxPower`	`[TK]D-Fender: Ah.`
`14:55.08`	`ManxPower`	`We use A102's`
`14:55.29`	`[TK]D-Fender`	`ManxPower: Technially that should read "A102u"`
`14:55.31`	`[TK]D-Fender`	`:)`
`14:55.34`	`Vulture-`	`Manx: I updated the Kernel, then did fresh installs of libpri and zaptel and wanrotuer`
`14:55.39`	`Vulture-`	`http://www.pastebin.ca/375682`
`14:55.57`	`[TK]D-Fender`	`ManxPower: The only unchannelized card they have is their DS3 card.`
`14:56.09`	`*** join/#asterisk Daejeo1 (n=chatzill@124.62.144.63)`
`14:56.14`	`Vulture-`	`I have a few 102s but switched to 101s because we only needed 1 PRI`
`14:56.26`	`ManxPower`	`Vulture-: Perhaps there was a kernel issue and a newest 1.2 zaptel was not installed`
`14:56.27`	`[TK]D-Fender`	`Vulture-: Ok, pastebin your zaptel & zapata`
`14:56.59`	`ManxPower`	`We use 102's because we want to be able to standardize and keep spares.`
`14:57.13`	`Daejeo1`	`anyone have idea about integrating ASR-engine with asterisk server?`
`14:57.35`	`Vulture-`	`http://www.pastebin.ca/375686`
`14:57.49`	`Vulture-`	`Manx: that is a good idea`
`14:58.04`	`ManxPower`	`Vulture-: then the only thing left is a bad PRI.`
`14:58.17`	`ManxPower`	`What do you see when you do a pri debug span 1`
`14:58.22`	`Vulture-`	`I hope it isn't a kernel issue those are a PIA :(`
`14:58.47`	`Vulture-`	`debugging on, I will pastebin any output`
`14:58.57`	`ManxPower`	`Vulture-: are the date/time stamps on the kernel modules what you expect`
`14:59.00`	`[TK]D-Fender`	`Vulture-: Looks kosher... how about "ztcfg -vvvv"`
`14:59.30`	`Vulture-`	`everything is looking good, I even compared it right next to a working PRI with the same Sangoma card`
`14:59.33`	`ManxPower`	`i.e. find /lib/modules -name "zaptel.*" -exec ls -l \{\} \;`
`14:59.44`	`*** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it)`
`15:00.08`	`ManxPower`	`Vulture-: call up the telco and say "I don't see a D-Channel"`
`15:01.06`	`Vulture-`	`Manx: thats what I did all day yesterday and they kept confirming it was working`
`15:01.10`	`ManxPower`	`Vulture-: can you plug the working PRI into the non-working server and see if it works`
`15:01.19`	`Vulture-`	`I had a tech hook up his handset and that worked for inbound/outbound`
`15:01.35`	`ManxPower`	`Vulture-: handset?`
`15:01.44`	`ManxPower`	`You mean the T-Berd, right?`
`15:01.49`	`Vulture-`	`Manx: different locations`
`15:01.57`	`Vulture-`	`correct, test set sorry`
`15:02.39`	`ManxPower`	`Vulture-: pastebin the output of find /lib/modules -name "zaptel.*" -exec ls -l \{\} \;`
`15:02.43`	`Vulture-`	`I don't really see any debug info on the PRI`
`15:03.12`	`Vulture-`	`http://www.pastebin.ca/375687`
`15:03.28`	`tzanger`	`[TK]D-Fender: I've tried a half dozen incarnations of the device.prov.user/poassword thing`
`15:03.32`	`tzanger`	`gonna contact polycom`
`15:03.41`	`tzanger`	`this is the only downside to these phones... documentation's there but not specific`
`15:03.43`	`\|Vulture\|`	`stupid IRC client`
`15:03.51`	`\|Vulture\|`	`ManxPower: I am still here`
`15:04.23`	`[TK]D-Fender`	`tzanger: Well thats the first thing I've seen that isn't pretty well documented....`
`15:04.34`	`tzanger`	`nah`
`15:04.38`	`tzanger`	`all of the config entries are vague`
`15:04.39`	`ManxPower`	`\|Vulture\|: mv /lib/modules/2.6.9-42.0.8.ELsmp/extra/zaptel.ko /tmp`
`15:04.54`	`tzanger`	`i.e. you don't know if you need a container, and if you do, you don't know if you repeat the container name in the leement name`
`15:04.55`	`[TK]D-Fender`	`\|Vulture\|: And about that "ztcfg -vvvv" ?`
`15:04.59`	`ManxPower`	`mv /lib/modules/2.6.9-11.ELsmp/extra/zaptel.ko /tmp`
`15:05.33`	`\|Vulture\|`	`http://www.pastebin.ca/375690`
`15:05.40`	`[TK]D-Fender`	`tzanger: I understand... its that this is a FIRST of its kind in my experience. Pretty much everything else they documented is spot-on`
`15:05.47`	`tzanger`	`:-) ah vell`
`15:05.56`	`[TK]D-Fender`	`tzanger: Oi`
`15:06.02`	`tzanger`	`[TK]D-Fender: I appreciate your efforts in locating that though, I'll go bug my polycom guy`
`15:06.10`	`\|Vulture\|`	`ManxPower: both moved, restart * and wanrouter?`
`15:06.13`	`ManxPower`	`\|Vulture\|: after the mv's stop wanrouter confirm zaptel is not loaded with an lsmod`
`15:06.16`	`tzanger`	`[TK]D-Fender: reseeller pricing on polycoms is insane .. wowza`
`15:06.18`	`ManxPower`	`then start wanrouter again`
`15:06.20`	`Daejeo1`	`ManxPower:??????????????????`
`15:06.35`	`[TK]D-Fender`	`\|Vulture\|: Whats your actual error again? What do you get in pri debug span 1"?`
`15:06.52`	`[TK]D-Fender`	`tzanger: Low or high?`
`15:06.55`	`\|Vulture\|`	`ManxPower: zaptel is still running, force it to stop?`
`15:07.01`	`tzanger`	`[TK]D-Fender: stupidly low`
`15:07.07`	`tzanger`	`like CAD$160 for a 501`
`15:07.27`	`\|Vulture\|`	`[TK]D-Fender: I get PRI up/PRI up/PRI up/PRI up/PRI down/"chan_zap.c:2438 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway!"`
`15:07.30`	`ManxPower`	`\|Vulture\|: rmmod zaptel.ko`
`15:07.43`	`\|Vulture\|`	`ERROR: Module zaptel is in use by zttranscode`
`15:07.53`	`ManxPower`	`\|Vulture\|: rmmod zttranscode`
`15:08.01`	`\|Vulture\|`	`okay both removed`
`15:08.16`	`ManxPower`	`and lsmod confirms this?`
`15:08.17`	`\|Vulture\|`	`and confirmed with lsmod`
`15:08.27`	`ManxPower`	`ok, now start wanrouter and asterisk`
`15:08.49`	`\|Vulture\|`	`ztcfg -vv checks out`
`15:09.35`	`[TK]D-Fender`	`\|Vulture\|: Whats your telco have to say about their D-Chan?`
`15:09.41`	`\|Vulture\|`	`PRI showing "Status: Provisioned, Down, Active" no data from the PRI yet`
`15:10.10`	`\|Vulture\|`	`[TK]D-Fender: they say it is fine, because their tech used his testing device to make outbound and rx inbound calls`
`15:10.19`	`ManxPower`	`\|Vulture\|: you are SURE you rebuilt asterisk and libpri after installing the latest zaptel.`
`15:10.21`	`*** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93)`
`15:10.28`	`\|Vulture\|`	`ManxPower: possitive`
`15:10.32`	`\|Vulture\|`	`I will do it again though`
`15:10.46`	`ManxPower`	`\|Vulture\|: I have no more suggesntions then`
`15:10.55`	`\|Vulture\|`	`is it possible to uninstall libpri/zaptel?`
`15:11.07`	`[[blah]asfd`	`i have a group of phones that are hearing the hold music periodically while takling to people on the phone... sounds like bleed over almost.`
`15:11.09`	`\|Vulture\|`	`just so I can clean it out and start fresh`
`15:11.18`	`[[blah]asfd`	`as well as other calls`
`15:11.23`	`[[blah]asfd`	`what can I do to stop this?`
`15:11.44`	`\|Vulture\|`	`ManxPower: I am going to see if the new cable fixes this (Fingers crossed)`
`15:11.53`	`*** join/#asterisk shinux__ (n=shinux@196.1.179.225)`
`15:11.54`	`\|Vulture\|`	`oh got a debug!`
`15:12.02`	`\|Vulture\|`	`http://www.pastebin.ca/375698`
`15:12.39`	`\|Vulture\|`	`I am not sure what that debug means though`
`15:13.11`	`ManxPower`	`\|Vulture\|: IT means you are not seeing a D-Channel`
`15:13.27`	`\|Vulture\|`	`oh okay`
`15:13.39`	`[TK]D-Fender`	`\|Vulture\|: Sanity check - pastebin wanpipe1.conf please`
`15:13.42`	`\|Vulture\|`	`well another hr and I will know if it is the cable or not`
`15:14.08`	`\|Vulture\|`	`http://www.pastebin.ca/375700`
`15:14.25`	`\|Vulture\|`	`btw ManxPower and [TK]D-Fender thanx for the assistance`
`15:14.37`	`[[blah]asfd`	`i thought that cross talk was something I would never have again having left pots lines. this group of phones is fed via sip`
`15:15.10`	`*** join/#asterisk kikoafonso (n=rafonso@cronopio.rits.org.br)`
`15:15.17`	`[TK]D-Fender`	`\|Vulture\|: Looks good, but try setting dchan in ther to "0", restart wanrouter & *.`
`15:15.19`	`*** join/#asterisk MarkWD (n=Mark@rrcs-67-78-88-186.sw.biz.rr.com)`
`15:16.06`	`MarkWD`	`[TK]D-Fender: can you spam the wiki address ?`
`15:16.12`	`[TK]D-Fender`	`~wiki`
`15:16.16`	`[TK]D-Fender`	`~wikis`
`15:16.18`	`jbot`	`extra, extra, read all about it, wikis is http://www.voip-info.org`
`15:16.26`	`MarkWD`	`thanks`
`15:16.31`	`wunderkin`	`~tikiwiki`
`15:16.53`	`\|Vulture\|`	`JESUS`
`15:16.59`	`[[blah]asfd`	`where?`
`15:17.00`	`aydiosmio`	`HEY_ZOOS`
`15:17.01`	`\|Vulture\|`	`[TK]D-Fender: I owe you a beer`
`15:17.07`	`[TK]D-Fender`	`:D`
`15:17.10`	`\|Vulture\|`	`a big one`
`15:17.24`	`aydiosmio`	`[TK]D-Fender is the man`
`15:17.26`	`wunderkin`	`our darned (!@$@#!) key problem is back, even with 1.6.7... it was ok monday but tuesday i checked with one person and she says it started again a little bit.. this is on a NEW ip501.. skafjsdkfadsf`
`15:17.29`	`\|Vulture\|`	`thank you so much! man I should have checked there before`
`15:18.04`	`[TK]D-Fender`	`\|Vulture\|: Yeah, that can be a strange one to interpret for sure....`
`15:19.39`	`wunderkin`	`i only see things on the polycom website for resellers.. how can an average joe get polycom certified if they arent a reseller? what do they call it?`
`15:19.42`	`[[blah]asfd`	`I can understand cross talk on zap devices and such, but this is SIP`
`15:21.00`	`*** join/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net)`
`15:21.06`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`15:21.29`	`aydiosmio`	`[[blah]asfd: your call recording servers, were they voip channels?`
`15:21.29`	`wunderkin`	`yes, time for another one, the last one is full`
`15:21.43`	`*** join/#asterisk bigred (n=ian@nat-vlan0200.sat4.rackspace.com)`
`15:22.13`	`Chris-NB`	`hi`
`15:22.37`	`thinwires`	`I have a question, is there such a thing as an IAX IP phone? or are they SIP only, and if so then i need to configure my sip.conf right?`
`15:22.41`	`Chris-NB`	`if I have a hangup extension, in the cdr the dst is allways the hangup extension`
`15:22.48`	`[[blah]asfd`	`aydiosmio: what do you mean?`
`15:23.05`	`Dovid`	`thinwires: there are several of them`
`15:23.11`	`Dovid`	`google iaxcom`
`15:23.16`	`aydiosmio`	`were you recording analog calls for recording voip calls (extra cpu load)`
`15:23.17`	`Chris-NB`	`is there a possibility to save the original destination extension?`
`15:23.19`	`thinwires`	`David: ok thank you`
`15:23.29`	`aydiosmio`	`s/for/or`
`15:24.36`	`wunderkin`	`[TK]D-Fender, you aren't a reseller.. right?`
`15:24.38`	`JoNate`	`does a`
`15:24.43`	`JoNate`	`woops`
`15:25.20`	`JoNate`	`I wish I was smarter...`
`15:26.32`	`[TK]D-Fender`	`wunderkin: Not yet, but I'm either going to be, or the tech cert. (can't co-exist currently)`
`15:26.45`	`wunderkin`	`ah`
`15:27.02`	`[TK]D-Fender`	`thinwires: Yes there are IAX phones. They all SUCK however`
`15:27.15`	`aydiosmio`	`it's called inter-asterisk or a reason`
`15:27.29`	`thinwires`	`ok, maybe that was the wrong question to ask, Can anyone explain how difficult it is to get SIP phones working with NAT's`
`15:27.30`	`wunderkin`	`you have to be under a reseller though dont you?`
`15:27.41`	`[TK]D-Fender`	`thinwires: Easy for most`
`15:27.56`	`cpm`	`thinwires, because sip SUCKs`
`15:28.05`	`[TK]D-Fender`	`wunderkin: No, this isn't some psycho pyramid scheme...`
`15:28.13`	`cpm`	`sorry, that wasn't helpful`
`15:28.21`	`cpm`	`just digging at [TK]D-Fender`
`15:28.22`	`*** join/#asterisk santibiotico (n=santi@ip23498.bcn.altecom.net)`
`15:28.27`	`santibiotico`	`hi`
`15:28.46`	`[TK]D-Fender`	`cpm: And missing the mark squarely :)`
`15:28.53`	`santibiotico`	`is there any way to use an external smtp server for voicemail to email feature instead of using sendmail/etc..??`
`15:28.55`	`thinwires`	`well I ask because at my home I might have to dual NAT to get a phone in my room, running a wire would take a fort night`
`15:29.03`	`[TK]D-Fender`	`cpm: I didn't knock the PROTOCOL, just the hardware that uses it :)`
`15:29.05`	`cpm`	`yeah, but if you are flaming, it doesn't matter if you have a clue`
`15:29.10`	`cpm`	`Okay`
`15:29.28`	`wunderkin`	`[TK]D-Fender, maybe i read their pyramid requirements incorrectly then, trying to find it again :D`
`15:29.33`	`[TK]D-Fender`	`cpm: I know plenty of shitty SIP phones too...`
`15:29.35`	`[TK]D-Fender`	`~gs`
`15:29.36`	`jbot`	`well, gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice`
`15:29.44`	`[TK]D-Fender`	`^^^^`
`15:29.47`	`*** join/#asterisk Rick999 (n=rpulido_@adsl-074-164-111-083.sip.bct.bellsouth.net)`
`15:29.52`	`thinwires`	`lol`
`15:30.07`	`[[blah]asfd`	`I have been happy with the linksys spa942 sip phone`
`15:30.13`	`thinwires`	`well i'm looking at these Polycom's they look sexy..`
`15:30.33`	`[TK]D-Fender`	`[[blah]asfd: Yeah, Linksys are pretty stable, but "not worth it" in North America.`
`15:30.51`	`[[blah]asfd`	`what do you mean by "not worth it"`
`15:30.56`	`*** part/#asterisk dorel__ (n=liran@80.179.31.43.static.012.net.il)`
`15:30.59`	`aydiosmio`	`[[blah]asfd: were you recording analog calls or recording voip calls (extra cpu load)?`
`15:31.17`	`[[blah]asfd`	`calls from sip as well as t1 on zap card`
`15:31.22`	`santibiotico`	`is there any way to use an external smtp server for voicemail to email feature instead of using sendmail/etc..??`
`15:31.22`	`[TK]D-Fender`	`[[blah]asfd: They offer nothing special at all, and have inferior call handling and low LCD readability.`
`15:31.26`	`aydiosmio`	`oh okay, thanks`
`15:31.45`	`[TK]D-Fender`	`[[blah]asfd: For the fact they are too closely price to Polycom and the Aastra 480i`
`15:32.01`	`[TK]D-Fender`	`[[blah]asfd: Which both severly thrash it`
`15:32.13`	`aydiosmio`	`dude, totally`
`15:32.14`	`[[blah]asfd`	`[TK]D-Fender: wow... I have had the exact opposite experience. as compared to the polycom sp 501`
`15:32.28`	`[[blah]asfd`	`same price, better quality`
`15:32.30`	`[TK]D-Fender`	`[[blah]asfd: Should have asked for help on it earlier....`
`15:32.42`	`[[blah]asfd`	`and they dont take 5 minutes to reboot`
`15:33.03`	`[TK]D-Fender`	`[[blah]asfd: Don't think you'll find anyone but our "Sacrifice" here to attest to your POV.`
`15:33.20`	`aydiosmio`	`muahaha`
`15:33.25`	`[TK]D-Fender`	`[[blah]asfd: And typically my phones never NEED to reboot. actual time is more like 2 mins`
`15:33.42`	`[[blah]asfd`	`reboot required when making setting changes...`
`15:33.47`	`[[blah]asfd`	`company is always making changes`
`15:33.52`	`[TK]D-Fender`	`[[blah]asfd: To what?`
`15:34.03`	`[TK]D-Fender`	`[[blah]asfd: How often do your regs change?`
`15:34.06`	`*** join/#asterisk siddu999 (n=siddu999@adsl-074-164-111-083.sip.bct.bellsouth.net)`
`15:36.55`	`[[blah]asfd`	`[TK]D-Fender: about once a month.`
`15:36.58`	`Strom_M`	`in my experience, if you're always fiddling with the phone system, then you never made the initial effort to adequately assess the needs of the client and engineer the system to meet those needs`
`15:36.59`	`TJBraza`	`Fender dude`
`15:37.01`	`[TK]D-Fender`	`[[blah]asfd: Linksys does factor in to the "consideration list" though, ust further down for their lack of features`
`15:37.03`	`[TK]D-Fender`	`~phones`
`15:37.08`	`jbot`	`somebody said phones was at http://bani.anime.net/phones/, or is In order of quality: Polycom (Any), Aastra 480i, Cisco 7940+, Linksys SPA-94x`
`15:37.22`	`TJBraza`	`what exactly are the benefits of a polycom over a linksys one, in general?`
`15:37.31`	`\|Vulture\|`	`Polycom IP501 is my fav ;)`
`15:38.25`	`thinwires`	`I'm looking at the IP501's now... we're going to buy three of them, it looks sound and almost everyone that has them likes them`
`15:38.43`	`[TK]D-Fender`	`TJBraza: Superior call handling, better audio quality, larger screen (IP 501+), MicroBrowser (IP 501+), All phones have a passthrough port, massive configurability`
`15:39.04`	`[TK]D-Fender`	`TJBraza: Presence support.... I could go on....`
`15:39.18`	`Strom_M`	`I'm less impressed with the 480i than I am with the 7940/7960`
`15:39.27`	`thinwires`	`is voipsupply.com a good place to purchse phone gear? (they are like a 15 minute drive away from me :-)`
`15:39.35`	`Strom_M`	`the 480i isn't a bad phone, but it has a cheapy feel about it`
`15:39.39`	`[TK]D-Fender`	`\|Vulture\|: IP 501 is nice, but I'd prefer an IP 650. Mind you you'd never see me PAY for it ;)`
`15:40.01`	`[TK]D-Fender`	`Strom_M: It is a bit lower on feel & audio quality, but HUGE on functionality.`
`15:40.03`	`\|Vulture\|`	`are the 650s in color?`
`15:40.19`	`thinwires`	`greyscale`
`15:40.25`	`\|Vulture\|`	`we use to use 7960s but dumped them in favor of the ip501`
`15:40.32`	`\|Vulture\|`	`never messed with a 650`
`15:40.40`	`[TK]D-Fender`	`\|Vulture\|: nope, just backlit`
`15:40.40`	`Strom_M`	`yeah, but physical build and audio quality are a fairly major factor with my clients`
`15:40.45`	`wunderkin`	`[TK]D-Fender, do you have 5 fingers?`
`15:40.47`	`wunderkin`	`ha ha ha ha`
`15:40.48`	`\|Vulture\|`	`ah backlit is nice`
`15:40.58`	`[TK]D-Fender`	`wunderkin: 8 actually ;)`
`15:41.12`	`\|Vulture\|`	`tried the 301.. but just didn't like it`
`15:41.22`	`*** join/#asterisk dasenjo (n=dasenjo@190.24.177.189)`
`15:41.29`	`[TK]D-Fender`	`wunderkin: My discount structure works in a differnt manner ;)`
`15:41.47`	`wunderkin`	`oh.. fall off of the truck discount :-)`
`15:41.49`	`\|Vulture\|`	`man 12 lines...`
`15:41.53`	`\|Vulture\|`	`sexy`
`15:41.58`	`thinwires`	`lol`
`15:42.14`	`\|Vulture\|`	`that would be nice for testing`
`15:42.22`	`*** join/#asterisk codefreeze (n=steve_mu@216.166.159.235)`
`15:42.30`	`[TK]D-Fender`	`wunderkin: no the "resell as new near/at full market value when actually originally purchased at wholesale" ;)`
`15:42.50`	`wunderkin`	`heh`
`15:43.20`	`[TK]D-Fender`	`\|Vulture\|: Keep in mind thats 12 REGISTRATIONS. That actually capable of handling a SIC number of calls.`
`15:44.08`	`\|Vulture\|`	`well one thing I liked about my 7960 is that I had 6 lines and could keep a registration in multiple locations for testing`
`15:44.09`	`tzanger`	`SIC?`
`15:45.00`	`thinwires`	`that 7960 is expensive though eh?`
`15:45.29`	`\|Vulture\|`	`next to the ip501 it is`
`15:45.42`	`wunderkin`	`i think we need to go with grandstream budgetone phones... :-D`
`15:45.49`	`thinwires`	`yeah, it is, but it supports more lines and such`
`15:45.57`	`tzanger`	`I want polycom to make a wifi phone with bluetooth`
`15:45.59`	`tzanger`	`that's all`
`15:46.10`	`Strom_M`	`wunderkin: no no, i propose tooling up to do to grandstream what grandstream is doing to polycom`
`15:46.13`	`tzanger`	`no color screen, some programma ble soft buttons... nothing much`
`15:46.20`	`JT`	`wifi is a bad idea anyway`
`15:46.22`	`tzanger`	`maybe a vibrate and a decent set of NORMAL ringtones`
`15:46.26`	`tzanger`	`JT: compred to what`
`15:46.31`	`JT`	`dect`
`15:46.33`	`Strom_M`	`we can make phones out of hair-thin strands of gossamer moonlight`
`15:46.40`	`tzanger`	`JT: dect's not really available in north america`
`15:46.46`	`JT`	`or similar`
`15:46.46`	`tzanger`	`although I have contemplated bringing in DECT phones`
`15:46.50`	`JT`	`just not wifi`
`15:47.12`	`tzanger`	`meh, I find it's passable`
`15:47.12`	`Strom_M`	`jt: why not wifi? in theory, you can create a wifi network with wide coverage`
`15:47.17`	`thinwires`	`what's wrong with wifi?`
`15:47.18`	`tzanger`	`and easy to cover`
`15:47.48`	`JT`	`erm`
`15:47.53`	`JT`	`the whole mobile terminals`
`15:48.00`	`JT`	`with variable lag, jitter and packet loss`
`15:48.02`	`tzanger`	`wifi's jittery but honestly it's fine enough`
`15:48.12`	`tzanger`	`JT: so it's basically the internet but confined to your office`
`15:48.58`	`JT`	`no`
`15:49.17`	`JT`	`the Internet doesn't have wireless terminals moving about the place in the backbone, usually`
`15:49.42`	`thinwires`	`so with the IP501, those soft buttons, I'm assuming those are programable buttons?`
`15:50.08`	`tzanger`	`JT: minor problem`
`15:50.26`	`*** join/#asterisk ars247 (n=no@64-142-43-180.dsl.static.sonic.net)`
`15:50.43`	`JT`	`...`
`15:51.04`	`Dr-Linux`	`hi guys`
`15:51.16`	`Dr-Linux`	`question, how can i hangup this channel:`
`15:51.17`	`Dr-Linux`	`Channel Location State Application(Data)`
`15:51.17`	`Dr-Linux`	`Local/45001@users-bd 45001@users:3 Ring Queue(mcp-support\|tT\|\|\|600)`
`15:51.42`	`macTijn`	`soft hangup channel Local/45001@whatever`
`15:51.43`	`macTijn`	`on console`
`15:51.56`	`*** join/#asterisk rdb_ (n=rdb@gw.avila.edu)`
`15:51.58`	`*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)`
`15:52.40`	`Dr-Linux`	`macTijn: that i aready tried:`
`15:52.41`	`Dr-Linux`	`ivr1*CLI> soft hangup Local/45005@users-f6`
`15:52.41`	`Dr-Linux`	`Local/45005@users-f6 is not a known channel`
`15:53.05`	`[TK]D-Fender`	`thinwires: Depends. The soft-keys under the screen? No. The line keys? Sure. for call handling, or speed-dials (w/ presence support)`
`15:54.15`	`[TK]D-Fender`	`Dr-Linux: thats because you don't have the full channel name`
`15:54.19`	`*** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com)`
`15:54.26`	`JoNate`	`and its a different number`
`15:55.52`	`Dr-Linux`	`[TK]D-Fender: that's all i have. So what's the solution to kill them?`
`15:56.00`	`*** join/#asterisk topping (n=topping@204.152.96.238)`
`15:56.06`	`[TK]D-Fender`	`Dr-Linux: Get better info`
`15:56.19`	`file`	`tab complete the channel name?`
`15:56.49`	`Dr-Linux`	`file: channel name is complete, and there not one, but about 8 channel hanged`
`15:56.55`	`Dr-Linux`	`now i wanna kill them`
`15:56.55`	`siddu999`	`Hello all, I am having trouble with outbound calls on asteriskNOW. Can anyone give me some help pls?`
`15:57.00`	`Strom_M`	`press the "LOL" button`
`15:57.09`	`Dr-Linux`	`usually i do as "soft hangup" but that doesn't seems to work`
`15:57.12`	`ManxPower`	`siddu999: no.`
`15:57.18`	`thinwires`	`suddy999: /join #asterisknow`
`15:57.21`	`ManxPower`	`siddu999: because this is not the #asterisknow channel`
`15:57.22`	`Strom_M`	`it's right between "OMG" "ROFL" and "any"`
`15:57.24`	`Dr-Linux`	`LOL button? :S`
`15:57.25`	`wunderkin`	`after that press the any key!`
`15:57.27`	`file`	`that is not a complete chan_local channel name`
`15:57.55`	`siddu999`	`Thanks guys. I will join on asteriskNOW.`
`15:57.59`	`thinwires`	`Where's the "ANY" key?`
`15:58.13`	`aydiosmio`	`next to the abort key`
`15:58.23`	`wunderkin`	`and eject`
`15:58.25`	`thinwires`	`abortion is murder.`
`15:58.30`	`thinwires`	`lol jk`
`15:58.33`	`aydiosmio`	`yarly`
`15:58.33`	`Dr-Linux`	`file: have a look here: http://phpfi.com/209763`
`15:58.47`	`Dr-Linux`	`these channel got hanged due to internet issue`
`15:58.56`	`[TK]D-Fender`	`"Guns don't kill people... I kill people"`
`15:59.00`	`file`	`like I said, those are not the complete names.. just hit tab to do tab completion`
`15:59.00`	`Dovid`	`haha`
`15:59.08`	`Dr-Linux`	`but i wanna kill them, i'm looking for any solution except restart asterisk`
`15:59.16`	`Dr-Linux`	`aww`
`15:59.17`	`aydiosmio`	`okay how about this, how can I start a timer from when a Dial() Answers to when it hangs up and retrieve that value?`
`15:59.18`	`Dr-Linux`	`ok`
`15:59.20`	`Dr-Linux`	`lemme try`
`15:59.24`	`[TK]D-Fender`	`Dr-Linux: Do I really have to say it again. You Do Not See The Full Channel Name There <---`
`15:59.35`	`tuan_modulis`	`Dr-Linux: try service asterisk restart`
`15:59.35`	`file`	`this is not rocket science.`
`15:59.37`	`thinwires`	`i <3 tab completion`
`15:59.38`	`Dovid`	`doc: what do u get from show channels verbose ?`
`15:59.38`	`tuan_modulis`	`oops didnt read`
`15:59.43`	`aydiosmio`	`Dr-Linux, more like Mr-Linux amirite?`
`15:59.43`	`Dovid`	`hehe`
`16:00.06`	`[TK]D-Fender`	`aydiosmio: "Physician heal thyself" <-`
`16:00.10`	`Dovid`	`certain people should be banned from using computers and technology in general (like a few clients of mine)`
`16:00.24`	`wunderkin`	`file, zOmg.. damn... puts away the rocket science book`
`16:00.35`	`Dr-Linux`	`well, i can see the full channel name but ..`
`16:00.36`	`Dr-Linux`	`hhm..`
`16:00.37`	`Dr-Linux`	`ivr1*CLI> soft hangup Local/4040@users-1cfd,1`
`16:00.37`	`Dr-Linux`	`Requested Hangup on channel 'Local/4040@users-1cfd,1'`
`16:00.37`	`Dr-Linux`	`ivr1*CLI>`
`16:00.41`	`[TK]D-Fender`	`Dovid: Yes, for "crimes against technology"`
`16:00.45`	`Dovid`	`haha`
`16:01.03`	`TaiSHi`	`Bleh`
`16:01.06`	`TaiSHi`	`Use christian linux`
`16:01.10`	`file`	`that means the channel is hung, you'll have to restart... how did the agents login?`
`16:01.12`	`Dovid`	`i got a list of stories - like a friend that had "issues with his wireless" it wasnt plugged in - happend last week`
`16:01.21`	`TaiSHi`	`It has no kill, no abort, etc`
`16:01.34`	`Dr-Linux`	`file: agents logged in through an application`
`16:01.38`	`coppice`	`does christian linux evolve?`
`16:01.56`	`wunderkin`	`:-)`
`16:02.00`	`file`	`Dr-Linux: didn't exactly answer the question... are they dynamic queue members, callback agents, what?`
`16:02.16`	`wunderkin`	`maybe it spawns into other distros`
`16:02.22`	`Dr-Linux`	`file: they are callback agents`
`16:02.43`	`file`	`Dr-Linux: that could be the reason why.`
`16:03.00`	`coppice`	`I would have thought its development was probably slowed by sex`
`16:03.14`	`Dovid`	`TK: have a look at this. it was on the users list a while back. I wanted to yell RTFM !!!`
`16:03.15`	`Dovid`	`http://pastebin.ca/375749`
`16:03.20`	`Dr-Linux`	`file: actually the reason i'm asking here, to understand the issue so i can do better in future`
`16:03.37`	`Dovid`	`doc its ok. we goto rip on some one - i get the abuse too ;)`
`16:03.53`	`TaiSHi`	`coppice: :O`
`16:04.01`	`TaiSHi`	`I think they don't use pipe either...`
`16:04.05`	`TaiSHi`	`It would mean too much contact`
`16:04.39`	`Dr-Linux`	`i restarted asterisk and everything is fine now`
`16:05.14`	`*** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il)`
`16:05.45`	`TaiSHi`	`Did you kill it ?`
`16:05.57`	`*** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse)`
`16:07.14`	`Dr-Linux`	`TaiSHi: no`
`16:07.24`	`Dr-Linux`	`TaiSHi: as i said, i restarted asterisk`
`16:07.33`	`Dr-Linux`	`and for sure that could fix my issue`
`16:07.43`	`Dr-Linux`	`but i don't like restarting solutions`
`16:08.01`	`tuan_modulis`	`it's not exactly atomic...`
`16:08.46`	`tuan_modulis`	`in my case, i have credit cards involved`
`16:08.59`	`tuan_modulis`	`so it;s like.... charged to the max`
`16:09.02`	`tuan_modulis`	`heheh`
`16:13.07`	`tuan_modulis`	`but i'll have to test it out more`
`16:13.21`	`tuan_modulis`	`i think the hangup is reliable`
`16:13.41`	`*** join/#asterisk Giofe (n=opera@190.81.4.169)`
`16:15.08`	`TaiSHi`	`Dr-Linux: ... be careful, you can't kill it >.< or abort!`
`16:16.08`	`*** join/#asterisk jarg (n=jarg@200.56.225.61)`
`16:16.31`	`Dr-Linux`	`S:`
`16:16.50`	`*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)`
`16:18.07`	`*** join/#asterisk Seyr (i=user@cpe-67-10-136-212.houston.res.rr.com)`
`16:18.18`	`*** join/#asterisk Bazy (n=bazy@eclipse.upcnet.ro)`
`16:19.00`	`TaiSHi`	`Can a user have multiple alias ?`
`16:19.22`	`Seyr`	`Anyone know why when recording a phone call, you only capture the extensions side of the audio? when calling extension to extension, it records both sides fine, but when talking to an outside person, it only captures the extensions side of the audio`
`16:19.28`	`TaiSHi`	`Like "Debora" and "843" (internal number) and 398493 (external number)`
`16:20.17`	`*** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com)`
`16:20.38`	`aydiosmio`	`ah sweet, I got my setup in just a few dialplan lines... I love asterisk`
`16:21.14`	`TheCops`	`Someone had problem with SIP TO SIP audio quality (one-way audio etc) ? I have a SIP tunnel between a tekelec 7000 and Asterisk, on the other side I have 1 phone Polycom ip601 to my Asterisk.`
`16:21.36`	`*** join/#asterisk Dimik (n=Dimik@unaffiliated/dimik)`
`16:22.27`	`jarg`	`heya all, i have a problem when i outgoing calls from broadsoft (another voip box) i do it using a softphone and all works well, but when i send the call from * it doen;t work, i can see that my invite is diferent, from the softphone i have: From: "MYACCOUNT" <sip:MYACCOUNT@MYDOMAIN> and To: NUMBERTOCALL@MYDOMAIN, from my * i have From: "NUMBERTOCALL" <sip:MYACCOUNT@MYDOMAIN> and To: NUMBERTOCALL@IP:PORT`
`16:22.44`	`TaiSHi`	`Can a user have multiple alias ? Like "Debora" and "843" (internal number) and 398493 (external number)`
`16:22.52`	`jarg`	`the broadsoft people said me that i need send the DOMAIN in the TO`
`16:23.04`	`jarg`	`the question is, how i can do it in *`
`16:23.05`	`jarg`	`?`
`16:24.10`	`simplexio`	`im "using" asterisk realtime wiht postgresql, any idea can i read somehow is sip user registered or not from database. far as i can see it dosent touch database if regester expires`
`16:24.37`	`nfi\|ermes`	`when i logon in asterisk-gui, asterisk goes to segmentation fault: dbg of core dump ---> http://pastebin.com/890614`
`16:25.25`	`*** join/#asterisk toot (n=toot@84.19.255.123)`
`16:25.26`	`TaiSHi`	`nfi\|ermes: Still ?`
`16:27.20`	`JT`	`jarg: fromdomain= ?`
`16:27.24`	`nfi\|ermes`	`yes`
`16:28.21`	`toot`	`hey folks - just wondering anyone taken the dCAP - i can't find much in the way of example info or syllabus for it`
`16:28.35`	`TheCops`	`Someone had problem with sip to sip audio quality?`
`16:28.48`	`jarg`	`JT: fromdomain modify the domain from the From, no from the To`
`16:28.59`	`JT`	`relm= ?`
`16:29.10`	`jarg`	`ok, let me try, thanks`
`16:29.17`	`JT`	`i'm going off memory`
`16:29.19`	`JT`	`i might be off`
`16:30.14`	`eald`	`realm?`
`16:30.27`	`JT`	`probably`
`16:30.40`	`JT`	`it's 3am here`
`16:31.35`	`Dovid`	`3am ? u in aus land?`
`16:31.56`	`JT`	`otherwise known as "Australia"`
`16:32.00`	`Dovid`	`hehe`
`16:32.08`	`JT`	`what is it with people making up strange names for Australia`
`16:32.32`	`*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)`
`16:32.55`	`Nugget`	`"aus" isn't a "strange name." it's merely a tacit admission by David that he doesn't know how to spell "Australia." Like when people say "def" because they're afraid to type "definitely."`
`16:33.23`	`simplexio`	`beware tyops`
`16:33.30`	`JT`	`haha`
`16:33.32`	`Dovid`	`yup. i never finishe high school`
`16:33.38`	`simplexio`	`:)`
`16:33.38`	`*** join/#asterisk Strom_M (n=strom@63.110.13.126)`
`16:33.47`	`Dovid`	`finished*`
`16:34.02`	`aydiosmio`	`the best course of action is to use .au`
`16:34.03`	`simplexio`	`australia vs austria is another common mistake`
`16:34.06`	`aydiosmio`	`it's cool`
`16:34.16`	`JT`	`that's such a poor excuse, not finishing high scool :P`
`16:34.20`	`aydiosmio`	`how anyone can confuse those countries is beyond me`
`16:35.00`	`vlt`	`Hello. I defined a [peerB] in peerA's iax.conf and a [usernameA] section on "peerB" (both type friend). I can place calls from A on B, but can't register to receive calls: "Registration of 'usernameA' rejected: 'Registration Refused' from: '84.179.52.xxx'". On peerB I get "No registration for peer 'usernameA' (from 87.234.124.xxx)". How is this possible?`
`16:35.43`	`simplexio`	`well i have heard this tale about students from USA who wanted spend year in warm australia and see kengurus, but ended few thousand kilometer away.`
`16:35.49`	`TaiSHi`	`Can a user have multiple alias ? Like "Debora" and "843" (internal number) and 398493 (external number)`
`16:36.01`	`ManxPower`	`vlt: you have register => peerB:password@ip.of.peer.a on Peer B`
`16:36.02`	`JT`	`simplexio: wow, that really is pretty dumb`
`16:36.09`	`JT`	`kangaroos`
`16:36.18`	`aydiosmio`	`and KOALAHS`
`16:36.28`	`JT`	`argh :P`
`16:36.38`	`aydiosmio`	`and DINGOHES`
`16:36.46`	`ManxPower`	`TaiSHi: Yes, multiple exten => lines pointing to the same device`
`16:36.47`	`JT`	`BHEARS`
`16:36.50`	`JT`	`RHACEWNS`
`16:37.10`	`*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)`
`16:37.23`	`TaiSHi`	`Great`
`16:37.32`	`TaiSHi`	`And now... the conclusion`
`16:37.32`	`aydiosmio`	`oh oh an d RATTLAHS`
`16:37.35`	`vlt`	`ManxPower: hmmm, I tried "register => usernameA:pw@peerB" on A`
`16:37.35`	`TaiSHi`	`Ok, that's from Heroes`
`16:37.57`	`*** part/#asterisk Giofe (n=opera@190.81.4.169)`
`16:38.05`	`ManxPower`	`vlt: and you have a [usernameA] on Peer B's sip.conf?`
`16:38.06`	`vlt`	`ManxPower: @ip.of.peerB*`
`16:38.22`	`vlt`	`ManxPower: Yes. iax.conf.`
`16:38.36`	`ManxPower`	`asterisk uses [whateverisinhere] as the expected incoming userID for registratons and calls`
`16:38.53`	`TaiSHi`	`If I have 13 lines, how do I make that when someone wants to call to the outside, it passes an if (to check region) and select the available line(s) (I have more than 1 line for each city)?`
`16:38.56`	`vlt`	`ManxPower: that's why A can already place calls there.`
`16:39.36`	`ManxPower`	`TaiSHi: see the group= option in /etc/asterisk/zapata.conf for PSTN / Zap ports`
`16:39.47`	`TaiSHi`	`Mmm`
`16:39.57`	`TaiSHi`	`I have 2 FXO and 11 VoIP`
`16:40.10`	`ManxPower`	`TaiSHi: you have 11 voip providers?`
`16:40.18`	`TaiSHi`	`11 VoIP lines...`
`16:40.23`	`TaiSHi`	`3 diff providers u_U`
`16:41.26`	`ManxPower`	`There is no such thing as a "voip line"`
`16:41.41`	`JacksLivr`	`anyone know how to make x-lite not wake up Anna Nicole Smith everytime it rings? No matter how low i turn the volume or if i plug headphones in, the PC speaker deafens me.`
`16:42.04`	`TaiSHi`	`Well`
`16:42.12`	`TaiSHi`	`I have 11 HandyTone 486 devices`
`16:42.17`	`TaiSHi`	`With 3 different providers`
`16:42.34`	`pigpen`	`There is no such thing as a SIP trunk either. :)`
`16:42.47`	`JacksLivr`	`the tooth fairy is also a myth`
`16:42.48`	`pigpen`	`ManxPower, I couldn't get it to work to save my life.`
`16:43.09`	`pigpen`	`dialout fine.....inbound couldn't register.`
`16:43.18`	`vlt`	`ManxPower: O damn, sorry, my fault. I still got type=user on B ...`
`16:43.43`	`ManxPower`	`TaiSHi: so you have 3 providers and 2 PSTN lines.`
`16:43.53`	`TaiSHi`	`Yes`
`16:44.09`	`ManxPower`	`can all phones place calls thru any provider or line?`
`16:44.31`	`TaiSHi`	`We're using hardphones now`
`16:44.37`	`TaiSHi`	`There is 1 phone per handytone`
`16:44.43`	`TaiSHi`	`So 11 phones to VoIP`
`16:44.47`	`ManxPower`	`that was not my question.`
`16:45.12`	`TaiSHi`	`All phones can call wherever they want`
`16:45.26`	`aydiosmio`	`How do I get the time between Answer and Hangup on a Dial()?`
`16:45.28`	`ManxPower`	`good, just set up your dialplan accordingly`
`16:45.40`	`TaiSHi`	`That's why I'm worried...`
`16:45.45`	`aydiosmio`	`or the absolute time/date`
`16:45.49`	`ManxPower`	`aydiosmio: billseconds in the CDR`
`16:45.52`	`TaiSHi`	`I dont want someone to call Barcelona from a Mexico line`
`16:46.09`	`ManxPower`	`TaiSHi: "mexico line"?`
`16:46.27`	`TaiSHi`	`Well, our VoIP providers give us different countries`
`16:46.31`	`TaiSHi`	`I mean`
`16:46.40`	`ManxPower`	`TaiSHi: you use your dialplan to match specific dialed numbers to send the call out the required provider.`
`16:46.40`	`TaiSHi`	`We have lines for each country/region`
`16:46.47`	`TaiSHi`	`I thought of that`
`16:46.53`	`TaiSHi`	`Now...`
`16:46.53`	`ManxPower`	`TaiSHi: if you insist on using the wrong terms I cannot help you.`
`16:47.03`	`TaiSHi`	`Ok, I'm not pro :P`
`16:47.08`	`ManxPower`	`The only kind of line you have are the telco lines.`
`16:47.22`	`TaiSHi`	`Okay, the Mexico VoIP phone`
`16:47.42`	`ManxPower`	`Why would you not want the mexico phone to be able to call barcilona?`
`16:47.52`	`TaiSHi`	`Because we have a Barcelona phone`
`16:48.19`	`ManxPower`	`TaiSHi: one of us is terribly confused.`
`16:48.33`	`TaiSHi`	`Who?!`
`16:48.41`	`JT`	`ManxPower: cheaper rates to different countries with different providers`
`16:48.44`	`JT`	`dialplan`
`16:48.46`	`JT`	`easy done`
`16:48.52`	`TaiSHi`	`Great :P`
`16:48.59`	`TaiSHi`	`Now, let's move to the next part`
`16:49.21`	`TaiSHi`	`I have 3 phones (2 VoIP 1 analog) for the same city`
`16:49.38`	`TaiSHi`	`How can I make that, if an operator calls`
`16:49.56`	`TaiSHi`	`And the first line is busy`
`16:50.01`	`TaiSHi`	`He will move to the next one`
`16:50.05`	`ManxPower`	`TaiSHi: you have phones (handytone + hardphone), you have telco lines, and you have voip providers`
`16:50.13`	`aydiosmio`	`ManxPower: I checked the CDR, it only has the billseconds of the incoming call, not the separate Dial`
`16:50.16`	`ManxPower`	`I assume you have telephone numbers`
`16:50.30`	`TaiSHi`	`I think we missed on the voip providers`
`16:50.32`	`ManxPower`	`aydiosmio: you need to find the cdr for the outgoing call`
`16:50.33`	`TaiSHi`	`I ment my ITSPs`
`16:50.47`	`TaiSHi`	`I have phones (HT+Hardphone) and 2 telco lines, yes`
`16:51.00`	`*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de)`
`16:51.05`	`aydiosmio`	`oh wait, I think I found it, there's no billseconds but there is two timestamps about the same duration as my last Dial`
`16:51.07`	`ManxPower`	`TaiSHi: you need to figure out what telephone numbers (or patterns) you want to send to each provider.`
`16:51.21`	`ManxPower`	`aydiosmio: see CDR.txt`
`16:51.22`	`aydiosmio`	`need to find the column names of Master.csv`
`16:51.25`	`aydiosmio`	`thx`
`16:51.29`	`TaiSHi`	`Yeah, dialplan to redirect calls for each country/region`
`16:51.34`	`TaiSHi`	`Now I ment something else`
`16:51.42`	`TaiSHi`	`I want that, if one phone is busy`
`16:51.49`	`TaiSHi`	`Call from another or give busy tone`
`16:52.05`	`ManxPower`	`TaiSHi: you need to disable call waiting on the handytones.`
`16:52.06`	`*** join/#asterisk gr1ncheux_ (n=devine@unaffiliated/gr1ncheux)`
`16:52.16`	`TaiSHi`	`HandyTones will not be used anymore`
`16:52.16`	`ManxPower`	`TaiSHi: see [macro-stdexten] in extensions.conf.sample`
`16:52.23`	`aydiosmio`	`ah [answer] and [end]`
`16:52.38`	`TaiSHi`	`Let me check`
`16:52.41`	`ManxPower`	`TaiSHi: you need to disable call waiting on whatever endpoint you are using.`
`16:52.47`	`*** join/#asterisk juanjoc (n=juanjoc@200.69.219.113)`
`16:52.54`	`TaiSHi`	`softphone (will use)`
`16:52.56`	`TaiSHi`	`X-lite`
`16:53.14`	`*** join/#asterisk drako (n=ljd@unaffiliated/luisjose)`
`16:53.42`	`TaiSHi`	`As far as I see`
`16:53.50`	`TaiSHi`	`[macro-stdexten] will do the job`
`16:53.53`	`TaiSHi`	`Thank you ManxPower`
`16:54.01`	`TaiSHi`	`Once again :)`
`16:54.14`	`*** join/#asterisk slayer192 (n=slayer19@pirus.securax.be)`
`16:56.07`	`*** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93)`
`16:56.15`	`TaiSHi`	`bb everyone, off to lunch-time`
`16:56.41`	`aydiosmio`	`is there a variable I can set in a channel that will be recorded to the CDR? like ${LOG-CUSTOM}`
`16:57.03`	`Qwell[]`	`CDR(userfield)`
`16:57.14`	`aydiosmio`	`bitchin.`
`16:57.22`	`*** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju)`
`16:59.31`	`Nugget`	`camaro.`
`16:59.43`	`*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)`
`17:00.23`	`cpm`	`I ran over my neighbor`
`17:00.52`	`*** join/#asterisk viperdude (n=jon@195.74.96.120)`
`17:03.06`	`ManxPower`	`aydiosmio: "show applications like cdr" in the Asterisk CLI`
`17:05.31`	`tzanger`	`oooh baby`
`17:05.34`	`tzanger`	`chan_cellphone fucking ROCKS`
`17:07.34`	`heh_v_water`	`If anyone has pretty good asterisk skills I was just contacted for work as an Asterisk analyst for a company in Lowell, MA, USA... if interested message me and I will paste you the information`
`17:08.16`	`aydiosmio`	`ManxPower: I'm using cdr_mysql, is it a big pain to add a new column and have * insert it or should I just put my data in an existing column?`
`17:08.22`	`tzanger`	`who's dbowerman on mantis? I owe him a big wet sloppy kiss`
`17:10.17`	`Corydon-w`	`tzanger: ME!!!`
`17:10.39`	`*** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)`
`17:10.40`	`tzanger`	`hahahaha`
`17:10.42`	`tzanger`	`you horny old goat`
`17:10.49`	`Corydon-w`	`rofl`
`17:11.06`	`tzanger`	`that was one of THE most painless bluetooth implementations I've ever seen`
`17:13.07`	`*** join/#asterisk marv[work] (n=timr@24.214.206.254)`
`17:13.25`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`17:14.47`	`eald`	`cdr_mysql doesn't support adding custom fields, someone here said that is easy to add your own columns in the cdr_mysql addon`
`17:15.47`	`*** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir)`
`17:16.48`	`aydiosmio`	`right, I saw something about having to recompile though`
`17:17.05`	`aydiosmio`	`still looking for a howto`
`17:17.25`	`eald`	`mm, recompile... if you are reading wiki then that is kinda old about the topic`
`17:17.36`	`*** join/#asterisk stkn\|work (n=stkn\|wor@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM]`
`17:18.04`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM]`
`17:18.24`	`eald`	`see, you downloads addons and yet you won't get cdr_mysql to use custom fields (I'm talking about adding others, not userfield)`
`17:18.37`	`anonymouz666`	`is there any problem running mpg123 different than 0.59r ?`
`17:18.47`	`*** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines)`
`17:18.49`	`eald`	`custom fields only works with csv backend for cdr`
`17:18.50`	`JunK-Y`	`anonymouz666: yes`
`17:18.52`	`*** part/#asterisk Seyr (i=user@cpe-67-10-136-212.houston.res.rr.com)`
`17:20.53`	`anonymouz666`	`JunK-Y: what problem?`
`17:21.08`	`anonymouz666`	`mpg123-0.59r does not work properly on x64 arch`
`17:21.23`	`*** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla)`
`17:21.23`	`*** mode/#asterisk [+o russellb] by ChanServ`
`17:22.08`	`aydiosmio`	`my Dial doesn't seem to be getting it's own CDR record`
`17:23.20`	`aydiosmio`	`maybe I need ForkCDR()?`
`17:23.24`	`JunK-Y`	`anonymouz666: use the native MOH?`
`17:23.29`	`*** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net)`
`17:24.36`	`*** join/#asterisk foobar778 (i=johhny@ip68-100-210-15.dc.dc.cox.net)`
`17:24.52`	`rbd`	`hey guys, simple question. I have two asterisk servers (no connection between then or anything via IAX, etc)... I have a call come into one (via SIP), and I'd like to transfer it to a SIP extension on the other server in an AGI script. Would I use the dialplan apps Transfer, Dial or what to do this?`
`17:25.29`	`*** join/#asterisk foobar778 (i=johhny@ip68-100-210-15.dc.dc.cox.net)`
`17:25.36`	`*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)`
`17:26.07`	`foobar778`	`[TK]D-Fender: u around`
`17:26.27`	`[TK]D-Fender`	`foobar778: .... maybe?`
`17:26.43`	`foobar778`	`ok have a question`
`17:26.57`	`[TK]D-Fender`	`foobar778: Antoher one? I just finished answering your first!`
`17:27.28`	`foobar778`	`ok Im using a did`
`17:27.48`	`foobar778`	`and having it go to disa`
`17:27.57`	`foobar778`	`then making an outbound call`
`17:28.06`	`foobar778`	`with voip provider`
`17:28.09`	`pigpen`	`here it comes....`
`17:28.11`	`*** join/#asterisk ronki (i=PJIRCWeb@ALyon-256-1-61-232.w90-9.abo.wanadoo.fr)`
`17:28.14`	`pigpen`	`:)`
`17:28.18`	`ronki`	`hi everyone`
`17:28.21`	`*** join/#asterisk angel--- (n=Dark@195.251.166.34)`
`17:28.32`	`[TK]D-Fender`	`pigpen: What pray-tell? :)`
`17:28.42`	`ronki`	`I have a question if someone could help me`
`17:29.01`	`pigpen`	`[TK]D-Fender, I was working with foobar778 last night on this...`
`17:29.04`	`pigpen`	`late.`
`17:29.05`	`Nugget`	`The answer is 42.`
`17:29.23`	`pigpen`	`[TK]D-Fender, just me being a smart ass... :)`
`17:29.23`	`ronki`	`what is the role of the function "answer call"`
`17:30.02`	`foobar778`	`yes right pip`
`17:30.49`	`[TK]D-Fender`	`foobar778: Continue...`
`17:30.51`	`aydiosmio`	`ah yes`
`17:30.57`	`ronki`	`someone could help me it's very important`
`17:31.14`	`aydiosmio`	`I needed to use ForkCDR before Dial to get the billseconds for the Dial`
`17:31.28`	`foobar778`	`sorry fender call`
`17:31.29`	`foobar778`	`<PROTECTED>`
`17:31.40`	`foobar778`	`whn the voip call is made`
`17:31.47`	`foobar778`	`and then hung up`
`17:31.56`	`foobar778`	`then call persits`
`17:32.13`	`foobar778`	`it was routed did to disa then voip provider`
`17:32.29`	`foobar778`	`so when I hangup disa is still gouing`
`17:32.35`	`foobar778`	`going`
`17:32.42`	`foobar778`	`make any sense??`
`17:32.53`	`ronki`	`what is disa?`
`17:32.55`	`[TK]D-Fender`	`foobar778: Your problem is unclear.`
`17:33.10`	`rbd`	`looking at the docs, it looks like I could do an automated blind transfer to another SIP extn on the other asterisk server by doing something like Dial(SIP/XXXX@OTHERHOST\|TD(#)) ...meaning that the call will be transferred if the calling party presses a key (# in this case for blind transfer) and the # DTMF will automatically be sent after call setup...make sense?`
`17:33.18`	`foobar778`	`sec fender`
`17:33.43`	`[TK]D-Fender`	`rbd: "show application transfer"`
`17:34.06`	`ManxPower`	`ronki: it answers the call.`
`17:34.09`	`*** join/#asterisk svenna_ (n=svenna@p548D3999.dip0.t-ipconnect.de)`
`17:35.09`	`ManxPower`	`ronki: Most applications will answer the line if it is not already answered, so it is not needed all that often.`
`17:35.56`	`jesster_`	`hey all - running 79x1 phones trying to get the background image to load out-of-the-box. Any suggestions? Right now i have to goto Settings -> User Pref. -> Background Images`
`17:36.07`	`foobar778`	`fender give me a minute`
`17:36.54`	`foobar778`	`fender be back in 5 minutes would like to pick up`
`17:38.09`	`*** join/#asterisk AlfaScorpii (n=alfascor@64-12-16-190.fibertel.com.ar)`
`17:38.15`	`*** join/#asterisk ronki (i=ronki@ALyon-256-1-61-232.w90-9.abo.wanadoo.fr)`
`17:38.29`	`ronki`	`excuse me I have some connexion pb`
`17:38.55`	`ronki`	`so "answer call" means that if a call persist it will be routed at another post?`
`17:39.04`	`AlfaScorpii`	`need help with outband calls (pstn) cutting at 40 seconds`
`17:39.14`	`ManxPower`	`ronki: no. It means answer the call so you can play audio`
`17:40.10`	`ronki`	`hum I think I made a mistake I talk about the function "answer call" presents in asterisk and asterisk java`
`17:40.24`	`*** join/#asterisk topping (n=topping@dsl093-079-130.sfo1.dsl.speakeasy.net)`
`17:40.30`	`ronki`	`in the package fastagi more exactly`
`17:40.51`	`ManxPower`	`ronki: I cannot help you with asterisk java. I am referring to "show application answer" in the Asterisk CLI and Dialplan`
`17:41.16`	`ronki`	`but it's a function of the dialplan but I don't understand the goal`
`17:41.50`	`ManxPower`	`ronki: the function is not used very much`
`17:41.50`	`TheCops`	`Someone had problem with sip to sip audio quality? (One way, hear weird thing on the phone, ppl have problem to hear the caller etc etc..)`
`17:41.51`	`AlfaScorpii`	`ManxPower: dou you know why my outbaund calls pstn cutting in 40 seconds?`
`17:42.13`	`ManxPower`	`AlfaScorpii: if I knew I would tell you.`
`17:42.25`	`ronki`	`Manxpower : but my project require this function`
`17:42.43`	`AlfaScorpii`	`ManxPower: :)`
`17:42.47`	`*** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com)`
`17:42.55`	`AlfaScorpii`	`mercestes: HI!`
`17:43.05`	`ManxPower`	`ronki: most applications and functions will automatically answer the call before proceeding.`
`17:43.42`	`ronki`	`Manxpower: yes but in asterisk there's the function "answer call" too`
`17:44.09`	`AlfaScorpii`	`mercestes: my asterisk is finally working... but i have a little problem yet coz my outband calls pstn r cutting everytime in 40 seconds, can u help me?`
`17:45.04`	`ManxPower`	`ronki: See http://pastebin.ca/375849`
`17:46.40`	`ronki`	`I read it`
`17:48.03`	`ronki`	`so....I haven't found information maybe I don' t see it`
`17:50.24`	`ronki`	`Manxpower: french I suppose?`
`17:50.34`	`*** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net)`
`17:53.07`	`jesster_`	`Hey guys, when I try to ssh to my 7961 with the values in sshUserId and sshPassword, i am authed for SSH and am presented with a new login: for the phone, any idea how to get through?`
`17:54.01`	`CrashHD`	`why would a sip channel show in use (when setup as a queue member) when in fact it is not in use`
`17:54.48`	`[TK]D-Fender`	`CrashHD: Pastebin all the CLI backup of this evernt please.`
`17:55.03`	`AlfaScorpii`	`Does any body know why my outbaund calls psth only can live for 40 seconds�?`
`17:55.20`	`[TK]D-Fender`	`AlfaScorpii: You haven't given us any details. We know NOTHING.`
`17:55.28`	`CrashHD`	`[TK]D-Fender: http://www.pastebin.ca/375860`
`17:55.38`	`CrashHD`	`basically phones show (in use) but nobody is on them`
`17:56.13`	`[TK]D-Fender`	`CrashHD: And the "show channels" to back it up?`
`17:56.13`	`CrashHD`	`[TK]D-Fender: http://www.pastebin.ca/375864`
`17:56.21`	`AlfaScorpii`	`[TK]D-Fender: ok, look when i make an outbaund call using pstn lines the call is cutting in 40 seconds`
`17:56.47`	`CrashHD`	`in fact it is showing channels not in use when in fact they are in use`
`17:56.54`	`CrashHD`	`and vise versa`
`17:57.05`	`CrashHD`	`call-limit set to 100 on all sip channels`
`17:57.17`	`[TK]D-Fender`	`CrashHD: Yikes. reload the Queue`
`17:57.20`	`AlfaScorpii`	`[TK]D-Fender: i thik the problem may be the audio codecs or tone config on my gateway...`
`17:57.37`	`[TK]D-Fender`	`AlfaScorpii: If it was a codec issue you wouldn't even GET 40 seconds`
`17:57.50`	`[TK]D-Fender`	`AlfaScorpii: SHOW us sonething USEFUL.`
`17:57.53`	`CrashHD`	`[TK]D-Fender: just a reload doesn't clear in use propertes`
`17:58.04`	`anonymouz666`	`when using MOH is running asterisk is doing transcode?`
`17:58.13`	`[TK]D-Fender`	`CrashHD: "reload app_queue.so"`
`17:58.21`	`AlfaScorpii`	`im using g711A-Law`
`17:58.25`	`[TK]D-Fender`	`anonymouz666: Depends on the format of your music`
`17:58.35`	`CrashHD`	`[TK]D-Fender: did that, did not reset the in use stats`
`17:58.40`	`anonymouz666`	`mp3`
`17:58.53`	`[TK]D-Fender`	`anonymouz666: I'll let you think on that ;)`
`17:59.50`	`anonymouz666`	`[TK]D-Fender I am doing a sipp test on intel duo`
`18:00.11`	`anonymouz666`	`120 calls and nothing`
`18:00.19`	`anonymouz666`	`and can handle much more than that`
`18:00.26`	`anonymouz666`	`the load is ridiculous low`
`18:00.59`	`[TK]D-Fender`	`AlfaScorpii: Since you don't seem to have a clue : Pastebin the ENTIRE CLI output of a failed call with SIP debug enabled.`
`18:02.40`	`AlfaScorpii`	`[TK]D-Fender: ok`
`18:02.41`	`*** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud)`
`18:05.20`	`CrashHD`	`[TK]D-Fender: any ideas?`
`18:05.37`	`[TK]D-Fender`	`CrashHD: "restart now" ?`
`18:05.49`	`[TK]D-Fender`	`CrashHD: (will kill your calls of course)`
`18:05.52`	`CrashHD`	`ya`
`18:05.53`	`CrashHD`	`but`
`18:05.57`	`CrashHD`	`I meant`
`18:06.00`	`CrashHD`	`why this would be happening?`
`18:07.48`	`*** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl)`
`18:08.04`	`AlfaScorpii`	`Ups`
`18:08.17`	`AlfaScorpii`	`cli is changing all the time`
`18:08.24`	`AlfaScorpii`	`cant paste all`
`18:08.27`	`AlfaScorpii`	`:(`
`18:08.54`	`AlfaScorpii`	`how can i only see the debug of an specific call?`
`18:09.09`	`[TK]D-Fender`	`AlfaScorpii: Use a better clieint or learn how to use the one you have properly.`
`18:09.33`	`AlfaScorpii`	`[TK]D-Fender: client?`
`18:09.41`	`[TK]D-Fender`	`AlfaScorpii: Yes`
`18:09.46`	`AlfaScorpii`	`[TK]D-Fender: u meen the ssh client?`
`18:09.52`	`[TK]D-Fender`	`AlfaScorpii: yes`
`18:10.40`	`AlfaScorpii`	`[TK]D-Fender: i need to see only the info abaut one call but cli is showing all including the nat changes`
`18:12.03`	`[TK]D-Fender`	`AlfaScorpii: do it without debug, but on verbose 10 at least to start. You are not being pro-active on this at all...`
`18:12.48`	`JunK-Y`	`AlfaScorpii: you cant debug only one specific call.`
`18:12.55`	`JunK-Y`	`this is "in progress"`
`18:15.37`	`*** join/#asterisk sav_mcfly (n=mtaipe@pergamo.zonaz.net)`
`18:15.54`	`*** part/#asterisk siddu999 (n=siddu999@adsl-074-164-111-083.sip.bct.bellsouth.net)`
`18:16.32`	`[TK]D-Fender`	`Amazing how it takes over 15 minutes to squeeze a friggen pastebin outta someone....`
`18:18.47`	`thinwires`	`<3 pastebin's`
`18:20.09`	`type0`	`anyone running asterisk on solaris 10?`
`18:20.25`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`18:20.43`	`mercestes`	`if I do a reload chan_zapata.so will it kill current calls?`
`18:21.00`	`type0`	`im sure it will`
`18:21.05`	`JunK-Y`	`not a reload`
`18:21.12`	`jesster_`	`Hey guys, when I try to ssh to my 7961 with the values in sshUserId and sshPassword, i am authed for SSH and am presented with a new login: for the phone, any idea how to get through?`
`18:21.26`	`JunK-Y`	`but chan_zap.so requires restart much of the time, which will kill ur current calls.`
`18:22.38`	`tzafrir_laptop`	`JunK-Y, there a little 'zap restart' in recent asterisk versions (1.4?)`
`18:22.47`	`tzafrir_laptop`	`so you at least won't kill sip calls`
`18:23.00`	`tzafrir_laptop`	`It needs some further debugging`
`18:23.14`	`AlfaScorpii`	`resonm hungup normal clearing`
`18:23.18`	`AlfaScorpii`	`whats that?`
`18:24.24`	`tzafrir_laptop`	`and 'reload' of chan_zap.so sets most of the things you need`
`18:24.32`	`*** join/#asterisk RoyK (n=roy@ti211310a080-5608.bb.online.no)`
`18:24.34`	`*** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net)`
`18:24.37`	`generalhan`	`hey all !`
`18:24.57`	`mercestes`	`does reload chan_zap.so kill current calls?`
`18:26.19`	`aydiosmio`	`I sure hope so`
`18:28.16`	`ManxPower`	`mercestes: I don't believe so.`
`18:28.54`	`ManxPower`	`confirmed. reload chan_zap.so does not kill active calls.`
`18:29.39`	`ManxPower`	`sigh When will users understand that when we say we will be rebooting the phone system at time X, that if they are on a call at time X it will disconnect.`
`18:30.06`	`CrashHD`	`ManxPower: never`
`18:30.22`	`Nugget`	`Probably the day after people will figure out that it's not any easier to read the asterisk documentation after they've asked us to paste it into IRC than it is to just read the documentation directly.`
`18:32.24`	`jesster_`	`hey all - running 79x1 phones trying to get the background image to load out-of-the-box. Any suggestions? Right now i have to goto Settings -> User Pref. -> Background Images`
`18:33.09`	`[TK]D-Fender`	`AlfaScorpii: Its now been over half an hour for you to provide a pastebin of a defective call.`
`18:35.40`	`ManxPower`	`jesster_: I suggest you check the Wiki and the Cisco web site for docs on how to do this.`
`18:35.42`	`*** join/#asterisk friedrich\| (n=friedric@e177249067.adsl.alicedsl.de)`
`18:37.27`	`*** join/#asterisk AlfaScorpii (n=alfascor@64-12-16-190.fibertel.com.ar)`
`18:37.30`	`AlfaScorpii`	`ouch`
`18:37.59`	`AlfaScorpii`	`what meens when sip debug "hungup reason normal clearing"`
`18:39.24`	`pigpen`	`so is metermaid integrated into 1.4?`
`18:39.29`	`AlfaScorpii`	`http://pastebin.gulic.org/267`
`18:39.35`	`AlfaScorpii`	`is this ok? http://pastebin.gulic.org/267`
`18:41.02`	`*** join/#asterisk friedrich\| (n=friedric@e177249067.adsl.alicedsl.de)`
`18:41.47`	`creature_`	`anyone here using pap2 with asterisk?`
`18:42.35`	`*** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)`
`18:43.28`	`ManxPower`	`AlfaScorpii: what IS that pastebin.`
`18:44.18`	`ManxPower`	`sip debug ip.address.of.phone`
`18:44.48`	`ManxPower`	`creature_: PAP2 is locked to the provider. PAP2-NA is not locked to a provider.`
`18:45.21`	`*** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net)`
`18:45.26`	`*** join/#asterisk fiber0pti (n=John@207.114.199.107)`
`18:45.48`	`*** join/#asterisk friedrich\| (n=friedric@e177249067.adsl.alicedsl.de)`
`18:45.57`	`fiber0pti`	`Does anyone know where I could find statisitcs on Asterisk adoption?`
`18:46.13`	`*** join/#asterisk saftsack (n=oliver@pD9E07AD7.dip.t-dialin.net)`
`18:46.33`	`*** join/#asterisk topping (n=topping@dsl093-079-162.sfo1.dsl.speakeasy.net)`
`18:47.08`	`creature_`	`ManxPower: i'm using a PAP2T not locked`
`18:47.38`	`creature_`	`just got it and trying to make it register with my asterisk`
`18:47.41`	`*** join/#asterisk drako (n=ljd@unaffiliated/luisjose)`
`18:48.00`	`creature_`	`load of parameters, but i hope i will find out how to register it pretty soon :)`
`18:48.23`	`ManxPower`	`creature_: reset it to the factory defaults. You only need like 3 things set.`
`18:48.33`	`ManxPower`	`proxy, userid, secret/password`
`18:48.43`	`*** join/#asterisk jart (n=user@ool-43551046.dyn.optonline.net)`
`18:48.58`	`jart`	`how do i disable this packet2packet bridging in 1.4? it doesn't work`
`18:49.56`	`file`	`jart: what doesn't work with it?`
`18:50.08`	`creature_`	`ManxPower: yeah, i will have a look at it now`
`18:50.32`	`[TK]D-Fender`	`AlfaScorpii: No, that is not at all what I asked for.`
`18:50.32`	`jart`	`file: i have two 501s calling each other, and no audio at all`
`18:50.51`	`file`	`jart: what does an rtp debug show? NAT between them?`
`18:51.01`	`jart`	`no nat between them, same lan`
`18:51.10`	`[TK]D-Fender`	`fiber0pti: No real odds on that. there are no sales metrics to guage this by`
`18:51.24`	`ManxPower`	`jart: disallow=all and allow=ulaw`
`18:51.25`	`file`	`jart: okay pastebin an rtp debug, and what version of 1.4?`
`18:51.31`	`file`	`1.4.0?`
`18:51.55`	`jart`	`file: 1.4.0, the rtp debug stops once it bridges`
`18:51.56`	`jart`	`Packet2Packet bridging SIP/300-081dd028 and SIP/301-081e43e0`
`18:52.00`	`creature_`	`ManxPower: yeah, that was easy. registered now :)`
`18:52.20`	`file`	`jart: get 1.4 from SVN, it has some fixes`
`18:52.24`	`jart`	`ok`
`18:52.25`	`file`	`jart: in relation to that`
`18:53.09`	`pigpen`	`any idea when svn will commit to a full release?`
`18:53.10`	`ManxPower`	`jart: does canreinvite=no fix it?`
`18:53.14`	`*** join/#asterisk bitbandit (n=polx@65-103-228-59.slkc.qwest.net)`
`18:53.34`	`[TK]D-Fender`	`pigpen: "when its ready"`
`18:53.37`	`*** join/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.wa.comcast.net)`
`18:53.43`	`pigpen`	`[TK]D-Fender, :P`
`18:53.54`	`[TK]D-Fender`	`pigpen: You asked for it... by NAME no less ;)`
`18:54.10`	`pigpen`	`yeah....I deserve a beating....`
`18:54.20`	`file`	`patience is a virtue`
`18:54.31`	`pigpen`	`....and test in the meantime....`
`18:54.40`	`[TK]D-Fender`	`file: Forgive me Lord for .......`
`18:54.59`	`pigpen`	`.... doing RTA with postgres....`
`18:55.04`	`thinwires`	`hey D-Fend, you said you have IP501's right?`
`18:55.23`	`jart`	`devil bunnies! now it's being really weird`
`18:55.36`	`pigpen`	`I must say...1.4.0 has been running well in my limited environment...`
`18:56.03`	`foobar778`	`Fender Im back`
`18:56.06`	`foobar778`	`sorry`
`18:56.09`	`bkruse`	`[TK]D-Fender: run!`
`18:56.13`	`foobar778`	`lol`
`18:56.15`	`foobar778`	`ok`
`18:56.23`	`foobar778`	`The issue is this`
`18:56.33`	`jart`	`i'm trying to update our phone system in the vain hope that faxing will work`
`18:56.38`	`foobar778`	`I have a did number`
`18:56.48`	`foobar778`	`I call that number`
`18:56.57`	`foobar778`	`I have it set to goto DISA`
`18:57.07`	`foobar778`	`so now Im in my internal pbx`
`18:57.09`	`jart`	`because wasted hours of my bloated salary pulling my hair out with asterisk is a better solution than buying a phone line`
`18:57.09`	`LostFrog`	`I need teliax quality at broadvoice prices. :(`
`18:57.43`	`foobar778`	`then I dial a prefix and make an outbound call thru a provider`
`18:57.55`	`foobar778`	`If I hangup my analog phone`
`18:58.11`	`foobar778`	`it doesnt kill the DISA bridge`
`18:58.21`	`creature_`	`ManxPower: this was easier then i thought, took me 10 mins to make my first successfull call in/out :)`
`18:58.36`	`foobar778`	`therefore I will use more minutes on the ld call than wanted`
`18:58.39`	`*** join/#asterisk reza_ (i=reza@abort.boom.net)`
`18:58.58`	`reza_`	`ok, my voip provider nufone.net is down; too many other choises`
`18:59.04`	`reza_`	`what's a good reliabe one that supports iax2?`
`18:59.11`	`foobar778`	`The problem is that many bridges are being made thru this method`
`18:59.23`	`ManxPower`	`jart: I manage 4 asterisk servers and have been using Asterisk for 5 years. I run faxes thru a standard POTS line.`
`18:59.25`	`Qwell[]`	`reza_: all providers suck`
`18:59.28`	`Qwell[]`	`in one way or another`
`18:59.30`	`bitbandit`	`when i try to dial out all i get is "all circuts are busy..." and i use teliax, what are some things i can check to get this working ?`
`18:59.33`	`Qwell[]`	`JerJer: No offense`
`18:59.35`	`type0`	`teliax always worked for me`
`18:59.41`	`reza_`	`ok, i need a new one so i can get the phones up again. any suggestions?`
`18:59.43`	`ManxPower`	`bitbandit: the Asterisk CLI for one thing.`
`18:59.44`	`type0`	`I had a 310 did that was amazing`
`18:59.44`	`reza_`	`teliax? cool`
`18:59.46`	`reza_`	`us based?`
`18:59.49`	`type0`	`yeah`
`18:59.51`	`type0`	`great support`
`18:59.56`	`type0`	`lots of did blocks`
`18:59.58`	`reza_`	`what's 210?`
`19:00.00`	`reza_`	`er`
`19:00.00`	`reza_`	`310`
`19:00.02`	`Qwell[]`	`LA`
`19:00.02`	`type0`	`310 is socal`
`19:00.04`	`Qwell[]`	`county`
`19:00.09`	`type0`	`they dont issue it anymore`
`19:00.12`	`ManxPower`	`310 is west los angles`
`19:00.13`	`type0`	`i think 310 is full`
`19:00.19`	`Qwell[]`	`type0: They're all full :P`
`19:00.23`	`type0`	`haha`
`19:00.29`	`reza_`	`i use voxbone for did, great at night, sucky during congested hours`
`19:00.36`	`type0`	`teliax rocks anytime`
`19:00.44`	`type0`	`Qwell.. you ever use teliax?`
`19:00.46`	`Qwell[]`	`no`
`19:00.57`	`generalhan`	`Hey guys, im still working on this remote cisco7960, i was wondering if someone could take a look at my IPCop log and tell me whats going on here. the phone wont register, but it looks like its trying to get through on a strange port number.... http://generalhan.pastebin.ca/375901`
`19:00.59`	`bitbandit`	`ManxPower : i dont see any errors in there`
`19:01.05`	`Qwell[]`	`I use about 20 voip minutes a month...if that`
`19:01.09`	`bitbandit`	`says i am registered to teliax`
`19:01.14`	`reza_`	`how hard is it to port a did voip# to another carrier?`
`19:01.21`	`Qwell[]`	`reza_: meh...can be difficult`
`19:01.36`	`type0`	`umm.. pretty hard`
`19:01.36`	`Qwell[]`	`and usually costly`
`19:01.36`	`reza_`	`:(`
`19:01.36`	`type0`	`LNP isnt exactly, for voip`
`19:01.57`	`reza_`	`i wouldn't mind switching did providers, but dont want to lose the number`
`19:01.59`	`reza_`	`grumble`
`19:02.02`	`type0`	`forward :P`
`19:02.02`	`type0`	`haha`
`19:04.53`	`bitbandit`	`is there a pastebin for this channel ?`
`19:05.47`	`*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)`
`19:07.07`	`jesster_`	`hey all - running 79x1 phones trying to get the background image to load out-of-the-box. Any suggestions? Right now i have to goto Settings -> User Pref. -> Background Images`
`19:08.01`	`bitbandit`	`here is the output of the CLI when i try to dial out could somone take a peek and tell me what i am missin? http://generalhan.pastebin.ca/375955`
`19:08.19`	`generalhan`	`lol ... so take my pastebin .. thats cool ! lol`
`19:08.36`	`*** join/#asterisk friedrich\| (n=friedric@e177249067.adsl.alicedsl.de)`
`19:08.53`	`JoNate`	`i'm ready to kill someone...`
`19:09.05`	`JoNate`	`anybody have any experience with CyberGuard routers?`
`19:09.12`	`generalhan`	`JoNate: me too ,,, lets go on a rampage !`
`19:09.15`	`elriah`	`jesster_: There is no way to specify a 'default' background image that I've found.`
`19:09.20`	`bitbandit`	`generalhan : sorry`
`19:09.36`	`ManxPower`	`generalhan: We cannot help you with trixbox`
`19:09.38`	`elriah`	`Of course, the XML config isn't all that well documented...`
`19:09.38`	`jesster_`	`elriah: bah that's too bad. the older stuff let you`
`19:09.52`	`generalhan`	`ManxPower: this is Asterisk 1.2.10, not Trix`
`19:10.31`	`ManxPower`	`generalhan: then simplifiy the whole thing down to a single Dial() line and then as the next priority do a Noop(HANGUPCAUSE is ${HANGUPCAUSE})`
`19:10.45`	`generalhan`	`ManxPower: im still trying to get that cisco 7960 to register from a remote location .. but those entries on my local router are weird.`
`19:10.51`	`JoNate`	`damn router is blocking something, and I don't know why or where!`
`19:10.59`	`ManxPower`	`generalhan: I see the problem`
`19:11.01`	`generalhan`	`ManxPower: lol youre referring to bitbandit's post i think ... he used my pastebin`
`19:11.09`	`elriah`	`Is there a complete set of high quality asterisk stock audio for 1.2 in sln format yet?`
`19:11.11`	`ManxPower`	`generalhan: you are correct.`
`19:11.15`	`ManxPower`	`bitbandit: I see the problem`
`19:11.18`	`generalhan`	`lol`
`19:11.33`	`ManxPower`	`generalhan: (13:07:34) bitbandit: here is the output of the CLI when i try to dial out could somone take a peek and tell me what i am missin? http://generalhan.pastebin.ca/375955`
`19:11.40`	`bitbandit`	`sorry generalhan i thoguht i made new bin`
`19:11.51`	`bitbandit`	`i see the link that says new bin now`
`19:11.55`	`bitbandit`	`after the fack`
`19:11.59`	`bitbandit`	`oops fact`
`19:12.08`	`ManxPower`	`bitbandit: I see the problem.`
`19:12.21`	`bitbandit`	`what is it ?`
`19:12.22`	`generalhan`	`ManxPower: my post was at http://generalhan.pastebin.ca/375901 --- just trying to make sense of the router log and why my phone wont register`
`19:12.34`	`ManxPower`	`You do not have enough digits: -- Executing Dial("SIP/301-08dbf7c0", "IAX2/teliax/6692249\|300\|") in new stack`
`19:13.31`	`wunderkin`	`1 + area code, if us, also.. 300 sec timeout?`
`19:13.32`	`bitbandit`	`even when i do a 1-xxx-xxx-xxxx it doeds the same thing`
`19:14.02`	`*** join/#asterisk geejay101 (i=Tannenba@87.110.169.119)`
`19:14.11`	`JoNate`	`hey guys, if I've opened EVERY port on my router and forwarded it...why the HELL can't I make a call!`
`19:14.13`	`ManxPower`	`bitbandit: well paste bin a failed call when dialing 1-xxx-xxx-xxxx`
`19:14.23`	`bitbandit`	`k`
`19:14.50`	`*** join/#asterisk grEvenX (n=even@pc107-130.ktv.no)`
`19:14.57`	`ManxPower`	`bitbandit: also put a Noop(HANGUPCAUSE is ${HANGUPCAUSE}) as the priority after the Dial`
`19:15.21`	`*** join/#asterisk angom (n=angom@red-corp-201.143.88.126.telnor.net)`
`19:16.34`	`bitbandit`	`http://pastebin.ca/375963`
`19:17.08`	`JoNate`	`so no one has any experience with cyberguard?`
`19:17.56`	`bitbandit`	`where wouldi put the Noop(HANGUPCAUSE is ${HANGUPCAUSE}) this is my first asterisk project`
`19:17.56`	`geejay101`	`Good evening gentlemen.`
`19:18.28`	`jart`	`yea so anyway, canreinvite=no works around the problem`
`19:18.28`	`ManxPower`	`put as the priority after the Dial line`
`19:18.33`	`jart`	`and i'm using 1.4 svn`
`19:18.36`	`bitbandit`	`ok`
`19:18.58`	`geejay101`	`As we seem to have quite a number of users who still use pulse dialing phones I wonder whether there is any way in asterisk to detect on a voip channel whether the user uses pulse dialing trying to use an IVR application ?`
`19:19.36`	`*** join/#asterisk diclophis-work (n=jbardin@65.203.37.58)`
`19:19.40`	`diclophis-work`	`hello all`
`19:19.54`	`jart`	`hello young grasshopper`
`19:19.57`	`diclophis-work`	`is there a way to set a variable in a channel thats being dial from the Queue command`
`19:20.13`	`diclophis-work`	`for example, i have a bunch of "Local channels" logged into a queue`
`19:20.36`	`diclophis-work`	`and i would like a way to tell if a call is being sent to that local channel via the queue command, or some other path in the dialplan`
`19:20.52`	`jart`	`if it's going past a local, prefix the var with _`
`19:20.54`	`diclophis-work`	`i tried setting a channel variable right before the Queue comamnd, but the variable gets lost`
`19:20.58`	`geejay101`	`I was reasoning that perhaps asterisk could detect a pulse dial attempt as a flash signal ? Possible ?`
`19:21.00`	`ManxPower`	`diclophis-work: Set(__HAPPY_VAR=sad)`
`19:21.18`	`ManxPower`	`the __ prefix means make the vairable be on all created child channels.`
`19:21.36`	`ManxPower`	`that is two _ (underscore)`
`19:21.38`	`diclophis-work`	`awesome`
`19:21.45`	`diclophis-work`	`well the one underscore worked...`
`19:21.51`	`jart`	`if you're dialing local twice`
`19:21.57`	`ManxPower`	`docelmo: you need to read README.variables`
`19:22.39`	`geejay101`	`diclophis the number of underscores defines the scope of the variable`
`19:22.51`	`[TK]D-Fender`	`thinwires: Yes, I have an IP 501, what about it?`
`19:23.07`	`diclophis-work`	`geejay101: can there be more than __ ?`
`19:23.25`	`ManxPower`	`diclophis-work: you need to read README.variables`
`19:23.35`	`jart`	`lol`
`19:23.40`	`geejay101`	`disclophis - I dont think so - read wikipedia on asterisk variables`
`19:24.21`	`creature_`	`FXS Port Impedance: 270+750\|\|150nF`
`19:24.30`	`*** join/#asterisk notoriousrab (n=robert_m@76.195.14.206)`
`19:24.34`	`ManxPower`	`creature_: where are you? Peru?`
`19:25.09`	`notoriousrab`	`can anyone help me configure an auto attendant with version 1.4, cant get the samples or instructions in the book to do any logic`
`19:25.55`	`ManxPower`	`notoriousrab: there really isn't anything special about auto-attendant in 1.4.`
`19:27.54`	`notoriousrab`	`manxpower: ok, i am using the book asterisk the future of telephony and the demo in the sample file which is generated from make samples, i point a call at a different context and then answer it then try to get user input by using "background" and "waitexten" - when the user presses the option nothing happens`
`19:28.20`	`ManxPower`	`notoriousrab: http://pastebin.ca/375977`
`19:28.22`	`geejay101`	`BTW: I spent the better half of a day to get an attended transfer work on a cordless phone connected to a Cisco ATA 186. Flash didnt work, 2 in features didnt work, 3 in features.conf didnt work, finally I came up with ** in features.conf - works. Anyone any clue why this is so ?`
`19:28.43`	`ManxPower`	`geejay101: your ATA is configured wrong`
`19:28.56`	`ManxPower`	`FLASH transfers on SIP devices are done by the device, not asterisk`
`19:29.10`	`ManxPower`	`notoriousrab: what specifically is not working?`
`19:29.37`	`ManxPower`	`geejay101: chances are the ATA was eating the *codes for it's own stuff, you would have to diable that`
`19:29.41`	`geejay101`	`MaxPower I thought so too - but it worked perfectly with a fixed phone.`
`19:30.04`	`ManxPower`	`geejay101: You just said it did not work on the ATA`
`19:30.46`	`geejay101`	`The cordless phone didnt work with the ATA with flash, 2, 3 but the fixed phone did.`
`19:31.00`	`ManxPower`	`geejay101: that is weird`
`19:31.20`	`geejay101`	`Very wierd - I banged my head against the wall several times.`
`19:31.30`	`generalhan`	`anyone know of a cmdlet or app that will let me do a port test to see if i can get to my remote phone via a specific port. like a PING but where i can specify a port number ?`
`19:31.55`	`ManxPower`	`generalhan: nmap`
`19:32.05`	`notoriousrab`	`manxpower i will use the pastebin script and try to figure out what is not working, thanks for that, what isnot working just now is that asterisk does not take input from the phone eg pressing a 1 then doing whatever exten => 1,xxxx says in the same context`
`19:32.07`	`generalhan`	`nmap eh !? ok thats ill take a look at that`
`19:32.40`	`ManxPower`	`notoriousrab: what kind of phone?`
`19:32.50`	`notoriousrab`	`manxpower - polycom ip501`
`19:32.52`	`geejay101`	`Manxpower, Sorry it is not quite right what I said, flash transfers didnt work with the fixed phone either - I am under the impression that flash transfers are broken in my Asterisk.`
`19:32.59`	`ManxPower`	`notoriousrab: maybe your REAL problem is that DTMF is not being recognixed.`
`19:32.59`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`19:33.24`	`ManxPower`	`geejay101: on a SIP ata when you do a FLASH, the ATA handles everything for the transfer.`
`19:33.42`	`ManxPower`	`notoriousrab: what is your dtmfmode= set to in sip.conf?`
`19:34.00`	`geejay101`	`back to my pulse dialing problem - how can I detect uses with a pulse phone in an IVR on asterisk ?`
`19:34.19`	`ManxPower`	`generalhan: I don't believe you can over VoIP`
`19:34.31`	`generalhan`	`im not on VoIP`
`19:34.44`	`generalhan`	`PRI lines here`
`19:34.51`	`geejay101`	`I have a bunch of grannies who get pretty upset that their pulse phones are not working in the IVR`
`19:34.57`	`ManxPower`	`there needs to be fewer people here with ge as the first 2 letters of their nick`
`19:35.07`	`ManxPower`	`geejay101: Then Asterisk is not a solution for you.`
`19:35.20`	`generalhan`	`ManxPower: lol ... thats TWICE now !!`
`19:35.37`	`Nugget`	`When he's right, he's right.`
`19:36.57`	`*** join/#asterisk RoyKa (n=roy@ti211310a080-0229.bb.online.no)`
`19:37.11`	`geejay101`	`Manxpower what is my solution ? I have difficulties believing that I am the first one facing the problem of people with pulse phones in asterisk.`
`19:38.45`	`ManxPower`	`geejay101: have you looked in the mailing list archives?`
`19:39.22`	`ManxPower`	`geejay101: there IS no solution. You want Asterisk to support a technology that was "old" in the 1970s`
`19:39.27`	`geejay101`	`Manxpower, I googled like crazy - Ifound only hints regarding the zapata channels doing pulse dialing.`
`19:39.29`	`type0`	`ummm.. pulse phones would not work with Voip`
`19:39.39`	`Qwell[]`	`type0: says who?`
`19:39.52`	`type0`	`how would you translate a pulse`
`19:40.02`	`Qwell[]`	`how did they do it 50 years ago?`
`19:40.07`	`notoriousrab`	`manxpower - will check in a min, the dtmf setting, thanks`
`19:40.09`	`ManxPower`	`type0: it would come in as audio`
`19:40.34`	`LostFrog`	`Count the clicks.`
`19:40.39`	`LostFrog`	`THat would be CPU intensive.`
`19:40.40`	`type0`	`heh`
`19:40.42`	`geejay101`	`Pulse phones work perfectly with Voip - ideally the pulses should be caught on the switch an be transmitted out of band. The problem is that the switch doesnt do it.`
`19:41.08`	`Qwell[]`	`geejay101: then yell at your provider`
`19:41.26`	`type0`	`a pulse is actually a break in the telephone line.. the audible part is the click`
`19:41.46`	`LostFrog`	`sign each of your clients up for a year of Sports Illustrated so they get the free clock radio phone. :)`
`19:41.56`	`geejay101`	`Once the call is connected the switches dont seem to look at pulses anymore. However they catch DTMF and transmit it.`
`19:42.01`	`Nugget`	`Nah, the phone that looks like a shoe is cooler.`
`19:42.09`	`Qwell[]`	`it's all about the football phone`
`19:42.12`	`Qwell[]`	`come on now people`
`19:42.25`	`LostFrog`	`I want a pink princess phone!`
`19:42.27`	`LostFrog`	`NOT!!!`
`19:42.29`	`Qwell[]`	`geejay101: They should be transmitting it... there is nothing you can really do`
`19:42.51`	`type0`	`that is pretty funny though, voice over ip with pulse dialing`
`19:42.52`	`ManxPower`	`Asterisk does not support pulse ivr as far as I know`
`19:43.14`	`type0`	`that's like printing an email, and sending it through the postal service`
`19:43.15`	`type0`	`heh`
`19:43.15`	`Qwell[]`	`well, as type0 pointed out - it isn't audio`
`19:43.16`	`creature_`	`omg the pap2 is so full of settings (http://www.mynetfone.com.au/faq/LinksysPAP2/Linksys%20PAP2%20Configuration.htm)`
`19:43.21`	`geejay101`	`My humble thought was that Asterisk might be able to "see" a pulse inband as a flash signal.`
`19:43.45`	`Qwell[]`	`geejay101: a "pulse" will sound different depending on various things`
`19:44.03`	`type0`	`a pulse is just really a hook flash`
`19:44.15`	`creature_`	`trying to get the phone ringing (sound) but that doesnt seem to be easy :)`
`19:44.34`	`geejay101`	`can asterisk detect hook-flash inband ?`
`19:44.36`	`ManxPower`	`geejay101: you are a shining example of why you have to fully test and prototype an asterisk system before deployment`
`19:44.40`	`Qwell[]`	`geejay101: no`
`19:45.01`	`ManxPower`	`geejay101: only on FXS ports`
`19:45.05`	`Qwell[]`	`geejay101: a hook is a physical layer thing, and like I said, it can sound very different`
`19:45.28`	`ManxPower`	`creature_: I doubt it will work with that impedience setting`
`19:46.10`	`geejay101`	`Manxpower I can assure you that we did - unfortunately the pulse phones didnt come to our mind - to no ones else apparently either otherwise there would be a solution in Asterisk.`
`19:46.12`	`Qwell[]`	`geejay101: Imagine trying to get the password somebody typed on a keyboard - using just sound`
`19:46.17`	`Qwell[]`	`geejay101: it just ain't gonna work`
`19:46.41`	`creature_`	`ManxPower: thats not mine`
`19:46.43`	`ManxPower`	`geejay101: Why not just send pulse dialers to an operator?`
`19:46.51`	`creature_`	`ManxPower: im trying with 270+750\|\|150nF now`
`19:47.00`	`Qwell[]`	`ManxPower: about all you can do is let them timeout`
`19:47.01`	`ManxPower`	`creature_: if you are in the usa you want 600`
`19:47.07`	`creature_`	`ManxPower: im in sweden`
`19:47.18`	`ManxPower`	`creature_: Ah. then you do not want 600`
`19:47.28`	`ManxPower`	`Qwell: exactly`
`19:47.34`	`creature_`	`dunno and cant google the difference between sinusoid and trapezoid waveform`
`19:47.37`	`geejay101`	`Qwell I get your point.`
`19:48.00`	`creature_`	`ManxPower: ok, that didnt work.. dunno if thats the problem though`
`19:48.56`	`geejay101`	`is there any DSP plugin for Asterisk available ?`
`19:49.03`	`*** join/#asterisk RoyK (n=roy@ti211310a080-0229.bb.online.no)`
`19:49.17`	`Qwell[]`	`geejay101: give up - you're going down the wrong path. It isn't possible.`
`19:49.23`	`creature_`	`the cable from my phone to the fxs line is only connected to the two middle ports, think it might has something to do with that`
`19:49.31`	`*** join/#asterisk boojit (n=boojit@gw.carter.to)`
`19:50.21`	`ManxPower`	`creature_: no, the center two pins are all that is required`
`19:50.29`	`creature_`	`ManxPower: ok`
`19:51.01`	`creature_`	`then i must have the wrong settings`
`19:51.08`	`geejay101`	`Qwell I guess the only way to tell pulse dialing people that they have the wrong phone would be to ask them to dial a specific digit - if nothing happens one can announce to them to go to the supermarket and buy a new phone.`
`19:51.17`	`ManxPower`	`Qwell: You gotta admit that seeing him run around flapping his arms, thinking he can fly if he just wants too badly enough is sort of funny`
`19:51.45`	`ManxPower`	`geejay101: there are several sound files for asterisk available telling people that.`
`19:51.49`	`Qwell[]`	`geejay101: Do what EVERY other IVR in the world does. "[...] otherwise, please stay on the line."`
`19:51.49`	`LostFrog`	`supermarket?`
`19:52.09`	`LostFrog`	`My local Giant doesn't sell phones.`
`19:52.15`	`wunderkin`	`press 1 for english, press 2 for spanish`
`19:52.23`	`ManxPower`	`LostFrog: you are not in the USA are you?`
`19:52.28`	`LostFrog`	`Yes, I am.`
`19:52.38`	`LostFrog`	`I've never seen phones in supermarkets.`
`19:52.40`	`ManxPower`	`you can buy phones at the local pharmacy`
`19:52.51`	`*** part/#asterisk boojit (n=boojit@gw.carter.to)`
`19:52.52`	`LostFrog`	`In Walmart/Target/Kmart.. yes..`
`19:52.54`	`ManxPower`	`usually near the camera stuff`
`19:53.05`	`LostFrog`	`I will have to check next time I go shopping.`
`19:53.18`	`geejay101`	`Qwell thanks for the idea`
`19:53.46`	`LostFrog`	`That's true.. CVS does have phones.`
`19:53.52`	`Qwell[]`	`You should just Playback(rotary)`
`19:53.53`	`LostFrog`	`CVS has just about everything.`
`19:53.55`	`*** part/#asterisk angom (n=angom@red-corp-201.143.88.126.telnor.net)`
`19:54.09`	`Qwell[]`	`LostFrog: except revisioned repositories`
`19:54.26`	`LostFrog`	`Well.. they have suppositories, which is close enough.`
`19:55.43`	`geejay101`	`I guess we should send people using the IVR and making two bad attempts off to a special application handling that.`
`19:56.01`	`*** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net)`
`19:56.03`	`LostFrog`	`Switch to voice recognition. :)`
`19:56.10`	`LostFrog`	`God I hate companies that do that.`
`19:57.30`	`geejay101`	`voice recognition over VoiP is hell - I was recently in an IVR abroad unable to do anything - no DTMF - no voice recognition,`
`19:57.49`	`type0`	`how did you fit yourself into an IVR?`
`19:57.59`	`geejay101`	`badly`
`19:58.01`	`type0`	`pulse dialing and living in IVR's`
`19:58.08`	`type0`	`you need to get your life together.`
`19:58.39`	`type0`	`"Please stay on the line or go to a store and purchase a telephone made this century, thank you"`
`19:59.08`	`*** join/#asterisk bkw_ (n=brian@dsl093-079-130.sfo1.dsl.speakeasy.net)`
`20:00.13`	`[TK]D-Fender`	`"If you don't have a touch-tone phone please stay ont he like while we find someone to tell you what a back-water hick you are and arrange burial"`
`20:00.23`	`geejay101`	`type0, you would be amazed to hear how many pulse phones are still around in the ex-USSR. They have nice buttons but make pulses`
`20:00.53`	`type0`	`I will trade russian women for DTMF 'enabled' telephones`
`20:00.58`	`type0`	`call the embassy, we'll make the deal`
`20:01.03`	`[TK]D-Fender`	`geejay101: You sound like a perfect candidate for chan_carrierpidgeon.so`
`20:01.12`	`type0`	`haha`
`20:01.19`	`wunderkin`	`we may need that too`
`20:01.31`	`type0`	`chan_monkeybird.so`
`20:01.37`	`type0`	`FLY MY PRETTIES!`
`20:01.39`	`[TK]D-Fender`	`wunderkin: NO! You are our Sacrifice! We need you right as you are!`
`20:02.03`	`wunderkin`	`i told you that we're going to get grandstream budgetone phones!`
`20:02.16`	`reza_`	`type0 - cool, it works, thanks for the suggestion`
`20:02.20`	`type0`	`I'm thinking about buying those Linksys WIP330's`
`20:02.25`	`wunderkin`	`heh :D`
`20:02.39`	`reza_`	`they do have good support, picked up right away, but really didnt want to spend much time helping me`
`20:02.40`	`type0`	`reza_.. teliax?`
`20:02.40`	`[TK]D-Fender`	`typo : Wifi phones suck. All of them. HARD.`
`20:02.40`	`reza_`	`just said 'something wrong with your config'`
`20:02.43`	`reza_`	`yeah teliax`
`20:02.52`	`type0`	`tell them you want better support`
`20:02.57`	`reza_`	`?`
`20:02.59`	`type0`	`or you will dos their entire network`
`20:03.04`	`type0`	`they'll help you out`
`20:03.06`	`type0`	`just dial *67`
`20:03.07`	`reza_`	`right`
`20:03.07`	`type0`	`;)`
`20:03.22`	`reza_`	`problem was that i forgot to put a hole in the firewall`
`20:03.23`	`reza_`	`fixed`
`20:03.33`	`type0`	`you are putting holes in your firewall`
`20:03.38`	`type0`	`you're adding "exceptions"`
`20:03.39`	`wunderkin`	`maybe i forgot to put a hole in these phones`
`20:03.50`	`wunderkin`	`:D`
`20:04.02`	`type0`	`its not good practice to tell people you have a firewall full of 'holes'`
`20:04.30`	`*** join/#asterisk l2cache (n=ghansen@64.128.254.98)`
`20:04.43`	`type0`	`however, teliax will help you if you tell them they arent`
`20:04.53`	`type0`	`just straight up say.. what you are saying sucks.. please help me`
`20:05.00`	`*** join/#asterisk ToyMan (n=Stuart@user-12lcqvl.cable.mindspring.com)`
`20:05.41`	`l2cache`	`i am writing a program that assigns a variable in asterisk, passes it to System(linux command here) .. i need to find a way to pass the variable back to asterisk...`
`20:05.44`	`type0`	`i'm still not understanding why wireless sip phones suck`
`20:06.12`	`Nugget`	`They don't have to suck. They just do. There is no "why", just "is"`
`20:06.16`	`reza_`	`type0 - hack into teliax's server, and you can then connect through a firewall hole to my asterisk server over one udp port - if you want`
`20:06.22`	`Moobius`	`l2cache: I CURL() them in...`
`20:06.37`	`reza_`	`though i don't that'll help you accomplish anything much`
`20:06.42`	`type0`	`or I could just nmap the interweb and hack the planet.`
`20:06.51`	`l2cache`	`Moobius: what do you mean?`
`20:07.01`	`reza_`	`nmap 255.255.255.255/0`
`20:07.06`	`geejay101`	`Nokia E60, E61 E70 also suck on VoIP over Wifi ?`
`20:07.06`	`reza_`	`i dare you`
`20:07.17`	`reza_`	`ping -f 255.255.255.255 while you're at it`
`20:07.18`	`type0`	`im not even fucking with nokia equipment`
`20:07.23`	`type0`	`not a CHANCE`
`20:07.29`	`*** join/#asterisk notoriousrab (n=robert_m@76.195.14.206)`
`20:07.30`	`Moobius`	`asterisk issues an http request to a web server which runs a script, gets the data i want, and returns it as a dialplan variable`
`20:07.35`	`wunderkin`	`type0, get out of my tube!`
`20:07.49`	`type0`	`there's no reason to have one, when I cannot get cell coverage`
`20:07.56`	`type0`	`I JUST NEED SOMETHING RELIABLE AND WIRELESS`
`20:08.06`	`type0`	`and I'm scared to buy a shitload of ATA's and hook them up`
`20:08.18`	`type0`	`cordless phones are bullshit sometimes`
`20:08.23`	`mafkees`	`type0: get the tiptel/kirk stuff`
`20:08.29`	`notoriousrab`	`manxpower sorry got internet problems, tried modifying the pastebin code you sent me, same problem, what was your suggestion with dtmf in sip.conf`
`20:08.32`	`[TK]D-Fender`	`type0: Like the sign over the bar says : "Good. Fast. Cheap. Pick TWO."`
`20:08.48`	`geejay101`	`type0 no cell coverage ? are you located in Mongolia ?`
`20:08.56`	`type0`	`I'm working on a site in remote alaska`
`20:08.59`	`type0`	`near the canadian border`
`20:09.07`	`[TK]D-Fender`	`type0: Cordless phone may bullshit sometimes, but Wifi phones bullshit ALL THE TIME/`
`20:09.09`	`type0`	`with 50 million dollars of radar equipment`
`20:09.17`	`l2cache`	`it seem like asterisk will pass a channel variable to System(grep $variable /var/log/data.dat) but it will not pass back from system to asterisk?`
`20:09.17`	`type0`	`a t-1`
`20:09.25`	`type0`	`and no electric or phone service`
`20:09.32`	`JunK-Y`	`[TK]D-Fender: i disagree, somes arent.`
`20:09.55`	`geejay101`	`type0, volountary ? that explains your humor I guess :-)`
`20:10.13`	`[TK]D-Fender`	`JunK-Y: I have not found one thats at all friendly with hopping around and has a decent battery lif, and an acceptable feature set.`
`20:10.14`	`*** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com)`
`20:10.18`	`type0`	`actually, I have FXS cards on my coastcom mux with phones that do DTMF.. I win.`
`20:10.54`	`type0`	`the problem is getting the phone service 5000 ft down a mountain and 15 miles into a city`
`20:11.05`	`*** join/#asterisk hellojoe (n=hijoe@natint3.juniper.net)`
`20:11.09`	`type0`	`I basically have 15,000$ to blow on this`
`20:11.20`	`geejay101`	`type 0 - radio link ?`
`20:11.22`	`*** join/#asterisk s1gny\|wrk (n=s1gny@p54916B9D.dip.t-dialin.net)`
`20:11.22`	`type0`	`since a satellite phone is 5000$/year plus equipment costs (1000 a phone)`
`20:11.34`	`type0`	`I already have a 5.8ghz microwave network running over 500 miles`
`20:12.34`	`*** part/#asterisk l2cache (n=ghansen@64.128.254.98)`
`20:13.05`	`*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-4-30.lsanca.dsl-w.verizon.net)`
`20:13.13`	`type0`	`I found these wireless range extenders that'll do up to 1.2mbps`
`20:13.23`	`geejay101`	`type0 - I can only guess that the radar data must somehow get off your mountain to some analysts - so presumably there is a link that can also carry speech.`
`20:13.24`	`type0`	`at 20 miles LOS`
`20:13.40`	`type0`	`geejay101.. of course.. I have an RLB on my coastcom`
`20:13.41`	`*** part/#asterisk s1gny\|wrk (n=s1gny@p54916B9D.dip.t-dialin.net)`
`20:13.54`	`type0`	`getting the speech from the mux to the city, is the problem`
`20:14.05`	`type0`	`I can pickup the phone and use the FXS card on the mountain perfectly`
`20:14.23`	`geejay101`	`neither a RLB nor a coastcom means anything to me.`
`20:14.33`	`type0`	`radio lan bridge on a coastcom multiplexor`
`20:14.44`	`type0`	`my adtran radios are connected to a mux`
`20:15.09`	`type0`	`so there is a mux on each end`
`20:15.29`	`*** join/#asterisk i3inary (i=i3inary@ip68-8-91-87.sd.sd.cox.net)`
`20:15.29`	`type0`	`one end is the mountain, the other is at an airforce base -- where the FXO cards live`
`20:15.34`	`Mahmoud`	`does * support video conferencing via sip?`
`20:15.36`	`geejay101`	`and that link also carries internet ?`
`20:15.41`	`type0`	`yessir`
`20:15.48`	`type0`	`I just assign timeslots to the card in the mux`
`20:16.08`	`type0`	`I have 512k of radar data, and the rest is fxs cards and "internet"`
`20:16.33`	`geejay101`	`surprisingly little radar data`
`20:16.47`	`type0`	`you'd be amazed of the shit they can transfer over 512k`
`20:16.50`	`type0`	`no joke`
`20:17.48`	`type0`	`http://i16.photobucket.com/albums/b30/type0/a4e567eb.jpg`
`20:17.50`	`type0`	`that's the radome`
`20:17.59`	`type0`	`on the left there, that's the microwave dish`
`20:18.13`	`geejay101`	`So when the Russian ICBMs are flying in then someone just has to stand in the sight of the radiolink and NY is history ?`
`20:18.27`	`type0`	`this isnt for national security`
`20:18.31`	`type0`	`this is for flight tracking`
`20:18.45`	`type0`	`http://i16.photobucket.com/albums/b30/type0/3596e393.jpg`
`20:18.47`	`type0`	`me at -70`
`20:19.28`	`type0`	`this site is for military training`
`20:19.35`	`type0`	`all the jets carry datapods`
`20:19.44`	`type0`	`and we simulate missile threats direct to the aircraft`
`20:19.56`	`type0`	`so they can travel at mach 2+`
`20:20.01`	`type0`	`and the data is being sent down to our sites`
`20:20.06`	`type0`	`and onto the military base`
`20:20.26`	`type0`	`so when a missile threat is received.. we can detect the aircrafts movement through the training ranges`
`20:20.28`	`type0`	`in real time`
`20:20.31`	`LostFrog`	`I hate my fellow IT staff.. I told them make install linux.. They did it as an unprivileged user.`
`20:20.32`	`LostFrog`	`:(`
`20:20.41`	`*** join/#asterisk bitbandit (n=polx@65-103-228-59.slkc.qwest.net)`
`20:20.41`	`LostFrog`	`Ooops.. make install asterisk`
`20:21.20`	`geejay101`	`type0 So how long do your stints on the mountain last ?`
`20:21.29`	`type0`	`depends on the weather.. and mode of how we get there`
`20:21.38`	`type0`	`if I snow machine the 100miles in.. I can stay there for up to a week`
`20:21.46`	`type0`	`helicopter I have about 6 hours on the ground.. weather permitting`
`20:22.14`	`type0`	`last time we were up there for 3 days, the helicopter showed up with another company who maintains the radar.. their helo wouldnt start after sitting there for 3 hours`
`20:22.30`	`JunK-Y`	`when leaving a vm, if ive: origdate=Wed Feb 28 03:17:48 PM EST 2007`
`20:22.31`	`JunK-Y`	`origtime=1172693868`
`20:22.31`	`geejay101`	`Thats not too bad - I thought they locked you up there for months .`
`20:22.37`	`type0`	`oh no..`
`20:22.38`	`type0`	`fuck that`
`20:22.46`	`type0`	`its an unmanned site`
`20:22.47`	`JunK-Y`	`but a date from shell is okay, what's wrong?`
`20:22.57`	`JunK-Y`	`i recorded that msg 5 min. ago`
`20:23.21`	`Qwell[]`	`JunK-Y: 5 minutes ago was 3:17 EST`
`20:23.35`	`JunK-Y`	`no`
`20:24.04`	`JunK-Y`	`i mean, yeah, but: origtime=1172693868`
`20:24.14`	`JunK-Y`	`vm is saying 8 oclock`
`20:24.24`	`thinwires`	`is that what it says? Feb28 (today) 3:17:48 EST (3, 17)`
`20:24.24`	`JunK-Y`	`origtime=1172693868`
`20:24.26`	`geejay101`	`type0, is it possible to barbecue something in the radar beam ?`
`20:24.31`	`type0`	`nah`
`20:24.34`	`type0`	`that's a myth`
`20:24.52`	`type0`	`standing in front of it isnt exactly, healthy`
`20:25.02`	`type0`	`but standing in front of the radar array will definately warm your ass up`
`20:25.10`	`Qwell[]`	`JunK-Y: that's seconds since epoch...it's right`
`20:25.26`	`Qwell[]`	`...give or take`
`20:25.54`	`JunK-Y`	`so why its saying received at 8:17 ?`
`20:26.15`	`geejay101`	`warm your ass up - might be beneficial in that height.`
`20:26.30`	`type0`	`heh, warm clothes are better than radiation`
`20:26.34`	`JunK-Y`	`(Pm) and my tz is eastern`
`20:26.45`	`type0`	`you can be in the radome when its powered up.. but you dont want to be in front of it`
`20:26.45`	`Qwell[]`	`JunK-Y: it's saying it in utc`
`20:27.10`	`type0`	`this is 3rd generation radar shit`
`20:27.14`	`type0`	`not even the FAA uses it`
`20:27.19`	`Mahmoud`	`this totally sucks.. my ISP blocks voip providers to force its people to use their damn analog costy telephony`
`20:27.22`	`SplasPood`	`Qwell[]: so turns out the problem with the TDM800P seems to be related to ACPI and irq sharing`
`20:27.23`	`[TK]D-Fender`	`JunK-Y: "load chan_fluxcapacitor.so"`
`20:27.43`	`SplasPood`	`Qwell[]: works fine in an older box (HP DL-140)`
`20:27.44`	`geejay101`	`Mahmoud where are you located ?`
`20:27.44`	`Mahmoud`	`i'm accessing voip websites to download SIP soft phones and feel as if i were browsing some pr0n sites`
`20:28.03`	`JunK-Y`	`Qwell[]: since? can i return to eastern time (like specified in my .conf) ?`
`20:28.07`	`Mahmoud`	`geejay101, UAE, where people do nasty things (palm island, tallest buildings.etc)`
`20:28.23`	`type0`	`is porn filtered?`
`20:28.47`	`Mahmoud`	`sure!`
`20:28.52`	`type0`	`Mahmoud can you VPN to anything/`
`20:28.53`	`Mahmoud`	`pr0n is acceptable, but voip?!`
`20:28.58`	`SplasPood`	`Does anyone know if its possible to get VoiceMailMain() to play msgs back from newest to oldest rather than the default oldest -> newest ?`
`20:29.11`	`creature_`	`Gah, i'm having problems with my PAP2T. I can answear incoming calls but i cannot hear the ring tone when someone calls my phone (though i can answear if i know someone's calling).`
`20:29.13`	`Mahmoud`	`type0, i use t0r to by pass their proxy.. but it is slower any way`
`20:29.15`	`Qwell[]`	`SplasPood: nope, not without hacking up the code`
`20:29.24`	`geejay101`	`Mahmoud - do they also block specific websites of VoIP providers ? Or only the VoIP traffic ?`
`20:29.36`	`SplasPood`	`Qwell[]: lame/exepected :) I wonder if I care enough to hack up the code for this client...`
`20:29.42`	`Mahmoud`	`geejay101, both, voip traffic + voip providers`
`20:29.59`	`type0`	`just vpn to your network, and browse through that`
`20:30.04`	`Mahmoud`	`geejay101, they did also block google's translation feature, because people can use it to view blocked sites heh`
`20:30.20`	`Mahmoud`	`geejay101, certain keywords are banned from being searched in google, yahoo or other popular search engines`
`20:30.22`	`*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de)`
`20:30.43`	`Mahmoud`	`and yet, their proxy is transparent.. means no way`
`20:31.10`	`geejay101`	`type0 . the VPN doesnt help if they block the IP of the remote site - hence my question. Unless he sets up some personal IP somewhere abroad.`
`20:31.27`	`type0`	`you'd be routing the traffic through the VPN`
`20:31.34`	`type0`	`thats a tunnel brotha.`
`20:31.43`	`Mahmoud`	`type0 is right =P`
`20:31.44`	`type0`	`they have no idea of whats going over it`
`20:31.54`	`type0`	`they just know you're connected to another network`
`20:32.04`	`type0`	`its not like you're sending the request to their proxy`
`20:32.08`	`type0`	`you're sending it to yours`
`20:32.42`	`type0`	`now the speed of the vpn is dependant on quite a few things`
`20:32.50`	`type0`	`however --- thats your solution`
`20:32.55`	`creature_`	`ManxPower: i was able to solve it now.. changed the ring frequency from 25 to 50.. oh i should have known ;)`
`20:33.02`	`geejay101`	`type0, if they know that the remote VPN provides VoIP they will block it I suppose.`
`20:33.14`	`type0`	`they cant block VPN`
`20:33.21`	`type0`	`well they could`
`20:33.24`	`type0`	`but they wouldnt`
`20:33.28`	`*** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com)`
`20:33.29`	`type0`	`that's 'i need this for business' shit`
`20:33.31`	`geejay101`	`They simply block the remote IP. Period.`
`20:33.38`	`Mahmoud`	`bingo`
`20:33.43`	`type0`	`they might filter the port`
`20:33.48`	`type0`	`but i'd cause a big stir about that`
`20:33.49`	`Mahmoud`	`shhhhh they may hear it!`
`20:34.05`	`type0`	`they have no idea what's going over a PPTP/IPsec tunnel`
`20:34.06`	`Mahmoud`	`they blocked udp 5060 i guess`
`20:34.09`	`*** part/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`20:34.23`	`Mahmoud`	`i'm using some other ports`
`20:34.34`	`geejay101`	`They can block ports, IPs and traffic type.`
`20:34.41`	`type0`	`seriously, just VPN and connect to your box through that`
`20:34.52`	`Qwell[]`	`meh, just move`
`20:35.05`	`Mahmoud`	`every channel has two guys loving to argue with each others.. you both seem the twh for #asterisk`
`20:35.08`	`LostFrog`	`It sucks, but I use IAX over tinc.`
`20:35.12`	`geejay101`	`Type0 suppose you wanted to offer commercial VoIP to UAE - how would you go about it ?`
`20:35.30`	`type0`	`I would colocate a box somewhere out of the UAE`
`20:35.31`	`type0`	`but near`
`20:35.44`	`LostFrog`	`Like Iraq?`
`20:35.51`	`aydiosmio`	`LOlraq`
`20:35.56`	`Mahmoud`	`heh`
`20:36.02`	`geejay101`	`If you would avertise that then they could easily find the IP of your VPN box.`
`20:36.18`	`type0`	`thats not true`
`20:36.19`	`Mahmoud`	`hmmm iraq is a good idea, every thing is so cheap`
`20:36.28`	`*** join/#asterisk backblue (n=moo@87-196-5-169.net.novis.pt)`
`20:36.44`	`type0`	`You could build an application similar to Hamachi`
`20:36.46`	`aydiosmio`	`we do pc2phone in the UAE`
`20:36.48`	`geejay101`	`how could they NOT detect it ????`
`20:37.16`	`aydiosmio`	`we use a proprietary SIP client that uses a non-standard SIP port and tunneling protocol`
`20:37.17`	`type0`	`ok.. as a provider.. they can filter whatever they want`
`20:37.35`	`type0`	`with stateful packet inspection`
`20:37.43`	`type0`	`they can do anything`
`20:37.46`	`ManxPower`	`aydiosmio:`
`20:37.46`	`ManxPower`	`then your client is not SIP`
`20:37.53`	`type0`	`bullshit`
`20:37.55`	`type0`	`SIP is a protocl`
`20:38.00`	`aydiosmio`	`haha it is SIP`
`20:38.01`	`type0`	`ports and tunneling have nothing to do with it`
`20:38.05`	`aydiosmio`	`it's just tunneled`
`20:38.18`	`type0`	`thats like tunneling a telnet session on port 7500`
`20:38.19`	`Nugget`	`telnet is eeeeeeevil!`
`20:38.22`	`ManxPower`	`Can X-Lite use the service?`
`20:38.24`	`type0`	`its still telnet`
`20:38.37`	`geejay101`	`aydiosmo, that means they are not very sophisticated.`
`20:38.41`	`type0`	`X-Lite is a softphone, im sure it could if you could add the tunneling`
`20:38.51`	`aydiosmio`	`ManxPower: yes, we offer both types of connections, tunneling on port xxxx and direct sip on 5900`
`20:38.53`	`ManxPower`	`As far as I am concerned, if a standard SIP device cannot use the service, then it is not a SIP service.`
`20:39.06`	`Mahmoud`	`any good free sip soft phone for windows that support video`
`20:39.09`	`type0`	`as far as the RFC is concerned..`
`20:39.12`	`LostFrog`	`I wonder how often there is a call on all five of my landlines at the same time..`
`20:39.27`	`aydiosmio`	`ManxPower: in many countries it's impossible to use the SIP ports/protocols directly`
`20:39.32`	`ManxPower`	`MY website supports HTTP, but no existing browser works with it.`
`20:39.42`	`aydiosmio`	`they're filtered at the application later in some cases`
`20:39.51`	`type0`	`who cares about the application layer`
`20:40.00`	`aydiosmio`	`what are oyu yammering about? we offer both`
`20:40.01`	`*** part/#asterisk RoyK (n=roy@ti211310a080-0229.bb.online.no)`
`20:40.02`	`ManxPower`	`perhaps a good term would be "modified SIP protocol"`
`20:40.12`	`type0`	`you arent modifying the protocol though`
`20:40.18`	`*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)`
`20:40.20`	`type0`	`you're modifiying the CONNECTION to the protocol`
`20:40.36`	`type0`	`the protocol is still the same`
`20:40.41`	`type0`	`the way you're connecting is different`
`20:40.50`	`*** join/#asterisk genz (n=chatzill@im.jobdig.com)`
`20:40.52`	`type0`	`the game is still the same, the rules have just changed`
`20:40.58`	`aydiosmio`	`enough anologies`
`20:41.05`	`aydiosmio`	`we offer both connection types`
`20:41.05`	`type0`	`a modified protocol is routing video of DNS`
`20:41.16`	`aydiosmio`	`we aren't offering a draconian voip service`
`20:41.56`	`*** join/#asterisk topping (n=topping@dsl093-079-162.sfo1.dsl.speakeasy.net)`
`20:42.45`	`ManxPower`	`I'm just sick and tired of vendors claiming to support a specific standard, but you eventually find out that nothing actually supports whatever changes they made to the "standard".`
`20:42.47`	`type0`	`http://www.doxpara.com/dns_tc/Black_Ops_DNS_TC_files/frame.htm`
`20:42.57`	`type0`	`that's what I would use if I were to make a voip provider in the UAE`
`20:43.04`	`type0`	`tunnel sip traffic over dns`
`20:43.06`	`aydiosmio`	`The client encapsulates the SIP data and sends it to our proxy tunnel and then forwards it to the softswitch, we specifically requested this feature for our version of the client`
`20:43.10`	`type0`	`because they can NEVER filter DNS`
`20:43.14`	`genz`	`Anybody running Cacti and have a T1 and want to try my script that monitors T1 usage?`
`20:43.17`	`aydiosmio`	`and we offer both clients`
`20:43.42`	`ManxPower`	`Nortel has done this with pretty much every VoIP protocol they claim to support.`
`20:43.46`	`LostFrog`	`They can filter based of heuristic queing..`
`20:43.59`	`LostFrog`	`Let a certain number of DNS messages per second.`
`20:44.00`	`aydiosmio`	`LostFrog: they can, but it's expensive`
`20:45.44`	`ManxPower`	`You'd think it would be easier just to get into a legal business, rather then an illegal voip service in a country where voip is not legal.`
`20:46.09`	`wunderkin`	`genz, yeah i can take a look`
`20:46.25`	`LostFrog`	`Luckily VoIP was legalized in India.`
`20:46.42`	`aydiosmio`	`they had too`
`20:46.47`	`aydiosmio`	`all those call centers...`
`20:47.00`	`aydiosmio`	`ManxPower: you'd think so.`
`20:47.12`	`type0`	`i think an illegal sip provider would do well in the UAE`
`20:47.34`	`type0`	`and tunneling it through DNS would let the country know you are a complete arab ninja`
`20:48.33`	`LostFrog`	`That's pretty good.. DNS is QoSed as interactive.. :)`
`20:48.33`	`generalhan`	`ManxPower: so im using nmap like you suggested and im confused about what im seeing. http://generalhan.pastebin.ca/376080 does that mean that the port is being forwarded at the remote router? or just that the port exists ? lol`
`20:48.54`	`mercestes`	`type0: "tunneling it through DNS" and "ninja" together in one sentence proves beyond all doubt that you are a virgin. :D`
`20:49.06`	`creature_`	`I have a problem with the PAP2T hanging up the line. The PAP2T is configured on line 1 where i have a analogue phone connected to that FXS port. On my computer i'm using a SIP Client. If i make a call from my SIP Client to my analogue phone and then hang up the analogue phone the PAP2T doesn't send a Hangup. I found a thread where another guy have the same problem and solved it using the CPC Setting, though that didn't work for me (https:`
`20:49.43`	`type0`	`I get more ass in a weekend that you've seen in a month`
`20:49.45`	`type0`	`i promise you that.`
`20:49.48`	`*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de)`
`20:49.51`	`ManxPower`	`generalhan: I don't know. You used -p 5060 and yet the response came back from 5062`
`20:50.00`	`aydiosmio`	`(https://www.clientbufferlongerthanserverbuffer.com)`
`20:50.06`	`LostFrog`	`type0: It's unfortunate that it's all man ass.`
`20:50.10`	`type0`	`just beacause I use the word ninja, does not count me out of the vaginal pool`
`20:50.10`	`aydiosmio`	`now now`
`20:50.15`	`generalhan`	`ManxPower: sorry, i must have mixed up the pastes ...`
`20:50.15`	`aydiosmio`	`let's put our dicks away`
`20:50.22`	`type0`	`man ass is sexy`
`20:50.28`	`creature_`	`The SIP Client stays connected.. when haning up the SIP Client i get this from asterisk cli: WARNING[1870]: chan_sip.c:12171 handle_response: Remote host can't match request BYE to call '460e99320874cf205646bade68674c61@10.10.0.2'. Giving up. (where 10.10.0.2) is the asterisk ip`
`20:50.46`	`aydiosmio`	`lolnat`
`20:50.51`	`creature_`	`aydiosmio: works great`
`20:50.55`	`geejay101`	`type0 - tunneling through "DNS" ????`
`20:50.58`	`aydiosmio`	`you kids and you silly imaginary IP addresses`
`20:51.10`	`creature_`	`aydiosmio: imaginary ip adresses, lol`
`20:51.16`	`generalhan`	`ManxPower: comes back in the same format for 5060 "5060/udp open\|filtered sip" which is why im confused, beause im not forwarding port 5062 and the test came back with the same results`
`20:51.57`	`*** join/#asterisk bkw_ (n=brian@dsl093-079-162.sfo1.dsl.speakeasy.net)`
`20:52.58`	`ManxPower`	`generalhan: I suspect I think, that when sending to that port the router send back ICMP prohibited, which would mean the port is open, but the source ip is not permitted to get thru the router filter`
`20:53.34`	`generalhan`	`hmm`
`20:53.47`	`generalhan`	`you know what the output would look like if it went through successfully ?`
`20:57.15`	`*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)`
`20:58.15`	`generalhan`	`maybe i can set this up via IAX instead of SIP ??`
`20:58.21`	`*** join/#asterisk dasenjo (n=dasenjo@190.24.177.189)`
`21:01.10`	`*** join/#asterisk darviria (n=dar@ACC9A92E.ipt.aol.com)`
`21:01.19`	`*** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-06e9ee9ba596d77d)`
`21:01.50`	`LostFrog`	`IAX rocks.`
`21:02.21`	`mercestes`	`I think I hurt Type0's feelings.`
`21:02.50`	`mercestes`	`I'm sorry, :( I was kidding.`
`21:04.40`	`type0`	`im on the phone with the bank`
`21:04.42`	`type0`	`getting a loan`
`21:04.42`	`type0`	`wee`
`21:07.31`	`*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)`
`21:07.51`	`*** part/#asterisk backblue (n=moo@87-196-5-169.net.novis.pt)`
`21:09.20`	`*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de)`
`21:10.46`	`*** join/#asterisk nasls_lsa (n=chatzill@athedsl-218920.otenet.gr)`
`21:13.07`	`*** join/#asterisk flying_Luck (n=melifaro@ppp85-141-155-106.pppoe.mtu-net.ru)`
`21:13.41`	`aydiosmio`	`I still haven't figured out why WwTtHh won't work on my Dial`
`21:16.11`	`geejay101`	`type0 - what do you need a loan for ? Just stay on they montain.`
`21:17.48`	`*** join/#asterisk ocgltd (n=support@CPE004063e0ee74-CM00159a010632.cpe.net.cable.rogers.com)`
`21:18.14`	`creature_`	`anyone knows what ERROR[1870]: chan_sip.c:14652 sipsock_read: We could NOT get the channel lock for SIP/16188-081ae4a8! means?`
`21:18.51`	`ocgltd`	`My asterisk 1.40 install, with 2000+ active channels, is showing the following every 1/2 second: chan_sip.c:2739 auto_congest: Auto-congesting SIP/my.domain.net-0e118a90. What does this mean?`
`21:19.16`	`*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)`
`21:19.50`	`*** join/#asterisk boch (n=fran@190.48.234.129)`
`21:20.17`	`aydiosmio`	`hey Hh worked from my landline!`
`21:20.23`	`*** join/#asterisk ocgltd (n=support@CPE004063e0ee74-CM00159a010632.cpe.net.cable.rogers.com)`
`21:20.25`	`aydiosmio`	`stupid cellphone.`
`21:21.01`	`boch`	`hello, anyone using Bridge() dialplan app ?`
`21:27.28`	`*** join/#asterisk funnymanva (n=carlton@dsl093-079-162.sfo1.dsl.speakeasy.net)`
`21:27.41`	`ocgltd`	`Sorry to repeat, I got disconnected. My asterisk 1.40 install, with 2000+ active channels, is showing the following every 1/2 second: chan_sip.c:2739 auto_congest: Auto-congesting SIP/my.domain.net-0e118a90. What does this mean?`
`21:28.15`	`*** join/#asterisk bkw_ (n=brian@dsl093-079-130.sfo1.dsl.speakeasy.net)`
`21:28.20`	`aydiosmio`	`oh hH is only working for the calling party, not the called. Damnit.`
`21:29.01`	`*** join/#asterisk mut (n=ana@65.111.222.120)`
`21:29.30`	`mut`	`having a problem with ringing not ringing correctly.. what happens is...`
`21:29.50`	`mut`	`user has polycom phone, call comes in, sent to polycom, user has a call forward setup to call her cell in the polycom`
`21:30.08`	`geejay101`	`ocgltd - 2000 active channels ? I wasnt aware that Asterisk can handle that. what are these channels doing ?`
`21:30.09`	`mut`	`so the call comes back out of the phone, and goes to pstn, the caller can't hear a ring, but the cellphone does ring`
`21:30.40`	`*** join/#asterisk dasenjo_ (n=dasenjo@190.24.24.69)`
`21:31.34`	`ocgltd`	`Coming in H323, our SIP. Asterisk is serving as a gateway.`
`21:31.42`	`ocgltd`	`But, at around 2500 calls Asterisk hangs`
`21:32.18`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`21:32.20`	`LostFrog`	`It sounds like you need a cluster.`
`21:32.30`	`ocgltd`	`I would like to limit the number of active channels to 1000. Is there a setting in Asterisk that will refuse calls once the max (1000) is hit?`
`21:33.22`	`JoNate`	`Anyone have experience with * and CyberGuard routers?`
`21:33.24`	`type0`	`asterisk on solaris is supposed to be able to handle 325 calls per second`
`21:33.39`	`type0`	`with the mtmalloc library you can do 1400calls per second`
`21:33.44`	`ManxPower`	`ocgltd: look in the Wiki for GROUP_COUNT`
`21:34.35`	`JunK-Y`	`type0: right, i doubt so.`
`21:34.36`	`*** join/#asterisk rushowr (n=rushowr@cpe-65-24-149-191.columbus.res.rr.com)`
`21:34.55`	`*** part/#asterisk rushowr (n=rushowr@cpe-65-24-149-191.columbus.res.rr.com)`
`21:34.59`	`type0`	`http://www.thrallingpenguin.com/articles/asterisk-solaris.htm`
`21:36.04`	`type0`	`solaris 10 with a sunfire is going 1400 calls per second`
`21:36.07`	`*** join/#asterisk bkw_ (n=brian@dsl093-079-162.sfo1.dsl.speakeasy.net)`
`21:36.09`	`type0`	`at 28% system cpu`
`21:36.17`	`type0`	`with 14,000-15,000 context switches`
`21:36.28`	`*** join/#asterisk topping (n=topping@dsl093-079-130.sfo1.dsl.speakeasy.net)`
`21:36.29`	`diclophis-work`	`how can a call be connected, with recieving audio not working during the application Voicemail ?`
`21:36.34`	`type0`	`with mtmalloc`
`21:36.47`	`ocgltd`	`Is there a way to send a message back the caller that all channels are congested? (H323 channel)`
`21:36.53`	`geejay101`	`ocglt, so Asterisk does protocol conversion H323-SIP, and handles also RTP traffic ?`
`21:37.09`	`aydiosmio`	`Dial(SIP/17275551234@70.52.18.10,,mghHwWM(record\|${recordid}))`
`21:37.34`	`aydiosmio`	`*/1 work for the calling party, but not the called, anyone have an idea why?`
`21:37.42`	`FuriousGeorge`	`what does it mean when you are making a sip call and the other end complains that you are too quiet?`
`21:37.52`	`type0`	`FuriousGeorge.. that you need to talk louder.`
`21:38.11`	`FuriousGeorge`	`type0: thanks, will you be here all week? if so, should i try the veal?`
`21:38.32`	`type0`	`try out turn up the gain on the channel special`
`21:38.54`	`boch`	`Do you know why i have no audio when Bridge() two channels ?`
`21:39.02`	`nasls_lsa`	`how can I set-up Asterisk to use skype for outgoing calls ?`
`21:39.06`	`type0`	`http://www.voip-info.org/wiki/index.php?page=Asterisk+config+vpb.conf`
`21:39.08`	`type0`	`much like that.`
`21:39.15`	`FuriousGeorge`	`on the sip phone i would assume its the headset volume. i assumed they had it all the way up, or that it was only for inbound audio`
`21:39.35`	`*** join/#asterisk sasch (n=sasch@host136-64.pool8253.interbusiness.it)`
`21:39.36`	`type0`	`what kind of card are you using?`
`21:39.46`	`type0`	`or this is just straight zaptel?`
`21:40.02`	`FuriousGeorge`	`if ur asking me, its all sip, no card involved though i do have a tdm400p for inbound POTS and timing`
`21:40.22`	`type0`	`so the sip call is strictly sip, or its a phone going out of the tdm?`
`21:40.33`	`mercestes`	`type0: batting 20 today are ya?`
`21:40.39`	`FuriousGeorge`	`no analog until (maybe) the answering party when i call out`
`21:40.40`	`mercestes`	`FuriousGeorge: Don't suppose yoru using a polycom??`
`21:40.44`	`FuriousGeorge`	`snom`
`21:41.17`	`mercestes`	`FuriousGeorge: on a polycom yon can adjust the tx.gains. not sure on snom. Try a new snom and see if it gets better.`
`21:41.33`	`FuriousGeorge`	`guess i could swap it`
`21:41.41`	`type0`	`zapata.conf`
`21:41.42`	`type0`	`rxgain: Adjusts receive gain. This is the audio recieved by Asterisk from the device. E.g: in a phone connected to a FXS channel, this would control the audio that is sent from the phone to Asterisk. This can be used to raise or lower the incoming volume to compensate for hardware differences. You specify gain as a decimal number from -100 to 100 representing 100% to 100% of capacity. Default value: 0.0`
`21:41.42`	`type0`	`<PROTECTED>`
`21:42.06`	`FuriousGeorge`	`type0: i know`
`21:42.07`	`mercestes`	`type0: yes, I always use zapata.conf to configure my sip calls.`
`21:42.11`	`ManxPower`	`type0: that is wrong`
`21:42.18`	`ManxPower`	`type0: the values are in DB`
`21:42.21`	`type0`	`that's wrong?`
`21:42.25`	`type0`	`heh`
`21:42.34`	`FuriousGeorge`	`the call that is too quiet goes from my sip phone out via iax provider`
`21:42.36`	`type0`	`do if the rxgain=4.2 that would be 4.2DB right?`
`21:42.43`	`type0`	`s/do/so`
`21:42.44`	`ManxPower`	`type0: In theory`
`21:42.55`	`type0`	`http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf`
`21:42.58`	`ManxPower`	`FuriousGeorge: you need to adjust the gain on the phone`
`21:43.15`	`geejay101`	`ocgltd - quite frankly I find asterisk incomprehensible in some aspects and error hunting fruitless since it takes a huge effort to understand the code - consider yate for h323-SIP conversion.`
`21:43.17`	`type0`	`i guess the real question is, is it the same for all phones on the same channel, or just one phone on any channel?`
`21:43.34`	`mercestes`	`there are no sip channels.`
`21:43.50`	`FuriousGeorge`	`ManxPower: yeah, i assumed that either they would have put it all the way up already, or that the setting im familiar with is only for the ear piece, not the mike`
`21:43.55`	`FuriousGeorge`	`but i can look into that`
`21:44.03`	`FuriousGeorge`	`*mic`
`21:44.09`	`type0`	`ok, sip is a channel.. i was wrong`
`21:45.03`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`21:45.05`	`type0`	`still though`
`21:45.07`	`type0`	`turn up the gain`
`21:45.21`	`ManxPower`	`FuriousGeorge: what brand of phone?`
`21:45.57`	`*** part/#asterisk mut (n=ana@65.111.222.120)`
`21:46.25`	`creature_`	`grr my atabox doesnt understand or send any information to asterisk that the phone has been hanged up`
`21:46.36`	`type0`	`hung up?`
`21:46.49`	`creature_`	`hung up maybe :)`
`21:46.50`	`FuriousGeorge`	`ManxPower: snom360`
`21:46.59`	`creature_`	`sometimes doesn't make the grammar right :)`
`21:47.05`	`nasls_lsa`	`how can I set-up Asterisk to use skype for outgoing calls ? is that do-able without buy software ?`
`21:47.23`	`ManxPower`	`FuriousGeorge: I've never used them.`
`21:47.40`	`type0`	`ive seen someone use vonage as an extension`
`21:47.42`	`FuriousGeorge`	`ManxPower: they work nice with asterisk i got a friend who is a part time conspiracy theorist`
`21:48.29`	`mercestes`	`all conspiracy theorists are out to get me.`
`21:48.32`	`FuriousGeorge`	`says other isps deprioritize voip`
`21:48.33`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-227-3.cablep.bezeqint.net)`
`21:49.37`	`creature_`	`nasls_lsa: http://www.voip-info.org/wiki-Skype+Gateways`
`21:49.58`	`nasls_lsa`	`thaaat I am reading right now , thanks :)`
`21:50.04`	`creature_`	`=)`
`21:50.15`	`creature_`	`don't know of any free software, never needed to use such a gateway`
`21:50.15`	`nasls_lsa`	`freeware right ?`
`21:50.18`	`ManxPower`	`FuriousGeorge: I imagine they would depriortize bittorrent first`
`21:50.24`	`nasls_lsa`	`aaah :(`
`21:50.33`	`creature_`	`nasls_lsa: they link to chanskype which isn't freeware`
`21:51.03`	`russellb`	`ugh, "freeware" is a windows term :-p`
`21:51.10`	`nasls_lsa`	`I checked that .. :/ ..`
`21:51.26`	`wunderkin`	`Phreewarez`
`21:51.33`	`russellb`	`i can send you the chan_skype source code, but it won't do you much good`
`21:51.34`	`nasls_lsa`	`:>`
`21:51.41`	`russellb`	`it requires a proprietary kernel module that does all the real work`
`21:51.50`	`russellb`	`but trust me, you don't want this code ... it's pretty terrible`
`21:51.52`	`ManxPower`	`I just realized today is the last day of the month. The used car salesmen should be desperate. evil laugh`
`21:52.19`	`boch`	`Do you know why i have no audio when Bridge() two channels ?`
`21:52.29`	`masked`	`who's ya daddy?`
`21:52.43`	`masked`	`LOLZ`
`21:52.45`	`masked`	`excuse me`
`21:52.47`	`PaulTech85`	`..?`
`21:52.55`	`nasls_lsa`	`I don't know what VoIP provider to use for outgoing calls .. any good ideas for Greece ?`
`21:52.57`	`funnymanva`	`chanskype is only $19 for a single channel. That's darn close to free.`
`21:53.23`	`nasls_lsa`	`$19 lifetime ? or /month / year ..?`
`21:53.40`	`nasls_lsa`	`does it run on linux so I can use one pc for that ?`
`21:53.59`	`russellb`	`funnymanva: too bad they infringe on copyright ...`
`21:54.18`	`creature_`	`oh i'm getting so frustrated over my atabox right now :(`
`21:55.29`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`21:57.04`	`funnymanva`	`it's $19 lifetime, and it runs skype in a VNC on your linux box and connects to Asterisk as a channel.`
`21:57.30`	`funnymanva`	`russelb: didn't know that.`
`21:57.43`	`FuriousGeorge`	`ManxPower: i said the same thing (that they would block bit torrent first), but then i pointed out that cablevision and to a greater extent verizon are the ISPs around here, and they are in the phone service, not file sharing`
`21:57.43`	`ocgltd`	`geejay101 - does yate do T.38 over both protocols?`
`21:58.39`	`*** join/#asterisk kore (i=kore@mindwipe.org)`
`21:59.16`	`FuriousGeorge`	`err meant to say "they are in the phone business"`
`22:00.51`	`aydiosmio`	`[TK]D-Fender: Dial(SIP/17275551234@70.52.18.10,,mghHwWM(record\|${recordid})) */1 works for the caller but not the callee, any ideas?`
`22:00.51`	`*** part/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com)`
`22:00.51`	`type0`	`i dont think its legal to deprioritize voip service`
`22:00.51`	`type0`	`since E911 is going over it, which is a critical service`
`22:01.24`	`*** join/#asterisk bkw_ (n=brian@dsl093-079-130.sfo1.dsl.speakeasy.net)`
`22:02.22`	`type0`	`they could make the arguement that selling dialtone over a tarriffed line is against the law i guess`
`22:02.43`	`aydiosmio`	`how would you know?`
`22:02.56`	`aydiosmio`	`any carrier on the route could drop your call`
`22:03.29`	`type0`	`right, and when someone dies because they called 911 from a vonage line and couldnt get through because the ISP dropped their data .. game over`
`22:03.40`	`aydiosmio`	`gane over for who?`
`22:03.43`	`aydiosmio`	`game`
`22:03.45`	`type0`	`the telco`
`22:03.52`	`aydiosmio`	`who is the telco?`
`22:04.01`	`type0`	`whoever's selling internet`
`22:04.15`	`creature_`	`Anyone here who uses PAP2T?`
`22:04.30`	`[TK]D-Fender`	`aydiosmio : Just dialing an IP like that I'm betting * can't guess a DTMFMODE and doesn't accept ANYTHING back. Make a peer entry for it and set a mode`
`22:05.03`	`aydiosmio`	`[TK]D-Fender: thanks I'll try it`
`22:06.35`	`JoNate`	`[TK], you have any experience with * and cyberguard routers?`
`22:06.55`	`[TK]D-Fender`	`JoNate : Nope.`
`22:07.22`	`JoNate`	`damn me! I've opened every port on the damn thing, and still no dice`
`22:07.55`	`aydiosmio`	`packet sniff both sides`
`22:08.01`	`[TK]D-Fender`	`JoNate : Perhaps you could explain your actual probelms, and not just mention a piece of its puzzle`
`22:08.09`	`aydiosmio`	`make sure thigns are getting through and addressed properly`
`22:09.40`	`geejay101`	`ocgltd unfortunately I am not familiar with T.38. But if T.38 is handled fully in the RTP stream then yate should be able to handle that because you could simply forward the RTP directly between endpoints - yate can simply handle the SIP-H323 signalling side and does not need to see the RTP.`
`22:09.49`	`JoNate`	`[TK]: I think I'm missing pieces of the puzzle, but I can't dial out, I get a busy/congested issue, when I am behind this firewall/router. However, if I set the * box with the public IP directly, everything is gravy`
`22:10.26`	`[TK]D-Fender`	`JoNate : So its a NAT router, * is behind it, and calls jsut aren't working?`
`22:10.47`	`JoNate`	`[TK]: Yep, I get a fast busy`
`22:11.17`	`diclophis-work`	`would codec negotiation cause a delay in the "answering" of a channel`
`22:11.33`	`[TK]D-Fender`	`JoNate : Pastebin your entire [general] section of sip.conf`
`22:11.34`	`[TK]D-Fender`	`~pb`
`22:11.42`	`jbot`	`i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`22:11.54`	`[TK]D-Fender`	`diclophis-work : Nope.`
`22:12.20`	`diclophis-work`	`what would cause a channel to "answer slowly", by that i mean, answer, but deliver silence for ~3-4seconds`
`22:12.36`	`JoNate`	`Gah! I'm getting called into a meeting!`
`22:12.53`	`aydiosmio`	`don't tell them you failed`
`22:13.20`	`[TK]D-Fender`	`diclophis-work : Describe the call-path from end-end`
`22:14.32`	`diclophis-work`	`[TK]D-Fender: ok, here goes.... A call is delivered to BOXA through a Zap channel, then delivered to BOXB over SIP, where by it get sent through a Queue, and down a Local channel, which then gets delievered back to BOXA over a SIP channel`
`22:15.00`	`diclophis-work`	`when BOXB is trying to get back to BOXA, there is a 3-4sec silence period before any voice is deliever to BOXB`
`22:15.27`	`creature_`	`When the other end user hangs up and i leave my cordless phone alone, it SHOULD detect that the phone call is over and end automatically. However when i hangup my analogue phone connected to my atabox (PAP2T) it just plays a "busy tone" followed by a louder (busy?) tone. I think its because the PAP2T doesnt handle PSTN and therefore its no reason for it to go off hook. I tried to change something called CSC but that wont solve it. Anyone`
`22:16.19`	`*** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189)`
`22:16.59`	`[TK]D-Fender`	`diclophis-work : Pastebin a complete call`
`22:17.11`	`tzafrir_laptop`	`do the span parameters in zaptel.conf (framing, coding, etc.) affect the layer 1 connectivity?`
`22:17.14`	`*** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-147-114.ny325.east.verizon.net)`
`22:17.16`	`boch`	`[TK]D-Fender, have you dealed with Bridge() app ?`
`22:17.41`	`[TK]D-Fender`	`boch : nope.`
`22:17.43`	`tzafrir_laptop`	`I fail to get layer 1 connectivity somewhere. I wonder if there's any point of playing with them`
`22:18.15`	`JT`	`yes they should affect L1, otherwise nothing does...`
`22:18.24`	`[TK]D-Fender`	`creature_ : AFAIK all of the linksys/sipura only give the annoying re-order tone on disconnect. Heck their HARD PHONES work that way too...`
`22:20.13`	`aydiosmio`	`is there a show that will display a channel dtmf mode? I can't tell if my outbound supports rfc2833`
`22:20.26`	`creature_`	`[TK]D-Fender: Ok. This is a big problem :(`
`22:20.51`	`[TK]D-Fender`	`aydiosmio : "sip show channel [channelnamefromshowchannels]"`
`22:21.14`	`[TK]D-Fender`	`creature_ : No... only YOURS. Try another ATA`
`22:21.37`	`creature_`	`[TK]D-Fender: If i hangup my phone connected to the PAP2T the channel isn't closed, remains open until the remote caller has hanged up`
`22:21.50`	`creature_`	`Ow, thats how it should work :D`
`22:21.54`	`[TK]D-Fender`	`creature_ : Wait... you mean hanging up the ATA side?!`
`22:22.15`	`Mahmoud`	`hmm where can i get libc.so.6?`
`22:22.19`	`creature_`	`[TK]D-Fender: Wait, i'll try to explain this in a better way`
`22:22.20`	`[TK]D-Fender`	`creature_ : Pastebin a sample of this.`
`22:22.26`	`creature_`	`Ok`
`22:24.28`	`type0`	`why are 1U voip servers so expensive`
`22:25.23`	`Qwell[]`	`"voip servers"?`
`22:25.30`	`FuriousGeorge`	`when i have her call me of course it works fine... maybe its just congestion with my isp/bane`
`22:25.49`	`FuriousGeorge`	`isnt 2-4 peak hours for internet usage? (after lunch)`
`22:25.59`	`aydiosmio`	`DTMF Mode: rfc2833`
`22:26.09`	`aydiosmio`	`called party supports it:(`
`22:26.16`	`[TK]D-Fender`	`aydiosmio : make a peer, make sure.`
`22:26.23`	`aydiosmio`	`I did`
`22:26.45`	`aydiosmio`	`I set dtmfmode=rfc2833 for the peer`
`22:27.42`	`[TK]D-Fender`	`aydiosmio :*/1 = ?`
`22:27.57`	`aydiosmio`	`hangup/automon`
`22:28.40`	`[TK]D-Fender`	`aydiosmio : Whats in that macro?`
`22:29.11`	`aydiosmio`	`Wait, SayDigits`
`22:29.18`	`aydiosmio`	`happens with or without the macro`
`22:29.42`	`JT`	`i'll second that Qwell[]`
`22:29.46`	`JT`	`type0: "voip servers"?`
`22:30.00`	`aydiosmio`	`why woudl you use 1U?`
`22:30.05`	`aydiosmio`	`2U are better for expansion cards`
`22:30.18`	`JT`	`aydiosmio: err, rackspace costs`
`22:30.30`	`JT`	`sure you don't always need expansion cards, especially if it's voip only`
`22:30.34`	`JT`	`or more than 1`
`22:30.36`	`LostFrog`	`If you just need network cards, 1U make senses..`
`22:30.42`	`JT`	`1RU servers can do 1 or 2 PCI cards`
`22:31.14`	`LostFrog`	`I'm tired of finding low-profile cards.`
`22:31.33`	`JT`	`just remove the backplate then :P`
`22:31.39`	`JT`	`some do full profile cards`
`22:31.44`	`LostFrog`	`Some do.`
`22:31.58`	`LostFrog`	`I'm going to blades soon.`
`22:32.04`	`JT`	`so get one that does :)`
`22:32.12`	`LostFrog`	`If I can talk my boss into it.`
`22:32.14`	`JT`	`some datacentres are banning or restricting blades now`
`22:32.23`	`LostFrog`	`Why so?`
`22:32.36`	`JT`	`exceeding their power density capacity`
`22:32.42`	`aydiosmio`	`hahaha`
`22:32.43`	`JT`	`W/sq m`
`22:32.55`	`aydiosmio`	`that's no fun`
`22:33.31`	`JT`	`indeed`
`22:34.36`	`creature_`	`[TK]D-Fender: i ran some tests now.. seems like i only have problem in one situation. that is if i call either from sip client to analogue phone or the other way around.. and then hangs up on the analogue phone.`
`22:34.46`	`creature_`	`i will pastebin that`
`22:38.21`	`creature_`	`http://www.pastebin.ca/376234`
`22:41.05`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`22:41.18`	`Mahmoud`	`hmmm i want to test my asterisk with someone here`
`22:41.24`	`Mahmoud`	`just to know if my ISP blocks 5060 or not`
`22:41.37`	`Mahmoud`	`any one wants to join me for testing ?`
`22:41.47`	`Mahmoud`	`i'll make for him a temp account + password`
`22:42.51`	`creature_`	`Mahmoud: no need for that, msg me the ip and i will check to see if i can get any connection on your port 5060`
`22:43.00`	`Mahmoud`	`cool`
`22:43.04`	`*** join/#asterisk deb_user (n=none@70-59-111-238.albq.qwest.net)`
`22:43.15`	`Mahmoud`	`are you behind nat?`
`22:43.16`	`wunderkin`	`192.168.2.290`
`22:43.24`	`deb_user`	`i'm trying to use mysql_cdr`
`22:43.28`	`generalhan`	`ok guys, i need to ask what maybe a really dumb question. on a remote phone's configs when it asks for registrar ip and proxy ip, those addresses will be different right ? like the proxy ip is the WAN IP that im going out of, and the registrar ip is the ip of the * box. is that correct ?`
`22:43.29`	`creature_`	`Mahmoud: no`
`22:43.59`	`deb_user`	`cdr status gives me: CDR mode: simple CDR registered backend: cdr-custom`
`22:44.08`	`deb_user`	`that doesn't seem right...`
`22:44.13`	`deb_user`	`can anybody offer some tips?`
`22:44.14`	`mercestes`	`generalhan: It's your PBX ip.`
`22:44.41`	`mercestes`	`generalhan: The idea is you can have a register server, and a proxy server (sip gateway), and a routing server, etc. I never got any of that to work.`
`22:44.50`	`mercestes`	`generalhan: Atleast...that's my understanding of it`
`22:45.10`	`deb_user`	`and...from the cli how can I tell if * is connecting to mysql?`
`22:45.26`	`generalhan`	`so the 2 ips WILL be the same then ?`
`22:45.43`	`generalhan`	`i dont technically have a proxy in the remote location.`
`22:45.54`	`generalhan`	`which i think is why im getting so confised here ! lol`
`22:46.06`	`generalhan`	`why cant the remote setup be as simple as the local ones !! lol`
`22:46.06`	`mercestes`	`generalhan: "proxy" to me has always been the server you dial out of in my phone configs`
`22:46.18`	`generalhan`	`ok ... lets try that i guess !`
`22:47.46`	`diclophis-work`	`if i am using a Local channel as an interface for a Queue member, i can put a Hangup in there right?`
`22:51.48`	`deb_user`	`ok...`
`22:51.54`	`deb_user`	`I can't connect to mysql server`
`22:52.15`	`*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)`
`22:52.37`	`tzafrir_laptop`	`deb_user, can you connect to mysql with the command-line mysql?`
`22:52.53`	`deb_user`	`tzafrir: i'm pretty sure`
`22:52.56`	`deb_user`	`lemme try though`
`22:53.34`	`*** join/#asterisk fastfeet (n=fastfeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com)`
`22:53.52`	`deb_user`	`tzafrir_laptop: yes, I can connect no prob`
`22:54.03`	`*** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud)`
`22:54.22`	`tzafrir_laptop`	`with the same username, pass and database name as in the asterik config?`
`22:54.36`	`JT`	`generalhan: they should all be the same unless your ITSP advises otherwise`
`22:54.43`	`deb_user`	`yes sir`
`22:55.10`	`JT`	`in which case your ITSP is stupid, as there is no technical reason why they need to be different IPs`
`22:55.28`	`generalhan`	`ITSP ?`
`22:55.40`	`tzafrir_laptop`	`deb_user, maybe there is an incorrect socket setting?`
`22:55.49`	`deb_user`	`nope`
`22:55.55`	`deb_user`	`I greped the socket setting for mysql`
`22:55.57`	`deb_user`	`to make sure`
`22:56.08`	`deb_user`	`and inserted it directly into the configuration file`
`22:56.36`	`deb_user`	`althoug cdr status does give me: CDR mode: simple`
`22:56.44`	`generalhan`	`see on all my configs here they are the same address ... but they are local. for some reason i can not get any of these phones to work from a remote location. so i was thinking that maybe the proxy addy was suposed to be the remote WAN IP`
`22:56.45`	`deb_user`	`not exactly sure what that means...`
`22:57.26`	`*** part/#asterisk dasenjo (n=dasenjo@190.24.24.69)`
`22:58.25`	`diclophis-work`	`this doesnt make sense, it was working ealier`
`23:00.17`	`*** join/#asterisk benno2 (n=chatzill@host118-110-dynamic.9-87-r.retail.telecomitalia.it)`
`23:00.21`	`*** join/#asterisk stridernzl (n=neville@222-155-234-32.jetstream.xtra.co.nz)`
`23:03.13`	`*** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud)`
`23:03.13`	`*** join/#asterisk J4k3 (n=J4k3@node164.140.136.67.1dial.com)`
`23:03.25`	`benno2`	`hi, question: normally on asterisk I use internal numbers (for example 10,20 etc) and to dial outside 0 + intl prefix + number eg 0 001 ... is it possible to tell asterisk that if one enters let's say at least 5-6 digits then it's automatically an external call and that asterisk should simply add the first 0 so that the user can type 001 ...`
`23:04.42`	`Mahmoud`	`could someome try if he can connect to my * srever?`
`23:04.49`	`Mahmoud`	`i changed the port number to bypass my stupid isp`
`23:04.54`	`Mahmoud`	`they blocked 5060`
`23:05.49`	`Mahmoud`	`if any one wishes to help me test my pbx, please pm me`
`23:05.50`	`JT`	`benno2: depends, you need a completely non interfering dialplan with internal vs external numbers`
`23:06.15`	`LostFrog`	`I just dial 9 to get out..`
`23:06.17`	`LostFrog`	`Simple.`
`23:06.25`	`JT`	`generalhan: Internet Telephony Service Provider`
`23:08.30`	`benno2`	`JT: I am asking this because I have a gsm/voip mobile phone which is quite cool (german tc-300) and if you look up the number in the address book you can choose ... dial via gsm or dial via voip. but the phone would then send 001 ... and not 0 001 ... this is why I asked.`
`23:09.08`	`benno2`	`but I could simply memorize all numbers in international format and then tell asterisk to look at numbers that begin with 00`
`23:09.14`	`tzafrir_laptop`	`anybody here uses zaptel/bri? specifically in Itally?`
`23:09.16`	`deb_user`	`mahmoud: is your isp trying to kill voip traffic`
`23:09.31`	`Mahmoud`	`they want to make more money by their analog telephony`
`23:09.35`	`tzafrir_laptop`	`or otherwise BRI in Italy?`
`23:09.36`	`benno2`	`tzafrir I use bri, zap-hfc in italy works well`
`23:09.56`	`tzafrir_laptop`	`can you please give me a sample zaptel.conf?`
`23:10.57`	`JT`	`benno2: that's pretty unusual but cool`
`23:10.57`	`eald`	`mmm I in that case, I guess that what you want is Dial for 0+{$EXTEN} or whatever the RIGHT syntax is for that`
`23:11.15`	`deb_user`	`mahmoud: what country?`
`23:11.27`	`JT`	`benno2: how does it call via voip... gprs?`
`23:11.37`	`benno2`	`tzafrir_laptop: I just downloaded bristuff stable from junghanns , compiled and it worked`
`23:11.43`	`benno2`	`JT: no via wlan`
`23:12.01`	`JT`	`benno2: hrm ok, range can't be good`
`23:12.11`	`JT`	`i recommend the latest bristuff testing branch generally`
`23:12.19`	`JT`	`stable was so old and... crappy`
`23:12.31`	`Mahmoud`	`De_Mon, UAE`
`23:13.00`	`diclophis-work`	`anyone work with Queues and Local channels`
`23:13.11`	`benno2`	`JT: http://www.t-one.de click on Die Endgeraete .. on the left upper side`
`23:13.24`	`tzafrir_laptop`	`benno2, so you use the settings from the sample zaptel.conf...`
`23:13.26`	`JT`	`benno2: shrug, more interested in how it works in the real world :P`
`23:13.30`	`tzafrir_laptop`	`this is what I use as well`
`23:13.45`	`benno2`	`JT: the range is the normal range a wlan device can achieve`
`23:14.01`	`JT`	`benno2: that's not been true for other wifi sip phones`
`23:14.12`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`23:14.14`	`benno2`	`JT: I have not yet tried roaming yet, which means placing multiple access points and walking through them`
`23:14.25`	`JT`	`doesn't sound like fun`
`23:14.50`	`benno2`	`JT: in what sense ? that they are very bad in terms of reception ? basically the phone sees the access point of the neighbor while my acer laptop does not`
`23:15.15`	`aydiosmio`	`bah, no dtmf works on my outbound calls.`
`23:15.31`	`JT`	`yeah, and sip would hate speed changes, variable lag, jitter and packet loss`
`23:16.29`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`23:17.06`	`benno2`	`JT: it's a pity that's not more widely available ... I'm italian but I know german so I got it from the german ebay site, but here it's unknown and generally on the inet it's hard to find infos about it. I accidentally found out about while surfing a german voip forum because I was looking for reviews about the siemens voip phone (it does not provide GSM, its pure voip) and then there was a guy that said the tc300 is`
`23:17.11`	`Mahmoud`	`any idea how to change x-lite's SIP soft phone destination port number statically without using SRV records?`
`23:17.52`	`*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)`
`23:18.06`	`JT`	`benno2: yeah i think i'll try and steer clear of it personally, 802.11b/g is not a good protocol for voip in a mobile configuration`
`23:18.11`	`generalhan`	`can i just use IAX to connect these remote phones instead of SIP ? stupid SIP ! lol`
`23:19.01`	`benno2`	`JT: as long as there are no alternatives what can we do ? for home use it's ok , I'm waiting for WiMax phones :)`
`23:19.17`	`JT`	`i'll stick to cordless phones :P`
`23:19.23`	`benno2`	`JT: is the bristuff testing branch stable in your opinion ?`
`23:19.37`	`JT`	`more stable in my opinion, if you get the right version`
`23:19.48`	`JT`	`and asterisk 1.0.x just sucks`
`23:19.55`	`JT`	`no priority n, wtf`
`23:20.28`	`benno2`	`JT: but with cordless phones you don't go to a foreign country, log in to a hotspot (the phone even has a small http/wap browser so hotspots that require http login should work) and then use it as a local extension of your pbx`
`23:20.51`	`JT`	`i wouldn't trust sip to work at a random hotspot anyway`
`23:21.29`	`benno2`	`JT: I compiled the asterisk version that is indicated in bristuff, basically I used download.sh & compile.sh :)`
`23:21.30`	`aptura`	`unless the hotspot was owned by the same company such as fatport.`
`23:21.48`	`benno2`	`aptura: what is fatport ?`
`23:21.51`	`JT`	`benno2: yes bristuff stable uses asterisk 1.0.x`
`23:21.59`	`JT`	`benno2: which is ancient`
`23:22.25`	`benno2`	`JT: so you think that my uptimes will not suffer if I use the testing branch (which uses asterisk 1.2.x)`
`23:22.35`	`JT`	`what card do you use?`
`23:22.36`	`[TK]D-Fender`	`benno2 : Bristuff uses ASTERISK.... tha somehow sounds massively BACKWARDS...`
`23:22.48`	`JT`	`what uptimes do you currently experience?`
`23:22.53`	`benno2`	`JT: zap-hfc compatible (an italian card)`
`23:23.02`	`JT`	`[TK]D-Fender: ?`
`23:23.26`	`benno2`	`[TK]D-Fender: sorry :) s/uses/requires :)`
`23:24.58`	`aptura`	`btw I am trying to find the most likely caue of why asterisk takes up to 8 seconds to dial a local pstn call but is instantanios with wholesale voip. Did some googling around did not come up with much of a answer. Cli has not responce until the eight second threshhold is passed. It could also be the sipura adapter.`
`23:24.59`	`benno2`	`JT well I cannot tell because I have no UPS attached to the mini-itx box running it so each week or so the power goes off .. but I must say it always worked and the ISDN has always worked. with older versions of asterisk (but it was not a hfc card) I remember the ISDN simply freezing after a few days or so`
`23:25.27`	`benno2`	`aptura: I had similar problems which were DNS related, if DNS was not working the SIP dialing was slow`
`23:25.37`	`JT`	`aptura: sounds like timeout dialling`
`23:25.51`	`JT`	`aptura: sharpen up your dialplans and you won't use timeout dialling`
`23:25.55`	`JT`	`on your ATA`
`23:26.10`	`aptura`	`okay I did read something about dns. Was your adapter on the local network logging in to the local asterisk box?`
`23:27.01`	`[TK]D-Fender`	`aptura : You are somewhat vague about the exact hardware in this call path. Please clarify`
`23:27.02`	`JT`	`benno2: i have bristuff systems running for over a month at a time`
`23:27.32`	`benno2`	`JT: nice but it never crashed or freezed ?`
`23:27.34`	`*** join/#asterisk bkw_ (n=brian@dsl093-079-130.sfo1.dsl.speakeasy.net)`
`23:27.42`	`*** join/#asterisk dseeb_ (n=dcb@CPE-58-169-73-237.vic.bigpond.net.au)`
`23:27.47`	`aptura`	`TK, I am going to try the DNS then the dial plan before going on to the card.`
`23:27.49`	`JT`	`benno2: mind you i'm using qozap no zap-hfc, but all indications are that zap-hfc is even more stable`
`23:28.13`	`JT`	`benno2: not since 0.3.0-PRE-1w`
`23:28.29`	`JT`	`benno2: i use TE and NT mode too`
`23:28.33`	`JT`	`most people only use TE`
`23:28.39`	`JT`	`which seems universally more stable`
`23:28.43`	`JT`	`due to more people using it`
`23:28.46`	`benno2`	`I use only TE`
`23:29.00`	`aptura`	`JT, which dns server did you use?`
`23:29.14`	`JT`	`aptura: what?`
`23:29.15`	`benno2`	`I think asterisk is godsend for people that travel a lot`
`23:29.32`	`JT`	`aptura: what has the dialplan got to do with dns?`
`23:29.35`	`aptura`	`benn02 thats for sure :)`
`23:29.35`	`[TK]D-Fender`	`aptura : "the" dialplan, "the" phone, "the" card. PRONOUNS GODDAMMIT!`
`23:30.12`	`benno2`	`what I find extemely disturbing are european cellphone roaming costs`
`23:30.32`	`benno2`	`with an italian phone while in germany I pay an arm and a leg to only receive calls`
`23:30.37`	`JT`	`heh`
`23:30.51`	`JT`	`yeah i think it's cheaper to use an Iridium satellite phone sometimes :P`
`23:30.56`	`JT`	`than roaming`
`23:31.03`	`benno2`	`this is why I am handing out my PSTN number instead of my cell phone`
`23:31.23`	`benno2`	`and will use call forwarding and possibly multiple SIM cards one for each country`
`23:31.36`	`generalhan`	`i was pleasantly suprised with european cell phones. i went to paris and walked into a wireless store picked up a new account on a new phone in about 15 minutes tops.`
`23:31.59`	`generalhan`	`worked great ... i didnt do any roaming though !`
`23:32.10`	`benno2`	`I'm curious when the GSM operator monopoly will fall`
`23:32.33`	`JT`	`gsm is dieing in most other parts of the world`
`23:32.39`	`JT`	`people moving to 3/4G services`
`23:32.47`	`benno2`	`I find 20-25cent/min to call my neighbors cellphone robbery`
`23:32.52`	`JT`	`and the fact that GSM works like shit in a lot of countries`
`23:33.33`	`benno2`	`JT in europe GSM is still the only alternative, video calls over UMTS are even more expensive not to talk about data plans ... 4-5euro/MByte :)`
`23:33.44`	`JT`	`heh`
`23:34.13`	`JT`	`gsm is the most common in .au`
`23:34.24`	`JT`	`but market share will be seriously erroded in a couple of years`
`23:34.35`	`JT`	`lots of 3G and 4G services now`
`23:35.37`	`benno2`	`I really hope that WiMax will put an end to this robbery. but on the other hand in many european countries UMTS licenses were sold for stratospheric prices ... so cellphone operators need to recoup costs and are probably "protected" by the state, which means they will probably place hurdles if wimax operators want to enter the scene`
`23:36.11`	`*** join/#asterisk Bobthehunter (n=Bobthehu@145-27.mc.cite.net)`
`23:36.11`	`Bobthehunter`	`<PROTECTED>`
`23:36.11`	`benno2`	`JT: but what are the 3g,4g prices ? same operators, same prices, same mafia ?`
`23:36.28`	`JT`	`i don't think wimax is a mobile phone technology, thought it was data really`
`23:36.30`	`Mahmoud`	`any free dynamic and public SRV services?`
`23:36.45`	`JT`	`benno2: yes and no, some providers are a lot cheaper than others`
`23:36.58`	`JT`	`the networks with the best coverage cost the most to use`
`23:37.34`	`benno2`	`JT: so what are typical cellphone to cellphone prices ? how many cent/min ?`
`23:38.08`	`JT`	`benno2: i think australia is recorded as having the highest, or one of the highest mobile phone adoption rates in the world, funny thing is it's definately not the cheapest prices`
`23:38.22`	`Bobthehunter`	`?`
`23:38.24`	`JT`	`benno2: varies between 15c/min to $1/min depending on network and plan`
`23:38.30`	`JT`	`AUD`
`23:38.56`	`JT`	`medium of around 35/min maybe`
`23:39.47`	`[TK]D-Fender`	`Bobthehunter : Perhaps you could actually phrase that in the form of an intelligable question...`
`23:40.09`	`creature_`	`http://pastebin.ca/376331 <- I have a problem with this macro. If the call get answered the NoOp(EXECUTING HANGUP); and Hangup; is never runned. Why's that?`
`23:40.36`	`creature_`	`in case it matches any dialstatus that part is runned.`
`23:40.40`	`creature_`	`which is correct`
`23:41.05`	`benno2`	`JT: so the prices are similar like here in europe. still expensive`
`23:41.09`	`*** join/#asterisk fastfeet (n=fastfeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com)`
`23:41.48`	`JT`	`benno2: actually they're pretty good if you are willing to commit to a cap`
`23:41.58`	`JT`	`often calls and text are cheaper to the same network too`
`23:42.19`	`JT`	`i've heard of couples sending 40000 SMS to each other a year, and not going over $50/mo cap plan`
`23:42.28`	`JT`	`they were on the same network`
`23:42.53`	`JT`	`benno2: the rates are also much better on the networks with less coverage`
`23:43.58`	`CrashHD`	`with 1.2.x do sip agents still show status (in use)?`
`23:44.36`	`benno2`	`JT: sure, calling people on the same network certainly saves money but murphy's law says that you've got friends of business partners that use different networks and you need to call them all the time. given that each country has at least 3-4 gsm operators one would need to run around with 4 cellphones in order to save money ... or just call asterisk and let it dispatch calls`
`23:44.47`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`23:44.59`	`benno2`	`but unfortunately voip provider still charge a lot to call cellphones`
`23:45.00`	`*** join/#asterisk Vec (n=Vec@dsl-244-219-12.telkomadsl.co.za)`
`23:45.05`	`JT`	`yeah calling asterisk is a cool hack :)`
`23:46.13`	`Vec`	`I have compiled zpatel and libpri and installed a TE110P pri card, loaded the module and everything seems fine, but when I start asterisk I get "WARNING[8775]: loader.c:362 load_dynamic_module: Error loading module 'chan_zap.so': libpri.so.1.0: cannot open shared object file:No such..", no idea why this is happening or even if its a problem ?`
`23:47.19`	`benno2`	`jt: but the problem is by going through asterisk in order to save money you loose the phonebook functionality of your phone. first dial asterisk and then manually dial the number you want to dial`
`23:47.36`	`aptura`	`JT, made some changes to the adapter. 4 seconds for cli to respond and 8 seconds for the other end to ring.`
`23:47.37`	`benno2`	`it's annoying ... but for long distance and longer duration calls it definitively pays off`
`23:48.13`	`JT`	`benno2: i think some phones let you send dtmf of an address book number once connected`
`23:48.23`	`aptura`	`Benn02, you travel alot with a ATA adapter over seas and had good results so far?`
`23:48.52`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`23:49.13`	`*** join/#asterisk oQPa (n=uawename@237.Red-81-39-146.dynamicIP.rima-tde.net)`
`23:49.30`	`Vec`	`brb`
`23:49.44`	`benno2`	`aptura: I used voip mostly using a windows laptop and a sip client like xten, it worked well in most cases. with an ATA you should achieve similar results`
`23:50.02`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`23:50.04`	`*** part/#asterisk oQPa (n=uawename@237.Red-81-39-146.dynamicIP.rima-tde.net)`
`23:50.32`	`benno2`	`JT: yes that dtmf feature is cool but unfortunately my combined voip/gsm phone does not have that feature :(`
`23:50.41`	`JT`	`hmm`
`23:50.53`	`FuriousGeorge`	`i think im gonna cycle my providers, collect data, and see if i can find any correlation between "robotic sound" or "choppy sound" and time of day or provider used... i talked to one of the higher level techs at the isp, and they were like "yeah, our optimum voice customers complain about that too"`
`23:50.56`	`*** join/#asterisk Vec (n=Vec@dsl-244-219-12.telkomadsl.co.za)`
`23:52.21`	`FuriousGeorge`	`so, maybe ill call verizon about a t1 or something`
`23:53.05`	`wunderkin`	`'Verizon.. The New AT&T'`
`23:53.09`	`FuriousGeorge`	`...no one offers sdsl around here for some reason, except brand x providers`
`23:53.25`	`FuriousGeorge`	`wunderkin: that doesnt make me feel any better :)`
`23:53.50`	`benno2`	`question, what does it cost to receive a call on a cellphone in the US ?`
`23:54.03`	`aptura`	`<PROTECTED>`
`23:54.43`	`benno2`	`in europe we don't pay for inbound calls, in the US AFAIK yes, but OTOH calling cellphones costs like calling fixed lines`
`23:54.50`	`aptura`	`JT, what did you do to speed up your dial responce for local pstn calls? I have done two of the three things mentioned.`
`23:55.11`	`techie`	`yeah, the NEW AT&T...`
`23:55.20`	`JT`	`aptura: checked the ata dialplan?`
`23:55.29`	`benno2`	`aptura: so can you give an example of a plan ? for example if I get a call from a landline ... 10min ... how much is it going to cost ?`
`23:55.41`	`aptura`	`yes. Included one for local area code then 7 x marks`
`23:55.59`	`aptura`	`benn I dont know`
`23:56.16`	`aptura`	`But here in canada it can be unlimited incomming`
`23:57.12`	`FuriousGeorge`	`i need a third provider for my variables, can anyone recommend a good one theyve been using`
`23:58.05`	`FuriousGeorge`	`<PROTECTED>`
`23:58.45`	`benno2`	`what I am wondering about (at least in italy and germany) there seem many people which get rid of their landline in order to save some money but on the other hand cellphone dataplans are very expensive. so people wanting internet are likely to keep their landline`
`23:59.06`	`*** join/#asterisk topping (n=topping@dsl093-079-130.sfo1.dsl.speakeasy.net)`
`23:59.08`	`benno2`	`it will be an interesting phenomena to follow`
`23:59.08`	`aptura`	`.`
`23:59.39`	`*** join/#asterisk topping (n=topping@dsl093-079-130.sfo1.dsl.speakeasy.net)`
`23:59.45`	`FuriousGeorge`	`if you dont want to incite a riot, feel free to /msg me your provider of choice`
`23:59.58`	`wunderkin`	`livevoip`

Generated by irclog2html.pl by Jeff Waugh - find it at freshmeat.net! Modified by Tim Riker to work with blootbot logs, split per channel, etc.