| 00:00.13 | RoyK | so bug mashals can always blame people for not having the latest version of whatever driver |
| 00:00.38 | *** part/#asterisk anthonypjshaw (n=parliame@cpc2-rdng3-0-0-cust656.winn.cable.ntl.com) |
| 00:01.02 | grandy | Hello... quick question: Is there a way in the dialplan to see if Dial(foo) is ringing? Or to fail it if it does anything other than ring? |
| 00:01.56 | RoyK | sip debug? |
| 00:02.31 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 00:04.07 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 00:09.10 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
| 00:09.43 | perd | well my extensions.conf snippet should work |
| 00:09.48 | perd | you can use that to verify your install |
| 00:12.25 | *** join/#asterisk battini (n=inittab@cpe-24-209-36-174.neo.res.rr.com) |
| 00:12.31 | *** join/#asterisk Dovid (n=Dovid@ool-43530a83.dyn.optonline.net) |
| 00:12.35 | Dovid | hello all |
| 00:12.39 | Dovid | long time no time ;) |
| 00:12.54 | perd | aloha |
| 00:12.58 | Dovid | using asterisk real time if I want to send a voicemail via email to more than one address what do I have to do |
| 00:13.02 | *** join/#asterisk kieranmullen2 (n=kieranmu@static-71-245-97-83.ptldor.fios.verizon.net) |
| 00:13.13 | Dovid | usera@domain.com;userb@domain.com ? |
| 00:13.29 | Dovid | or usera@domain.com|userb@domain.com ? |
| 00:14.00 | kieranmullen2 | Where are some good asterisk related forums? voip-info should have forums but I guess that they dont want to manage it |
| 00:14.02 | perd | i dont know, unix uses , to separate addresses normally though |
| 00:14.09 | kieranmullen2 | beside son the digiu, site of course |
| 00:14.12 | perd | by unix i mean mail/senxmail/mailx/etc |
| 00:14.32 | perd | asteriskguru.org also |
| 00:14.37 | perd | .com i mean |
| 00:15.18 | kieranmullen2 | how well is asterisk faxing working for people? |
| 00:15.27 | kieranmullen2 | I have ordered a pots line for it |
| 00:15.31 | grandy | Hello... quick question: Is there a way in the dialplan to see if Dial(foo) is ringing? Or to fail it if it does anything other than ring? |
| 00:15.34 | kieranmullen2 | I have not had a pots line in 5 years |
| 00:15.39 | perd | i use iaxmodem and faxing is awesome |
| 00:15.48 | perd | hooks right into hylafax |
| 00:16.43 | kieranmullen2 | if asterisk recioeevd the fax when do you need hyla for? |
| 00:16.50 | kieranmullen2 | outbound? |
| 00:16.53 | perd | it doesnt receive fax |
| 00:17.11 | kieranmullen2 | what "it"? |
| 00:17.24 | perd | asterisk passes the faxes to the iax channel, the iax channel is being listened to by hylafax via the iaxmodem, etc |
| 00:17.53 | perd | it's easy to set up and use if you're familiar with hylafax |
| 00:17.59 | kieranmullen2 | i see... I saw an option from within free pbx for faxes... |
| 00:18.09 | Dovid | using asterisk real time if I want to send a voicemail via email to more than one address what do I have to do |
| 00:18.11 | Dovid | usera@domain.com;userb@domain.com ? |
| 00:18.13 | Dovid | or usera@domain.com|userb@domain.com ? |
| 00:18.15 | kieranmullen2 | so I just assumed.. guess I was wrong.. wonder when they use |
| 00:18.16 | perd | yeah you can use spandsp and txfax rxfax |
| 00:19.35 | *** part/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 00:19.43 | kieranmullen2 | I am looking for someone to setup a faxbox/voicemail box for multiple users based on extensions call in to pots line to zaptel card...enter in extension.. listen to message.. leave message or start fax....then have it emailed off. |
| 00:19.49 | SomeOne1 | i'm having problems with DTMF, sometimes it counts a digit twice and sometimes it misses it... what should i do? |
| 00:20.23 | kieranmullen2 | previous virtual pbx service from accessline.com had such a feature but I canceled them |
| 00:20.39 | perd | easily doable with either spandsp and txfax/rxfax or iaxmodem |
| 00:20.54 | perd | zaptel gives you faxdetect |
| 00:21.08 | kieranmullen2 | not using hyla |
| 00:21.16 | kieranmullen2 | yeah its for 3 users... |
| 00:21.18 | perd | that would be the first option |
| 00:21.24 | perd | spandsp and txfax/rxfax |
| 00:21.40 | kieranmullen2 | perd you for hire? |
| 00:21.42 | perd | http://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax |
| 00:21.43 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 00:21.50 | perd | no, i'm still learning all this stuff myself |
| 00:22.50 | perd | and chan_skinny is currently pissing me off. |
| 00:22.54 | kieranmullen2 | oh :-) I am also looking at repairing my freepbx install |
| 00:22.59 | kieranmullen2 | somethign messed it up |
| 00:23.08 | perd | :( |
| 00:23.10 | perd | backups i hope |
| 00:23.22 | kieranmullen2 | yeah I made a backup in free pbx |
| 00:23.37 | kieranmullen2 | although I dont know how to uninstall the old version and go back |
| 00:24.04 | kieranmullen2 | the time scedule isnt working so I am closed... and I am getting mailbox erros when people try to leave me voicemail |
| 00:24.11 | jaxxan | kieranmullen2: sounds like you might wanna check out http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
| 00:24.55 | kieranmullen2 | I think it may be a username permissions issue |
| 00:25.02 | kieranmullen2 | since I installed in on a cpanel server |
| 00:25.23 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 00:26.05 | jaxxan | Dovid: i dunno how you're setup, but i just create alias's in postfix to send to multiple email addresses |
| 00:27.12 | jaxxan | grandy: just turn on verbosity (cli: set verbose 5) and watch the dialplan in progress |
| 00:27.24 | grandy | jaxxan: ok |
| 00:28.02 | jaxxan | ok so i provisioned 30,000 voicemail boxes and everything looks good |
| 00:28.35 | kieranmullen2 | jax - did you switch from another pbx? |
| 00:28.52 | kieranmullen2 | for a company you work for? |
| 00:29.07 | jaxxan | i ran a meridian pbx about 5 years ago and replaced it with asterisk |
| 00:29.19 | perd | oooh meridian, very snazzy |
| 00:29.22 | jaxxan | i'm replacing a glenayre voicemail box with another asterisk server right now |
| 00:29.39 | kieranmullen2 | Who is the person that does the merian voicemail annoucment? |
| 00:29.46 | jaxxan | i have 18,000 current subscribers with potential for 30,000 |
| 00:29.52 | perd | damn jaxxan, nice |
| 00:30.01 | perd | do you provide only voicemail? |
| 00:30.04 | jaxxan | no |
| 00:30.05 | kieranmullen2 | I was thinking that while driving home? You can determine the system type by the default voice used |
| 00:30.10 | jaxxan | i do all kinds of stuff with asterisk |
| 00:30.15 | perd | neat |
| 00:30.28 | jaxxan | every year I do dialin voting for a Eukele Contest |
| 00:30.38 | jaxxan | last year i recieved 10,000 calls in 2 hours |
| 00:30.41 | perd | haha nice |
| 00:30.48 | kieranmullen2 | call center? |
| 00:30.49 | perd | what kind of setup handles that many calls? |
| 00:31.03 | jaxxan | i have multiple PRI's to a DMS100 |
| 00:31.14 | perd | and where do you live that 10000 people want to vote on a eukele player :) |
| 00:31.15 | jaxxan | i have a small 5 person call center |
| 00:31.22 | jaxxan | i live in American Samoa (= |
| 00:31.30 | perd | haha, im on oahu |
| 00:31.31 | jaxxan | small island in the middle of the south pacific ocean |
| 00:31.40 | *** part/#asterisk YoYo (i=YoYo@pigpen.office.psknet.com) |
| 00:31.59 | perd | doesnt it get boring over there man |
| 00:32.09 | kieranmullen2 | Do you use digium hardware ? |
| 00:32.15 | kieranmullen2 | For the cards? |
| 00:32.32 | RoyK | digium hardware sucks |
| 00:32.44 | jaxxan | we sell mobile phones and stuff too, so i have sim cards provisioned on the DMS100 that only dial one number no matter what you dial on the demo phone that goes directly to asterisk and plays back a voice file telling you about the phone you're holding |
| 00:32.57 | jaxxan | i'm using a TE410P |
| 00:33.33 | kieranmullen2 | royk- What other PRI cards work with asterisk? Why do the digium cards "suck" |
| 00:33.49 | jaxxan | and a ummmm... wctdm... what card is that |
| 00:33.56 | jaxxan | the 4port FX0 card |
| 00:34.02 | perd | or 24 port |
| 00:34.21 | perd | fx[o|s] |
| 00:34.26 | jaxxan | i used a T400P for like 2 years after Digium stopped making it haha |
| 00:35.01 | jaxxan | in fact i'm using it for this voicemail server now |
| 00:35.10 | jaxxan | it's out of retirement |
| 00:35.30 | RoyK | kieranmullen2: I've used digium, but they have all these "incompabilites" with dell and ibm and hyperthreading and whatnot. now I just use sangoma |
| 00:35.48 | jaxxan | RoyK: i gotta be honest with you, i've not had any problems with my digium hardware and i've spent thousands on different pieces |
| 00:35.50 | perd | well there's your problem, dont buy crappy dell and ibm hardware :) |
| 00:36.06 | perd | supermicro servers for me! |
| 00:36.06 | jaxxan | RoyK: if you're having problems, you can contact digium support and they'll troubleshoot it for you fast as hell |
| 00:36.11 | kieranmullen2 | dell and ibm make nice labels |
| 00:36.30 | perd | i have a supermicro server with like 7 fans in it that will suck your face off if you get too close |
| 00:36.37 | perd | that's a server. |
| 00:36.38 | jaxxan | heh |
| 00:36.39 | undrdawg | perl = kr5kernel? |
| 00:36.44 | undrdawg | perd |
| 00:36.50 | perd | no |
| 00:36.53 | undrdawg | oh |
| 00:37.03 | undrdawg | someone else on a forum i guess |
| 00:37.05 | RoyK | perd: IBM really makes good hardware |
| 00:37.22 | undrdawg | perd = phone nerd? |
| 00:37.23 | perd | i was just trolling |
| 00:37.26 | RoyK | jaxxan: I tried, they said "try another server" |
| 00:37.27 | jaxxan | i dont care for IBM personally, I do have alot of Dell PowerEdge Servers though. |
| 00:37.34 | RoyK | that's not what I want to hear |
| 00:37.49 | RoyK | we only use IBM. it works |
| 00:37.52 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 00:37.57 | jaxxan | RoyK: eh, you must be running a piece of crap if digium said that to you haha |
| 00:38.09 | RoyK | hehe. right. |
| 00:38.13 | kieranmullen2 | since when do dell and ibm make hardware? they justa ssemble them and provide terrible support |
| 00:38.20 | RoyK | ibm 306 and 345 and 346 and 336 |
| 00:38.25 | RoyK | quite crappy |
| 00:38.28 | RoyK | not |
| 00:38.35 | perd | hahaha |
| 00:38.38 | jaxxan | i think the only way they'd say that is if you were using the wrong voltage PCI Slots |
| 00:38.55 | grandy | jaxxan: i see one thing that might be interesting... it says: Received mini frame before first full voice frame |
| 00:39.15 | RoyK | it's all 3.3V, but dougium hardware has lots of pci bugs |
| 00:39.40 | jaxxan | RoyK: /shrug i dont have any problems with my power edge servers |
| 00:40.05 | RoyK | jaxxan: I have no problems with my ibm servers with sangoma cards either |
| 00:40.17 | jaxxan | gotta go with what works for you (= |
| 00:40.31 | jaxxan | grandy: sorry, what's the problem ? |
| 00:40.54 | jaxxan | hey Qwell, you still around ? |
| 00:41.04 | grandy | jaxxan: trying to do follow me with cell phones and i am trying to figure out if i can check to see if it at least rang once... as opposed to going straight to the cell phone's vm |
| 00:41.19 | jaxxan | grandy: wanna know what i do ? |
| 00:41.28 | grandy | jaxxan: sure |
| 00:42.18 | perd | he baits you then leaves you hanging |
| 00:42.30 | perd | that's what he does. pretty nice guy. |
| 00:42.32 | grandy | :) |
| 00:42.40 | jaxxan | i dial SIP for 16 seconds then dial ZAP for 20 seconds, then send it to voicemail |
| 00:43.05 | jaxxan | it varies, but i use macros for it |
| 00:43.40 | grandy | jaxxan: ahh so you pretty much time it to see how long it needs to dial ? but what if the cell is off and it goes straight to vm? Won't it pick up sooner than 16 secs? |
| 00:43.45 | grandy | or 20? |
| 00:43.53 | RoyK | jaxxan: that's wrong. if you say digium's hardware is good, but only with certain hardware, it means you'r in there pockets. hardware is supposed to work regardless of other hardware, such as sangoma does |
| 00:44.03 | jaxxan | well i tie my cellphone voicemail into my asterisk voicemail |
| 00:44.07 | *** join/#asterisk Skarmeth (n=Skarmeth@201009022039.user.veloxzone.com.br) |
| 00:44.10 | jaxxan | so i have one voicemail box for my mobile and ip phones |
| 00:44.56 | jaxxan | http://www.pastebin.ca/311572 |
| 00:44.58 | grandy | jaxxan: ahh ok... so if you don't mind me asking how do you do that? when you put in an asterisk number for your cell phone's voicemail does it pass caller ID even when the phone is off? |
| 00:45.04 | jaxxan | check out that pastebin |
| 00:45.56 | jaxxan | i should update that... |
| 00:46.04 | jaxxan | i never get to 104-6 |
| 00:46.44 | grandy | jaxxan: ahh... but what happens when the cell phone gets a call on its own? |
| 00:47.03 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:47.24 | jaxxan | if you call my IP PHones i'll see your callerid, and if the call is forwarded to my mobile phone i'll see your callerid there too |
| 00:47.39 | *** join/#asterisk HeppyCat (n=hepcat@cpe-24-164-205-64.jam.res.rr.com) |
| 00:47.44 | jaxxan | it's the same thing, just a different route |
| 00:47.50 | grandy | jaxxan: what if the call is forwarded to your mobile and you don't answer? |
| 00:47.57 | grandy | then the mobile forwards it back to asterisk? |
| 00:47.58 | jaxxan | then it goes to my voicemail |
| 00:48.01 | jaxxan | no |
| 00:48.10 | grandy | jaxxan: to your cell voicemail? |
| 00:48.10 | jaxxan | see i have a different situtation than you do grandy |
| 00:48.14 | grandy | jaxxan: ahh |
| 00:48.17 | HeppyCat | evening |
| 00:48.22 | jaxxan | my cell phone voicemail is directly attached to my asterisk voicemail server |
| 00:48.41 | jaxxan | i have one voicemail box that's tied into my two ip phones and my mobile phone |
| 00:48.59 | grandy | jaxxan: ok... i'm just trying to figure out how cell phones work in that respect... i know mine (t-mobile) has a setting where you can set the vm phone number, and i imagine it probably does caller iD |
| 00:49.00 | jaxxan | i have direct access to a DMS100 Switch so i can do pretty much whatever i want |
| 00:49.14 | grandy | jaxxan: ahh |
| 00:50.32 | HeppyCat | whats a DMS100? |
| 00:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:50.39 | jaxxan | it's a telco switch |
| 00:50.46 | jaxxan | nortel specifically |
| 00:51.09 | HeppyCat | i spent most of my day dealing with an intertel pbx |
| 00:51.11 | HeppyCat | blargh |
| 00:52.16 | jaxxan | internal ? |
| 00:52.30 | *** join/#asterisk luke-jr (n=luke-jr@CPE-24-31-246-32.kc.res.rr.com) |
| 00:52.30 | HeppyCat | yeah |
| 00:52.49 | HeppyCat | 4 'nodes' |
| 00:53.38 | HeppyCat | according to the tech it's a rather old system |
| 00:54.22 | HeppyCat | guess the most advanced thing it does is some hacked on mgcp phones |
| 00:54.58 | jaxxan | replacing it with asterisk? |
| 00:55.20 | HeppyCat | any idea why my pstn provider says i should switch to SIP because IAX 'causes jitter during peak times' 'due to an asterisk bug' |
| 00:55.29 | HeppyCat | jaxxan: im workin towards it |
| 00:55.36 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 00:55.40 | jaxxan | i use SIP exclusively |
| 00:55.58 | jaxxan | i've never even tried IAX |
| 00:56.02 | perd | i wish i didnt have these damn 7902 phones |
| 00:56.12 | *** join/#asterisk karmatronic (n=boumkar@84.77.137.210) |
| 00:56.16 | karmatronic | hi there |
| 00:56.23 | jaxxan | yo |
| 00:56.27 | HeppyCat | evening |
| 00:56.31 | karmatronic | does rtp.conf need any configuration ? |
| 00:56.43 | jaxxan | not usually |
| 00:56.50 | karmatronic | in longer terms,i m running chan_bluetooth |
| 00:56.51 | HeppyCat | ive never touched it |
| 00:57.07 | karmatronic | and although connections are made , i just cant hear anything... |
| 00:57.22 | karmatronic | connection is from sip client to headeset |
| 00:57.43 | karmatronic | or from sip to cell phone usinf a cell in the middle |
| 00:58.00 | flenders | hey, I have a question about IVR. can I paste a 5 line dialplan here? |
| 00:58.07 | jaxxan | chan_bluetooth |
| 00:59.39 | karmatronic | jaxxan: yeah ? |
| 01:00.02 | karmatronic | jaxxan: chan_bluetooth s not maintained,shouldnt use it ? |
| 01:00.27 | jaxxan | so basically, you run that, it access your computers bluetooth hardware and directs calls to the bluetooth device of your choice within the coverage area ? |
| 01:00.49 | jaxxan | flenders: no you can't |
| 01:00.54 | jaxxan | flenders: use pastebin.ca |
| 01:01.07 | karmatronic | jaxxan: actually it reaches gsm cell phones using bluetooth cell phone as gateway |
| 01:01.24 | RoyK | nite |
| 01:01.47 | karmatronic | jaxxan: and you can also redirect incoming cell phones from gsm to the sip clients in your local network |
| 01:01.59 | jaxxan | sounds interesting |
| 01:02.07 | jaxxan | also sounds like a pain in the ass to setup |
| 01:02.19 | karmatronic | jaxxan: really reasy |
| 01:02.24 | flenders | if I use this dialplan (http://pastebin.ca/311592), should the call be dropped (hung up) after priority 5? |
| 01:02.32 | flenders | I thought it would wait for input, no? |
| 01:02.42 | karmatronic | jaxxan: worst part was compiling chan_bluetooth module |
| 01:03.02 | karmatronic | jaxxan: although...its not working at the moment |
| 01:03.25 | jaxxan | flenders: no, you have to specify the timeout |
| 01:03.57 | karmatronic | anyway , i changed rtp ports in sip,conf and just thought the guy that created chan_bluetooth expects standard rtp ports |
| 01:04.04 | jaxxan | flenders: specify the timeout with something like exten => t,1,Goto(s,9) |
| 01:04.16 | karmatronic | so here s my last try ...before going to sleep |
| 01:04.48 | jaxxan | flenders: but yeah it waits for input |
| 01:05.03 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 01:05.18 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 01:05.57 | perd | anyone see what might be wrong here? http://pastebin.ca/311600 |
| 01:06.02 | perd | i cant get skinny to work :/ |
| 01:06.05 | jaxxan | if you want it to hangup after 5, then exten => t,1,Hangup |
| 01:06.31 | flenders | jaxxan: I already have 'exten => t,1,Goto(inbound,s,1)' in there |
| 01:06.50 | flenders | but it drops the call as soon as it reaches priority 5 |
| 01:07.44 | jaxxan | change response to 10 |
| 01:07.48 | jaxxan | and try again |
| 01:08.24 | jaxxan | actually, you dont even need response |
| 01:08.34 | jaxxan | just set the time out |
| 01:08.42 | *** join/#asterisk X-Rob (n=Rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 01:08.43 | jaxxan | for the digit |
| 01:09.18 | flenders | this is what I get on the console: http://pastebin.ca/311608 |
| 01:09.56 | jaxxan | cause your response timeout is set to low |
| 01:10.11 | jaxxan | too low |
| 01:10.17 | jaxxan | change it to 10 and see if that helps |
| 01:10.50 | flenders | same thing |
| 01:10.54 | jaxxan | do you want the call to hang up ? |
| 01:10.54 | flenders | I changed both to 20 |
| 01:11.07 | flenders | no, I don't want it to hang up |
| 01:11.13 | jaxxan | then remove the response line |
| 01:12.08 | *** join/#asterisk unsucht (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 01:12.12 | jaxxan | response works like.... if you dont repond with keypress during the timeframe i'm hanging up the call |
| 01:12.13 | flenders | still drops right after setting the digit timeout |
| 01:13.51 | jaxxan | dunno, i just use exten => s,6,Set(TIMEOUT=3) |
| 01:14.03 | jaxxan | works fine for me |
| 01:14.25 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 01:14.52 | unsucht | hi, i'm trying to excute a dialplan when an extention hangs up, i got the extention to ring but it rings for only a split second and hangs up, can i do something about ths? |
| 01:15.28 | jaxxan | unsucht: so you want to process more dialplan when there's no call established ? |
| 01:15.47 | flenders | jaxxan: I think I had another server running the same way over here, and I didn't even need to set the timeouts |
| 01:16.09 | unsucht | jaxxan: what do u mean? |
| 01:16.22 | unsucht | Jaxxon: yes |
| 01:16.33 | flenders | jaxxan: it would wait for input. but now, it drops the call right after the message is heard, so, no time to enter digits. |
| 01:17.44 | jaxxan | have you tried not specifying digit and response ? just use the set(timeout=X) ?? |
| 01:18.07 | jaxxan | unsucht: good luck with that, you should process everything you need before the call is connectd |
| 01:18.35 | jaxxan | unsucht: as far as i'm aware, you can't process anything unless a call is established |
| 01:19.44 | unsucht | jaxxon: what i meant was the after i get this to work i want to executre more stuff like a macro, but for testing i 'm just dialing up an extention, but it only rings for a split second |
| 01:20.34 | jaxxan | paste your problem dialplan http://www.pastebin.ca |
| 01:21.39 | jaxxan | flenders: i'm not sure man |
| 01:21.48 | jaxxan | i can paste my autoattendant if you wanna compare |
| 01:21.51 | jaxxan | mine works fine |
| 01:21.59 | jaxxan | i'm running 1.2.6 |
| 01:22.07 | flenders | jaxxan: yeah, that'd help |
| 01:22.09 | unsucht | http://www.pastebin.ca/311622 |
| 01:22.12 | flenders | jaxxan: thanks |
| 01:24.32 | jaxxan | flenders: http://pastebin.ca/311627 |
| 01:26.33 | jaxxan | unsucht: that dialplan you pasted makes freaking no sense to me |
| 01:27.50 | unsucht | it's only past of it, there is also an agi script that sets the variables |
| 01:28.18 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 01:28.51 | km- | hello! Anyone here have a unlocked PAP2 and find that even though the unit registers with asterisk, picking up the line presents a fast busy? |
| 01:28.52 | jaxxan | i see the DEADAGI script being loaded and i see you spewing the variable data to your console and i see your if statements, but then you just hangup |
| 01:29.05 | jaxxan | and you make reference to 10 and 15, but wtf is that |
| 01:29.31 | jaxxan | you're never gonna get past 6,Hangup |
| 01:29.39 | jaxxan | cause you've *tada* hungup the call |
| 01:30.04 | jaxxan | rethink lines 4 and 5 |
| 01:30.24 | jaxxan | and use +101 |
| 01:31.37 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 01:32.02 | k-man__ | [TK]D-Fender, is grandstream in the "ok" list of brands? |
| 01:33.35 | unsucht | jaxxon: i'm just using 10 and 15 as reference points, is that a big deal, also if the gotoif things don't return true, i want the call to hang up[ |
| 01:33.37 | riddlebox | duck! |
| 01:33.51 | *** part/#asterisk karmatronic (n=boumkar@84.77.137.210) |
| 01:36.23 | riddlebox | k-man, I have a grandstream phone and it works great |
| 01:38.33 | k-man__ | riddlebox, ok |
| 01:38.35 | k-man__ | thanks |
| 01:38.57 | unsucht | the problem i |
| 01:39.10 | riddlebox | k-man, I have heard lots of people dont like them but I havent had a problem with my GXP-2000 |
| 01:39.21 | unsucht | 'm having is that the remote ext only rings for a split second |
| 01:42.29 | [TK]D-Fender | k-man__ : GrandSuck *shudder* |
| 01:42.49 | DocHolliday | heh |
| 01:43.15 | k-man__ | hehe |
| 01:43.25 | k-man__ | such different responses from riddlebox and [TK]D-Fender |
| 01:43.42 | [TK]D-Fender | k-man__ : Stick with the Linksys series for now... |
| 01:43.46 | k-man__ | yeah |
| 01:43.47 | k-man__ | i will |
| 01:43.56 | riddlebox | [TK]D-Fender, I knew you wouldnt take to long to answer |
| 01:44.05 | k-man__ | i was just wondering as somehow ebay keeps emailing me that someone is selling them |
| 01:44.11 | k-man__ | i must work out how to stop it |
| 01:44.21 | [TK]D-Fender | k-man__ Grandstram has been chronic for flakey firmware, HARDWARE ECHO, and all sorts of "fun" |
| 01:44.30 | *** join/#asterisk ctooley (n=ctooley@jc1-111.moment.net) |
| 01:44.54 | ctooley | isn't there a setting for a sip peer to limit the number of channels it will allow? |
| 01:45.09 | ctooley | (ie don't accept inbound connects anymore) |
| 01:45.09 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) |
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| 01:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:50.54 | k-man__ | yeah, good reason not to get one |
| 01:51.28 | [TK]D-Fender | k-man__ : another of my recent clients got them before I could advise him. The lose registration when put on hold.... |
| 01:51.39 | [TK]D-Fender | k-man__ Wierd shit happens... pain you don't want. |
| 01:51.47 | [TK]D-Fender | k-man__ : ymmv |
| 01:52.38 | k-man__ | don't worry |
| 01:52.44 | k-man__ | im sold on linksys or polycom |
| 01:53.14 | DocHolliday | or cisco >:) |
| 01:53.33 | [TK]D-Fender | k-man__ : while polycom is a better quality and functionality vs Linksys, the latter is at least SOLID. The same cannot be said for Snom or Grandstream |
| 01:53.55 | k-man__ | hehe |
| 01:53.55 | [TK]D-Fender | Cisco is even MORE expensive, and their SIP isn't exactly stable or as functional.... |
| 01:54.27 | DocHolliday | [TK]D-Fender, well yeah.. i wish someone would start a SIP firmware project |
| 01:54.47 | [TK]D-Fender | Aastra is a little less stable than some, but more functional that anything except Polycom. |
| 01:56.52 | *** join/#asterisk hot_wheelz (i=hot_whee@203-214-12-140.dyn.iinet.net.au) |
| 01:57.11 | riddlebox | [TK]D-Fender, do you like the actual Aastra pbx's? |
| 01:57.15 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 01:57.36 | HeppyCat | ive had a gxp for a year or so now |
| 01:57.45 | HeppyCat | wouldnt call it 'solid' |
| 01:57.52 | HeppyCat | gxp2000 |
| 01:58.02 | [TK]D-Fender | riddlebox : Never tried. |
| 01:58.17 | [TK]D-Fender | riddlebox : the 480i is a very god phone though |
| 01:58.22 | [TK]D-Fender | good* |
| 01:58.55 | flenders | jaxxan: if I use 'exten => s,4,WaitExten(15)' then it doesn't drop the call |
| 01:58.56 | riddlebox | [TK]D-Fender, I cant find that phone on voipsupply anymore, the only way I see it is with the cordless phone bundled |
| 02:00.11 | riddlebox | nevermind now I have :) |
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| 02:01.43 | ctooley | call-limit it is |
| 02:01.55 | [TK]D-Fender | flenders : == Auto fallthrough, channel 'SIP/203.161.160.244-081a0008' status is 'UNKNOWN' |
| 02:02.14 | [TK]D-Fender | flenders : thats your problem. You didn't set "autofallthough=no" in [globals] |
| 02:03.13 | *** join/#asterisk Globetrotter (n=eric@205.211.239.11) |
| 02:03.18 | wunderkin | *twitch* |
| 02:03.33 | Globetrotter | Hi Guys,, how do i recompile asterisk bussiness edition? |
| 02:04.10 | [TK]D-Fender | Globetrotter : the same way as normal. Its not magically different |
| 02:04.30 | file | they distribute the source with BE? |
| 02:04.33 | Strom_M | you don't recompile ABE. it's already compiled. |
| 02:04.35 | [TK]D-Fender | Globetrotter And besides which, having bought it that means you got the nifty manual that puts it in BIG PRINT for you too :) |
| 02:05.20 | file | ugh the manual |
| 02:05.48 | Globetrotter | yes, i made a mistake. i ran this command and recomplied :( |
| 02:05.50 | Globetrotter | svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
| 02:06.00 | Globetrotter | obviuoly i am a newbee |
| 02:06.24 | [TK]D-Fender | Globetrotter : And very clearly that should not be ABE you are downloading, otherwise anyone could do it. |
| 02:06.47 | Globetrotter | yep, gotcha |
| 02:06.53 | Globetrotter | thanks |
| 02:06.58 | [TK]D-Fender | Globetrotter : ABE dosn't DO SVN. they only have nice old stable. |
| 02:07.16 | [TK]D-Fender | Globetrotter : So did you PAY for ABE? |
| 02:07.22 | Globetrotter | yes |
| 02:07.25 | file | B is based off of 1.2 |
| 02:07.34 | Globetrotter | but, i was rrying to install the addons |
| 02:07.36 | file | so not quite "nice old" |
| 02:07.39 | Globetrotter | then i f*cked up |
| 02:08.10 | [TK]D-Fender | Globetrotter : Time to actually figure out what you're doing rather than downloading mis-matched bigs. Go use that support you paid for. |
| 02:08.16 | [TK]D-Fender | bits* |
| 02:08.35 | Globetrotter | thnbaks guys |
| 02:09.22 | *** join/#asterisk ManxPower (n=manxpowe@244.sub-70-196-166.myvzw.com) |
| 02:09.24 | [TK]D-Fender | Globetrotter : That IS why you piad for it isn't it? |
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| 02:13.51 | *** part/#asterisk hot_wheelz (i=hot_whee@203-214-12-140.dyn.iinet.net.au) |
| 02:14.53 | Globetrotter | very true :) |
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| 02:34.49 | flenders | [TK]D-Fender: thanks a lot mate! |
| 02:35.45 | flenders | [TK]D-Fender: I had autofallthrough=yes |
| 02:36.39 | niZon | anyone got ztdummy working on a 2.6 kernel without usb uhci? |
| 02:36.50 | *** part/#asterisk kieranmullen2 (n=kieranmu@static-71-245-97-83.ptldor.fios.verizon.net) |
| 02:37.37 | jaxxan | flenders: glad you fixed i |
| 02:37.42 | jaxxan | it |
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| 02:38.44 | jaxxan | omg did you guys see the new iphone ? |
| 02:39.17 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 02:39.47 | HeppyCat | yeah |
| 02:39.59 | HeppyCat | it's $600 subsidized |
| 02:40.33 | HeppyCat | im more interested in motorola's slim, no frills phone |
| 02:40.57 | HeppyCat | but then again, i just got a treo last month |
| 02:41.24 | DrCron | meh, i want one of the htc phones |
| 02:41.25 | jaxxan | i dont care if it was $1k dollars |
| 02:41.33 | jaxxan | i can't wait to get one |
| 02:44.12 | james_ | wow, a model consumer if ever there was one |
| 02:44.53 | james_ | brand/label + old technology + a couple of gimmicks = "i dont care if it was $1k dollars" |
| 02:44.56 | battini | yeah really |
| 02:45.01 | battini | "APPLE!" |
| 02:45.06 | *** part/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 02:45.14 | james_ | i'm not an apple hater |
| 02:45.25 | jaxxan | i'm a mac user so i'm all about apple hardware |
| 02:45.26 | battini | dude neither am I |
| 02:45.27 | james_ | but if you put the iphone next to the nokia n95 |
| 02:45.29 | battini | I have a G4 |
| 02:45.33 | james_ | it's nothing *spectacular* |
| 02:45.34 | battini | but jesus christ |
| 02:45.37 | battini | the ipod is worthless |
| 02:45.41 | [TK]D-Fender | flenders : You're welcome |
| 02:45.42 | jaxxan | i have a macbook pro |
| 02:45.43 | battini | its all name recognition |
| 02:45.56 | jaxxan | it kicks the shit out of my g4 and i liked that one too |
| 02:46.05 | russellb | i'm pissed that it requires a cingular 2-year contract |
| 02:46.07 | battini | I have a dual cpu 1.2 g4 |
| 02:46.12 | battini | that i use illustrator on |
| 02:47.07 | [TK]D-Fender | G4 was a puny processor and Intel saved Apple from being crushed on performance. Apple does make great products, but I LOATH their "our way or the highway" method of managing it. |
| 02:47.23 | [TK]D-Fender | itunes = crap, ipod = crap (software). |
| 02:48.12 | jaxxan | i hated Macs until OS X came around. OS X's BSD is hella better than running debian on a laptop |
| 02:48.32 | battini | the ipod is a way overpriced mp3 player |
| 02:48.42 | km- | Hey guys, any idea how I remove the @10.x.x.x from my callerid on the 7960? Do I need to reset the caller ID in my dialplan? |
| 02:48.44 | battini | My Rio Karma might be ugly, but its 20gb and it cost me 150 bucks |
| 02:48.47 | battini | and that was a few years ago |
| 02:48.49 | km- | or am I missing something like callerid=yes in sip.conf |
| 02:50.13 | jaxxan | i have a sony walkman w850i 4g and it's hella kewl |
| 02:50.27 | jaxxan | but the iphone kicks the hell out of it functionally |
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| 02:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:50.53 | jaxxan | if only i could route it thru asterisk (= |
| 02:52.16 | jaxxan | i wanna be able to dial extensions from my mobile phone |
| 02:52.23 | jaxxan | direct extensions that is |
| 02:53.39 | jaxxan | the ability to be travelling and make calls from my cell phone via IP from any access point would be the shit |
| 02:53.57 | jaxxan | i'm asking to much though |
| 02:54.09 | battini | you could just get a wireless sip phone |
| 02:55.17 | jaxxan | yeah, but i need it to play mp3's, video, sync with my macbook and make love to me too |
| 02:55.56 | battini | sounds like you want a laptop. |
| 02:55.56 | battini | =P |
| 02:56.20 | jaxxan | that's my macbook pro haha |
| 03:02.59 | luke-jr | Anyone up for a game of freeciv? |
| 03:03.00 | *** join/#asterisk SimoAmi (n=SimoAmi@ip67-91-253-242.z253-91-67.customer.algx.net) |
| 03:03.03 | SimoAmi | hi there |
| 03:04.44 | SimoAmi | what's the syntax for permit to allow all hosts |
| 03:07.16 | russellb | well, it's the default, so you don't have to configure it |
| 03:07.56 | SimoAmi | ah, I'll try that now |
| 03:07.58 | SimoAmi | thx |
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| 03:19.09 | flenders | I know this may sound stupid, but so far I've only used SIP, and just got a digium card installed... question is, how do I make calls using channel 1 on my digium card? |
| 03:19.29 | ManxPower | Dial(Zap/1/${EXTEN}) |
| 03:19.39 | flenders | I know that on sip it would be Dial(SIP/${EXTEN}@provider) |
| 03:19.48 | flenders | ManxPower: thanks mate |
| 03:19.51 | ManxPower | zap must be installed before asterisk of asterisk willl not build zap support |
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| 03:20.14 | flenders | ManxPower: it is, all modules loaded, and incoming calls are working |
| 03:20.50 | flenders | ManxPower: I was trying Zap/${EXTEN}/1 |
| 03:21.13 | ManxPower | That would call telephone number "1" using channel ${EXTEN} |
| 03:24.21 | flenders | ManxPower: makes sense now |
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| 03:25.39 | flenders | if I have 4 FXO modules on the card, and channel 1 is in use, channel 2 is in use but 3 and 4 arent, how do I dial using the first available channel? |
| 03:25.57 | flenders | I also want to dial using SIP, if all Zap channels are busy |
| 03:26.24 | Grnd-Wire | Can someone tell me what MeterMaid it is? |
| 03:27.43 | [TK]D-Fender | flenders : ou set each channel into the same group like "group=1", and dialthem like Dial(ZAP/g1/1234567) |
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| 03:28.30 | [TK]D-Fender | flenders : G1 an g1 each choose group 1, 1 in ascending order, the other in descending (of availability. I forget which) |
| 03:28.52 | flenders | in zapata.conf? |
| 03:28.53 | [TK]D-Fender | Grnd-Wire : Rita is indeed lovely :) |
| 03:28.57 | [TK]D-Fender | flenders : Correct |
| 03:29.06 | flenders | [TK]D-Fender: thanks again! |
| 03:29.14 | [TK]D-Fender | flenders ; np |
| 03:30.18 | flenders | [TK]D-Fender: if I add a second TDM400 card, will the channels be Zap/1-1, Zap/1-2, Zap/2-1...? |
| 03:30.18 | Grnd-Wire | TKD: hmm - I was serious.. I'm seeing that MeterMain is some sort of addon patch to the normal parking functionality, or something.. Is there documentation for it on voip-info.org ? |
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| 03:30.29 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 03:31.06 | [TK]D-Fender | flenders : zap/1 through zap/8 |
| 03:31.32 | [TK]D-Fender | flenders : Everything after the dash signifies a unique call on an active channel |
| 03:31.39 | flenders | so the first card on the PCI bus will be 1-4 and the other one 5-8? |
| 03:31.46 | [TK]D-Fender | flenders : Which is the way you'll see it in the CLI as commands are executing |
| 03:32.00 | [TK]D-Fender | flenders : Well.. the first of the 2 to initialize. |
| 03:32.09 | [TK]D-Fender | flenders : I don't know about guranteeing the order. |
| 03:32.28 | flenders | [TK]D-Fender: well, I don't think it makes a difference, does it? |
| 03:32.56 | clyrrad | [TK]D-Fender: per our converstaion earlier you were correct we need a global rate table - seems the way to get the rate is just use the CDR's and calculate it from duration of call and amount charged. Only downside is we are forced to go by carriers CDR's instead of our own - but works for now..... |
| 03:32.58 | flenders | [TK]D-Fender: all calls coming in on any of the channels will be on the same group, and context |
| 03:34.20 | [TK]D-Fender | flenders : In the grand scheme of things SORTA. if you don't have all 8 ports FILLED, that can be a porblem if the order is wrong. |
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| 03:36.14 | flenders | [TK]D-Fender: true |
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| 03:50.56 | TripleFFFF | whats the jumper for on sangoma card ? |
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| 04:01.47 | rue_mohr | if I have a T1 card and a newbridge channel bank, can anyone guide me through making a phone play music? |
| 04:03.50 | *** join/#asterisk osiris (n=osiris@71.205.27.131) |
| 04:05.12 | [TK]D-Fender | rue_mohr : http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 04:06.02 | rue_mohr | sweet |
| 04:06.27 | rue_mohr | what would you say is the simplest thing I can make using that system, would that be it? |
| 04:06.37 | rue_mohr | I would like something almost garunteed to work |
| 04:07.08 | [TK]D-Fender | rue_mohr : "Simplest thin you can make"? HUH? What are you really trying to do? |
| 04:08.34 | rue_mohr | I'm tryign to start, successfully if possable :) |
| 04:08.52 | rue_mohr | I have grand ambitions, but am no fool |
| 04:09.23 | [TK]D-Fender | rue_mohr : Do you get dialtone in the CB and does it WORK? |
| 04:11.43 | rue_mohr | well it would help if I had the centronics connector to hook into it |
| 04:11.51 | rue_mohr | should I get a dialtone with no T1 on it? |
| 04:12.03 | [TK]D-Fender | rue_mohr : So basically you're not equiped to really begin. |
| 04:12.13 | rue_mohr | I have 9/10ths the parts |
| 04:12.16 | [TK]D-Fender | run, no you probably shouldn't |
| 04:12.32 | [TK]D-Fender | rue_mohr : Congratulations. thats like have an engine without GAS :) |
| 04:12.55 | rue_mohr | well I have yet to make supper tonight, so gas might not be a problem |
| 04:14.10 | rue_mohr | newbridge 3624 |
| 04:14.18 | rue_mohr | and about 4 boxes of modules |
| 04:14.28 | rue_mohr | 2 echo cans with no edge connectors |
| 04:14.40 | [TK]D-Fender | rue_mohr : Ok, well ask when you're in a position to find out for sure that everything works. |
| 04:14.51 | rue_mohr | :) |
| 04:15.31 | rue_mohr | the newbridge should be able to act as the clock source, shouldn't it? I understand the clock on the card I have might be flakey |
| 04:15.36 | rue_mohr | what I need is a manual |
| 04:16.02 | rue_mohr | possibly a terminal cord |
| 04:16.16 | rue_mohr | I dont know how smart this thing is |
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| 04:19.49 | rue_mohr | anyone have a copy of "newbridge3624.pdf" as all copies ont eh web seem to have dried up ? |
| 04:21.35 | rue_mohr | digium, I think thats the T1 card I have |
| 04:28.47 | k-man__ | i know it's totaly off topic, but anyone here know how to program an nec Xen system? |
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| 04:36.29 | [TK]D-Fender | bkruse : what do you play? |
| 04:38.03 | bkruse_home | [TK]D-Fender: guitar! |
| 04:38.05 | bkruse_home | and drums and keyboard and bass |
| 04:38.05 | bkruse_home | but bass doesnt count because its just a cheap guitar :P |
| 04:38.20 | [TK]D-Fender | bkruse : got any recordings to share? |
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| 04:51.49 | Mattwj2005 | good evening all :) |
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| 05:11.30 | horsesgofaster | very quite tonight... |
| 05:11.45 | k-man__ | yeah |
| 05:11.47 | file | yup |
| 05:12.31 | horsesgofaster | sooooo, has anyone tried to config sip video? |
| 05:14.16 | [TK]D-Fender | horsesgofaster : Yes, it passes though jsut fine |
| 05:15.00 | horsesgofaster | excellent! I'm working on setting up my server now with beta4... how is the quality and refresh rate? |
| 05:16.24 | [TK]D-Fender | horsesgofaster : Doesn't work so well over smoke-signals, tin-can-and-string, or jungle-drums.... |
| 05:16.57 | file | horsesgofaster: 1.4.0 beta4? |
| 05:16.59 | horsesgofaster | [TK]D-Fender, I've been play with asterisk for a year on and off and decided to go up to NYC asterisk bootcamp |
| 05:17.11 | horsesgofaster | file, yep 1.4.0 beta4 |
| 05:17.23 | [TK]D-Fender | horsesgofaster : All signed up and ready to go? |
| 05:17.29 | file | horsesgofaster: uh, there's a 1.4.0 release - or you could grab 1.4 from SVN |
| 05:17.37 | horsesgofaster | [TK]D-Fender, yep! |
| 05:18.01 | horsesgofaster | file, grabbed it from asterisknow, used the iso. |
| 05:18.19 | [TK]D-Fender | horsesgofaster : well try to do as much reading and testing before you go so you can get the most out of it. |
| 05:18.23 | file | oh asterisknow |
| 05:18.35 | horsesgofaster | file, played with the liveCD, it was pretty cool |
| 05:19.26 | *** join/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net) |
| 05:19.30 | horsesgofaster | [TK]D-Fender, yep, last week and this week almost full time on asterisk config and voip-supply config... also read the 'book'... haven't done that in a long time :) |
| 05:20.31 | tim0123 | is the anyway to use xml with asterisk |
| 05:20.40 | Mattwj2005 | I am going to try a Asterisk build on Gentoo.....right now I am compiling the dependencies |
| 05:21.11 | Mattwj2005 | 1.4 :D |
| 05:21.13 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 05:21.54 | [TK]D-Fender | Thats what Gentoo is... a compiler :) GCC! |
| 05:22.06 | Mattwj2005 | I like it :) |
| 05:22.31 | *** join/#asterisk Mw3 (i=mw3@ip59934bd1.bp-1031.rubicom.hu) |
| 05:22.34 | Mattwj2005 | now if I had a nice gal.....my life would be prefect ;) |
| 05:23.30 | HeppyCat | Mattwj2005: what arch? |
| 05:23.40 | Mattwj2005 | x86 |
| 05:23.44 | [TK]D-Fender | HeppyCat : McD's :D |
| 05:23.57 | HeppyCat | [TK]D-Fender: disgusting |
| 05:24.08 | danp | ergh, i can't seem to get rid of these 500's from my polycom phones |
| 05:24.16 | Mattwj2005 | plain old 32 bit |
| 05:24.18 | danp | i updated to the latest svn |
| 05:24.20 | [TK]D-Fender | Definately a few fries short of a Happy meal! |
| 05:24.41 | Mattwj2005 | it is only a 600 Mhz system.....enough for a hobbyist such as myself |
| 05:24.53 | HeppyCat | im runnin asterisk on debian |
| 05:25.06 | HeppyCat | been too lazy to move it to my gentoo box |
| 05:25.14 | Mattwj2005 | that works pretty good too .... that is how I did my first systems |
| 05:25.47 | Mattwj2005 | nice thing about gentoo is they finally have an installer....it is cheating.....but at least it is easier |
| 05:25.55 | [TK]D-Fender | danp : You won't. They're a fact of life, but don't seem o cause any problems on my side. I think of them more as "warnings". |
| 05:26.11 | HeppyCat | an installer? |
| 05:26.21 | Mattwj2005 | yup |
| 05:26.29 | Mattwj2005 | it is even called "installer" |
| 05:26.35 | flenders | I installed gentoo 2 or 3 months ago and it was a nightmare |
| 05:26.41 | Mattwj2005 | download the livecd and try for yourself :) |
| 05:26.56 | flenders | only reason was cause I couldn't find proper SATA drivers for a dell server on any other distro |
| 05:27.35 | Mattwj2005 | the live cd allows you to use either a text based or a gui installer |
| 05:27.36 | HeppyCat | ive got most of an gentoo installation on this box |
| 05:27.46 | HeppyCat | oh, i bet you mean the gentoo installer |
| 05:27.51 | HeppyCat | not an asterisk installer |
| 05:28.12 | Mattwj2005 | yeah exactly |
| 05:28.18 | HeppyCat | that |
| 05:28.30 | flenders | Mattwj2005: so now you don't need to download all the stage3 and portege files and do it all yourself? |
| 05:28.31 | HeppyCat | i bet that means you used genkernel too |
| 05:28.52 | Mattwj2005 | nope....the installer is a bit buggy but it works fine |
| 05:29.00 | HeppyCat | i highly suggest *not* to use that installer |
| 05:29.03 | Mattwj2005 | for the most part |
| 05:29.04 | HeppyCat | nor genkernel |
| 05:29.29 | flenders | HeppyCat: in summary, not to use gentoo |
| 05:29.48 | HeppyCat | flenders: no no, im sayin if you're going to install gentoo, do it right ;) |
| 05:30.07 | HeppyCat | the installation isnt difficut, it's just long |
| 05:30.14 | flenders | I'm way past the times I used to enjoy installing linux, compiling kernels, etc... |
| 05:30.41 | flenders | it's way too long |
| 05:30.53 | HeppyCat | i hear you. i reserve gentoo for servers |
| 05:30.58 | HeppyCat | ones i dont put x on |
| 05:31.17 | flenders | the servers we got would be used as a firewall cluster, and it works just fine |
| 05:31.36 | Mattwj2005 | that is why I have chosen it for Asterisk....I just want a minimal os install |
| 05:31.38 | HeppyCat | firewall cluster? |
| 05:31.42 | flenders | but it could have an easier minimum install disk, like debian does, for example, and you take it from there |
| 05:31.57 | flenders | cluster of firewalls |
| 05:31.59 | Mattwj2005 | debian is a good choice too |
| 05:32.44 | HeppyCat | only reason im running debian is i was lazy and in a rush at the time |
| 05:32.57 | flenders | I'm always lazy |
| 05:33.00 | flenders | :o) |
| 05:33.11 | HeppyCat | well, i guess technically this box is running debian. it's ubuntu :) |
| 05:33.16 | HeppyCat | ahh the slack |
| 05:33.41 | flenders | do you run * on ubuntu? |
| 05:33.52 | HeppyCat | no, asterisk is running on debian sarge |
| 05:33.58 | HeppyCat | the only thing on that box |
| 05:34.21 | HeppyCat | i tried for some time to get it working on gentoo/sparc |
| 05:34.24 | flenders | I'm alo running sarge on the * box |
| 05:34.53 | flenders | speaking of sparc, even solaris is easier to install than gentoo |
| 05:34.56 | flenders | :o) |
| 05:34.59 | HeppyCat | haha |
| 05:35.18 | HeppyCat | too bad solaris isnt linux |
| 05:35.26 | flenders | true |
| 05:35.52 | flenders | I like it though... sun hardware could be a little bit cheaper to make it worth it |
| 05:36.34 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net) |
| 05:36.48 | [TK]D-Fender | Slackware = all |
| 05:37.25 | [TK]D-Fender | Binary w/o the bloat! |
| 05:37.32 | [TK]D-Fender | "Just works" |
| 05:37.39 | flenders | first linux I installed was a slackware running kernel 2.0.x |
| 05:38.15 | flenders | used slackware for years, but then I got old and lazy, and debian became my preferred distro |
| 05:38.59 | file | [TK]D-Fender: it's you! |
| 05:39.41 | [TK]D-Fender | file : oh noes! |
| 05:40.47 | [TK]D-Fender | Debian : Where the moral high ground gives way to : what the hell are we trying to acheive anyways? |
| 05:41.13 | flenders | :o) |
| 05:41.54 | flenders | I always thought slackware was a great distro, but during sometime, it was a nightmare to make things work |
| 05:42.07 | [TK]D-Fender | file : while the res isn't the best, the size/cost ration is pretty sweet :) http://www.tigerdirect.ca/applications/SearchTools/item-details.asp?EdpNo=2478106&Sku=A180-AT3220A%20CA |
| 05:42.27 | [TK]D-Fender | flenders : its instant success with e or the past 3 years really. |
| 05:42.34 | *** join/#asterisk RahaiL (n=Rahail@209-19-88-240.detroit.mi.D-Conn.net) |
| 05:42.36 | flenders | I had it as a desktop as well... oh god! why did I punish myself for so many years? |
| 05:42.37 | file | [TK]D-Fender: not bad |
| 05:42.52 | RahaiL | any one know who can do branded softphone |
| 05:42.58 | flenders | I got the latest version (11?) on a linux magazine, but haven't installed yet |
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| 05:44.28 | [TK]D-Fender | flenders : As much as I love it, I'm starting to fall into the "support" category of CentOS. its just more maintainable, and comes with more commonly expected packages. |
| 05:44.34 | file | [TK]D-Fender: how goes teh work? |
| 05:44.42 | [TK]D-Fender | flenders : I'm likely to rebuild my server on it. |
| 05:44.51 | [TK]D-Fender | file : Screw work, I'm *HOME* |
| 05:45.00 | flenders | that's the one based on RHEL, isn't it? |
| 05:45.02 | file | [TK]D-Fender: lol |
| 05:45.04 | file | fine |
| 05:45.09 | file | [TK]D-Fender: how goes teh home? it is... homely? |
| 05:45.16 | [TK]D-Fender | file : Guitar is going great though. Evenings of just goofing off with sweeping is really paying off. |
| 05:45.29 | [TK]D-Fender | flenders : 100% identical, minus the logos :) |
| 05:46.08 | [TK]D-Fender | file : Homely? Don' talk about my bitch like that! |
| 05:46.10 | [TK]D-Fender | :O |
| 05:46.26 | flenders | there was a asterisk@home(?) thing that was based on that, is that right? |
| 05:46.51 | file | [TK]D-Fender: pfft, you just want... |
| 05:47.03 | [TK]D-Fender | flenders : yes thats what A@H / Trixbox uses as a base. |
| 05:47.08 | [TK]D-Fender | file : ! ! ! |
| 05:47.14 | file | I thought so! |
| 05:47.46 | [TK]D-Fender | file : I'm mastering the art s of drugs & rock'n'roll so I can sway teh laydeez |
| 05:48.05 | file | [TK]D-Fender: sounds like a plan. |
| 05:49.20 | file | Juggie: speak... or implode |
| 05:49.59 | *** part/#asterisk Mattwj2005 (n=Matt@c-76-17-131-68.hsd1.mn.comcast.net) |
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| 05:52.18 | *** part/#asterisk horsesgofaster (n=dcantera@pool-72-82-176-152.cmdnnj.east.verizon.net) |
| 05:54.01 | pigpen | anyone know how to have asterisk format the voicemail audio file so that a Windows Mobile 5 phone can play it? |
| 05:54.18 | pigpen | Sorry I run a windows phone...the Apple iPhone won't ship until late June. |
| 05:54.19 | pigpen | :) |
| 05:54.39 | pigpen | I have tried the standard wav & wav49 |
| 05:56.53 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net) |
| 05:58.29 | [TK]D-Fender | pigpen : install codecs for it |
| 05:58.42 | pigpen | yeah...I have been looking for that too. |
| 05:59.13 | pigpen | So far I have found Codec Pack All in 1 by free-codecs.com |
| 05:59.40 | pigpen | gsm is fine with me if possible. |
| 06:03.10 | pigpen | Dam..I can't find anything. |
| 06:04.03 | *** join/#asterisk zeeesh (i=aadilism@9-237-154-202.wol.net.pk) |
| 06:04.13 | RahaiL | any one know who can do branded softphone |
| 06:04.43 | pigpen | look at idefisk (I don't know if they do it or not) |
| 06:04.49 | pigpen | or .... dam...I forget. |
| 06:05.01 | HeppyCat | asterisk doesnt dump wav or mp3? |
| 06:05.10 | pigpen | dump? |
| 06:05.33 | pigpen | RahaiL, it is the one that Vonage has branded. |
| 06:05.40 | pigpen | HeppyCat, dump? |
| 06:06.00 | HeppyCat | what is asterisk writing that you can play? |
| 06:06.30 | pigpen | Well, mp3's and from what I can figure out, pcm encoded wav files. |
| 06:06.38 | RahaiL | pgpen any softphone that work on windows 98+ |
| 06:06.55 | pigpen | RahaiL, yeah..check those out. |
| 06:07.16 | pigpen | I may have found a solution...but I need to try. |
| 06:07.32 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 06:07.39 | RahaiL | which one |
| 06:07.48 | pigpen | both. |
| 06:08.15 | RahaiL | cool |
| 06:08.23 | RahaiL | may I pm you |
| 06:08.31 | pigpen | I may be asleep. |
| 06:08.35 | pigpen | but go for it. |
| 06:08.38 | RahaiL | :) |
| 06:10.02 | *** join/#asterisk bigslam (i=test@c-69-246-89-208.hsd1.mi.comcast.net) |
| 06:13.45 | pigpen | "Unrecognized file type" |
| 06:13.47 | pigpen | dam. |
| 06:13.56 | pigpen | Oh well...maybe in the morning. |
| 06:15.06 | RahaiL | :( |
| 06:20.41 | *** join/#asterisk Deciphan (n=Deciphan@c-24-7-130-249.hsd1.ca.comcast.net) |
| 06:22.16 | Deciphan | Anyone know why I'd be getting tens of thousands of log files generated... messages.1891, messages.1892... event_log.5939, event_log.5940... queue_log.9998, queue_log.9999... etc? They seem to be generated very quickly |
| 06:25.42 | CunningPike | Deciphan: Check your logrotate configuration |
| 06:27.04 | Deciphan | is that in one of the asterisk conf files? |
| 06:27.54 | CunningPike | Deciphan: No - it's a system-wide setting |
| 06:28.00 | CunningPike | Deciphan: man logrotate |
| 06:28.53 | Deciphan | i don't have an entry for that... doesn't look like i have that.. this is Gentoo if that matters |
| 06:29.12 | clyrrad | Deciphan: what distro are you using? |
| 06:29.47 | Deciphan | i think it was originally the 2005 but the kernel's been updated to the latest version |
| 06:30.13 | clyrrad | no - I mean what Distro of Linux are you using? |
| 06:30.19 | Deciphan | Gentoo |
| 06:30.52 | clyrrad | hrm... I just checked what CunningPike suggested it works on CentOS / RedHat and Debian based installs of Linux |
| 06:31.23 | Deciphan | I just looked on a different Gentoo box that I've got running Asterisk and it doesn't seem to have logrotate either... but it doesn't have this problem.. i just have one log file for each function |
| 06:31.54 | clyrrad | are you looking in /var/logs ? |
| 06:32.20 | Deciphan | yah, /var/log/asterisk |
| 06:32.34 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 06:32.38 | Deciphan | I've run Asterisk on several Gentoo servers and never seen this before |
| 06:32.48 | clyrrad | and you have 10's of thousands of log files in there??? |
| 06:33.50 | Deciphan | yah.. so many that I can't even rm * .. i get an error and I have to ls | xargs rm to clear it out |
| 06:34.25 | Deciphan | it's not building up over time either... it happens overnight |
| 06:34.41 | clyrrad | that seems very strange to me - not only have I never seen that - I have never even heard of that |
| 06:34.46 | clyrrad | im not even sure what to suggest you |
| 06:35.16 | FaithX | Deciphan: What log files and what is in them ... fix the problem... |
| 06:35.56 | FaithX | Deciphan: What are they telling you? |
| 06:36.17 | Deciphan | Most of them actually appear to be empty... However.. the message.#### logs all seem to say "WARNING[30066] format_wav.c: Unable to find our position" |
| 06:37.03 | FaithX | you could turn syslog off I guess till you fix the problem... if it is crashing your box. |
| 06:37.05 | *** join/#asterisk shinux__ (n=shinux@196.220.25.29) |
| 06:37.26 | FaithX | 39deg C today... |
| 06:37.59 | FaithX | quite mild :) |
| 06:39.39 | clyrrad | FaithX: yea but wonder what causes him to get that many log files |
| 06:39.42 | clyrrad | that seems very strange |
| 06:39.44 | Deciphan | The queue_log.#### files all show something like "1168404289|NONE|NONE|NONE|CONFIGRELOAD|" |
| 06:39.45 | Deciphan | hehe |
| 06:39.47 | Qwell | Deciphan: You're recording a wav > 2GB.. don't do that :) |
| 06:40.00 | *** join/#asterisk at561 (i=wintermu@69.172.58.252) |
| 06:40.18 | at561 | is there still an asterisk cvs server somewhere? |
| 06:40.21 | Qwell | svn |
| 06:40.28 | Deciphan | Qwell, that's interesting that you mention that... because when I was upgrading to 1.2.14 and backing up the voicemail.. I noticed strange folders which HUGE wav files in some of the peoples voicemail boxes... |
| 06:40.29 | Qwell | svn.digium.com/svn/ |
| 06:40.41 | Qwell | Deciphan: That's what's causing that |
| 06:41.07 | Qwell | which means, you don't have proper hangup detection, and no max silence option |
| 06:41.08 | Deciphan | More importantly, what's causing asterisk to generate these multi-gigabyte wav files? |
| 06:41.12 | Deciphan | ah |
| 06:41.27 | Qwell | OR |
| 06:41.37 | clyrrad | Deciphan: are you sure they are not wave files that were recordings of callers that came in on Queues? |
| 06:41.43 | Deciphan | We were having a hangup detection problem with this box too... very intermittent tho |
| 06:41.46 | Qwell | somebody is doing something mischievous |
| 06:41.48 | clyrrad | a long converstaion on a Queue could cause a huge flie like that too |
| 06:41.57 | Deciphan | No, this box is not setup to record any calls... these were inside people voicemail folders |
| 06:41.58 | Qwell | long == several days |
| 06:42.03 | Qwell | like...2 |
| 06:42.19 | Deciphan | There aren't even any queues setup.. they don't use them... just rings straight to extensions |
| 06:42.24 | rue_mohr | :( I cant find the diagram on the web for the proper wiring of the back of a bix strip |
| 06:42.26 | clyrrad | I see.... |
| 06:42.47 | rue_mohr | well not wiring as much as how to bundle the wires properly |
| 06:42.49 | *** join/#asterisk dorphalsig (n=dorphals@pcsp163-73.supercabletv.net.co) |
| 06:42.50 | *** join/#asterisk conver2 (n=marc3234@206-248-132-60.dsl.teksavvy.com) |
| 06:43.04 | Qwell | Deciphan: in the short term, set voicemail to have a max silence of several seconds (something reasonable - maybe 6-10 seconds) |
| 06:43.14 | Qwell | in the long term, figure out what's causing the hangup detection to fail |
| 06:43.18 | dorphalsig | Hey |
| 06:43.20 | dorphalsig | Hi |
| 06:43.21 | Qwell | actually, both would be ideal |
| 06:43.30 | dorphalsig | Gnight =) |
| 06:43.49 | clyrrad | Qwell: you think something is hititng his voicemail and keep recording or something similar? |
| 06:43.50 | dorphalsig | I have a small q ... umm... is sip_mysql_friends |
| 06:43.58 | rue_mohr | is there another name for bix strips? |
| 06:43.58 | Qwell | clyrrad: I know so :) |
| 06:44.02 | dorphalsig | on in * 1.2 or is it a 1.4 feature? |
| 06:44.26 | Deciphan | yah.. i was trying to figure out the hangup detection.. it all looks good... but maybe the phone company is doing something weird... come to think of it, this is a voice/data circuit that's being split with some telco box with pots lines going into the asterisk box... maybe that splitter box is acting strange |
| 06:44.30 | clyrrad | Qwell: can that be done on purporse as an attack on the system, or its a mistake in the dial plan that does not stop the recording? |
| 06:44.52 | Qwell | Deciphan: very likely |
| 06:44.55 | conver2 | is there a way to specify the ringtime for each member when using rrmemory strategy ? |
| 06:45.19 | Qwell | conver2: I think ring time is per queue, however, if you use a Local channel, you can use Dial with a timeout option |
| 06:45.57 | conver2 | ringtime per queue meaning total ringtime for all members? |
| 06:45.57 | Qwell | so, instead of adding the users device to the queue, add their exten or something |
| 06:46.02 | clyrrad | conver2: you are looking for the timeout parameter |
| 06:46.04 | rue_mohr | bix is nortel... |
| 06:46.04 | Qwell | for each member |
| 06:46.17 | Qwell | ie; it's the same for all members of a given queue |
| 06:46.36 | Qwell | So, if you add like Local/bob@queues |
| 06:46.38 | Qwell | you could have |
| 06:46.40 | Qwell | [queues] |
| 06:46.55 | Qwell | exten => bob,1,Dial(SIP/bob||20) |
| 06:46.57 | Qwell | and |
| 06:47.00 | rue_mohr | We have 50,100,250,300 pairs metal frame with bracket, god, I hope I ever have to punch 50100250300 conductors... |
| 06:47.04 | clyrrad | Qwell: so could what I ask above hold true that someone can do it as an attack? Or its just a mistake in the dialplan? |
| 06:47.04 | Qwell | Local/steve@queues |
| 06:47.11 | Qwell | exten => bob,1,Dial(SIP/steve||8) |
| 06:47.22 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:47.23 | rue_mohr | hmm missing quotes ;) |
| 06:47.29 | Qwell | clyrrad: sure, it's possible |
| 06:47.36 | clyrrad | as an attack? |
| 06:47.38 | Qwell | sure |
| 06:47.49 | clyrrad | interesting... something to watch out for then |
| 06:47.57 | clyrrad | guess thats why you put /var in its on partition |
| 06:48.05 | clyrrad | own* |
| 06:48.10 | Qwell | well, there are other options, like max message length in voicemail |
| 06:48.16 | conver2 | ok, how do i specify the ringtime that applies to eacg member if a queue? |
| 06:48.19 | Qwell | I mean, NOBODY is going to get a 10 minute voicemail |
| 06:48.19 | clyrrad | yea that too |
| 06:48.28 | Qwell | so, set the max length to 600 seconds |
| 06:48.32 | clyrrad | Unless its a wife gone mad hahaha |
| 06:48.45 | Qwell | however, that doesn't stop somebody calling back in every 10 minutes. ;) |
| 06:48.58 | Qwell | that's just something you as an admin need to be aware of |
| 06:49.23 | Qwell | (of course, the way to alleviate that problem, would be to have a max message count...but...yeah) |
| 06:49.30 | Qwell | conver2: I just explained it above |
| 06:49.48 | clyrrad | so you need to set those 2 parameters to protect your server |
| 06:49.58 | dorphalsig | is SIP_MYSQL_FRIENDS on asterisk 1.2 or is it a 1.4 feature? |
| 06:49.58 | clyrrad | max message count and max lenght |
| 06:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 06:50.34 | clyrrad | Qwell: what happens if a mailbox count == Max mail count and someone tries to leave a message in that voicemail box? |
| 06:50.35 | conver2 | only using the queue app. |
| 06:50.39 | Qwell | it fails |
| 06:50.42 | clyrrad | does the system tell them its full? |
| 06:50.45 | Qwell | yeah |
| 06:50.48 | clyrrad | sweet! |
| 06:51.11 | Qwell | obviously there are some drawbacks - like legitimate messages being blocked |
| 06:51.13 | clyrrad | yea thats a good thing.... prolly one of the only ways you can save your Hard Drive |
| 06:51.33 | clyrrad | yea but at the same time if someone lets their mailbox fill up like that they clearly dont care about the messages |
| 06:51.38 | clyrrad | unless they are being spammed |
| 06:52.20 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:52.23 | conver2 | qwell-- so you are saying to implement the queue using a queue context, and not by the queue app? |
| 06:52.34 | Qwell | conver2: no, you would use the queue app too. |
| 06:53.01 | Qwell | conver2: see doc/queues-with-callback-members.txt in the 1.4 source |
| 06:53.22 | Qwell | that gives an example or two of using chan_local with queues. You should be able to see how you could add a timeout in the Dial() calls |
| 06:53.53 | clyrrad | Qwell: do you do coding for Asterisk as well? |
| 06:53.59 | Qwell | yes, full time |
| 06:54.13 | clyrrad | no wonder you know all the in's and out's :) |
| 06:54.38 | clyrrad | I cant wait to upgrade to 1.4 but I need to re-do most of my dial plan's :( |
| 06:54.44 | Strom_M | qwell is lying - he spends his days as a circus monkey groveling for peanuts ;) |
| 06:54.50 | clyrrad | hahahahah |
| 06:55.12 | clyrrad | Strom_M - see I remembered not to call you STORM hehe |
| 06:55.18 | Strom_M | woot |
| 06:55.20 | clyrrad | I know thats a pet peeve of yours |
| 06:55.24 | clyrrad | :) :) :) |
| 06:56.22 | clyrrad | so how you guys find the new variable lenght DTMF on 1.4? Do you guys fine you are able to use more IVR systems? |
| 06:56.33 | clyrrad | I have lots of issues on 1.2 of certain systems not getting the DTMF's |
| 06:56.46 | clyrrad | wonder how much improved you guys find this on 1.4 |
| 06:59.00 | Qwell | yes, vldtmf is MUCH better |
| 06:59.02 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 06:59.12 | clyrrad | so you are able to use more of these systems you finding? |
| 06:59.24 | clyrrad | I get lots of complaints that systems dont get the DTMF's |
| 06:59.29 | Qwell | yeah, there were several systems that we couldn't interop with, that we now can |
| 06:59.40 | clyrrad | we are waiting to upgrade - but need to re-do much of the dial plan syntax |
| 06:59.51 | clyrrad | Qwell: thats great news |
| 07:00.43 | clyrrad | Qwell: since your a DEV this is a good question to ask you |
| 07:01.03 | clyrrad | how come Asterisk dial plan is written like scripting - instead of a coding langauge like C or PHP or Java? |
| 07:01.10 | Qwell | see ael |
| 07:01.20 | Qwell | it's very c-like |
| 07:01.28 | clyrrad | im not famaliar with ael |
| 07:01.33 | Qwell | google it :) |
| 07:01.36 | Qwell | ~ael2 |
| 07:01.43 | clyrrad | but I can code in C and PHP :) |
| 07:01.43 | Qwell | ~ael |
| 07:01.44 | jbot | i guess ael is Asterisk Extension Language - a dialplan language with 'c like' syntax? |
| 07:01.44 | clyrrad | so thats exciting |
| 07:01.44 | Qwell | stupid bot |
| 07:01.45 | clyrrad | hahahaha |
| 07:01.48 | Qwell | there we go |
| 07:01.58 | clyrrad | oh cool |
| 07:02.03 | clyrrad | is it ment to replace AGI? |
| 07:02.29 | Qwell | no, it's just like extensions.conf, but a different format |
| 07:02.36 | Qwell | way cooler |
| 07:02.48 | Qwell | and very massively overhauled in 1.4 - it works great |
| 07:02.55 | Qwell | syntax checking and the whole 9 |
| 07:03.03 | clyrrad | oh so I could use extensions.ael instead of extensions.conf and code like I do in C / Java / PHP sytle? |
| 07:03.08 | Qwell | (props to codefreeze) |
| 07:03.17 | Qwell | clyrrad: basically |
| 07:03.24 | clyrrad | oh thats orgasmic!!!!!!! |
| 07:03.26 | Qwell | check out the wiki entry for ael/ael2 |
| 07:03.30 | clyrrad | im there now |
| 07:03.36 | Qwell | and the sample extensions.ael |
| 07:03.48 | clyrrad | can you create variables loops and functions and all that fancy jazz??? |
| 07:03.53 | Qwell | yep |
| 07:03.55 | Qwell | well |
| 07:04.00 | Qwell | macros.. |
| 07:04.12 | Rhizome | it even has goto :P |
| 07:04.26 | Qwell | loops are easy.. just a While/EndWhile loop (which uses the asterisk applications) |
| 07:04.30 | clyrrad | Rhizone: yea but thats a bad bad word in coding |
| 07:04.41 | clyrrad | called spagetti programming |
| 07:04.42 | Rhizome | hehe, yea I jsut thought it was nostalgic :P |
| 07:04.44 | Qwell | nah, goto is good for some things |
| 07:04.57 | clyrrad | yes but for the most part its to be avoided like the plague |
| 07:05.04 | clyrrad | like Global Variables |
| 07:05.20 | Qwell | globals are also good for some things |
| 07:05.36 | clyrrad | yes they have a very specific purpose other than that they are to be avoided :) |
| 07:05.37 | Qwell | hell, your kernel wouldn't be able to count without them ;) |
| 07:05.44 | clyrrad | this is true |
| 07:05.48 | Qwell | jiffies :p |
| 07:05.55 | clyrrad | but we know as a general rule not to use them right??? |
| 07:06.02 | clyrrad | when at all possible.... |
| 07:06.12 | Qwell | sure, always use the smallest scope possible |
| 07:06.19 | clyrrad | yep yep |
| 07:06.21 | Qwell | but, occasionally, global is the smallest scope |
| 07:06.30 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 07:06.40 | clyrrad | I like to declare my itterator varaibles inside the scope of the loop (ie.. in the for syntax) so they dont exist out side the loop |
| 07:06.50 | clyrrad | I see some devs declare and initalize them above the loop |
| 07:06.54 | clyrrad | then never use them outside the loop |
| 07:06.58 | clyrrad | makes no sense to me |
| 07:07.37 | Qwell | anyhow, time for bed |
| 07:07.50 | clyrrad | Ok have a good one - nice chatting with ya |
| 07:08.16 | *** join/#asterisk duckz (n=duckz@141.85.3.18) |
| 07:12.19 | *** join/#asterisk apardo (n=apardo@87.217.144.225) |
| 07:14.11 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 07:14.17 | *** join/#asterisk stephane_ (n=stephane@zafnib.cabale.net) |
| 07:25.42 | *** join/#asterisk angom_h (n=Angel@red-corp-200.79.148.93.telnor.net) |
| 07:25.56 | *** join/#asterisk unsucht (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 07:26.26 | unsucht | is there something available that will alert a user if an extention dialed earlier becomes available |
| 07:27.37 | zeeesh | how to verify that ... mysql installed or not ... at any server . what is the command and where to check in redhat /?? |
| 07:29.14 | *** join/#asterisk Mavvie (n=edwin@ppp12-76.lns2.syd7.internode.on.net) |
| 07:30.29 | tzafrir | zeeesh, you can use rpm to check the the packages |
| 07:31.01 | tzafrir | mysql server? or the command-line client? |
| 07:36.29 | *** join/#asterisk xnon (n=xnon@200.82.223.85) |
| 07:38.29 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 07:39.36 | *** join/#asterisk foxxtrot (n=craig@67.185.0.172) |
| 07:43.47 | CunningPike | zeeesh: mysql ? |
| 07:45.47 | dongs | does asterisk support session timers yet |
| 07:45.49 | dongs | in a usable way? |
| 07:49.33 | dongs | hello? |
| 07:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 07:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 07:50.48 | dongs | lol @ seeing mailing list posts about this shit back in 2003 |
| 07:50.50 | dongs | with NOTHING done. |
| 07:50.55 | dongs | thats just amazing |
| 07:52.24 | *** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
| 07:53.12 | *** join/#asterisk xlnc2002 (n=root@203-97-110-166.cable.telstraclear.net) |
| 07:53.32 | *** join/#asterisk intralanman (n=lanman@pool-71-253-204-107.nrflva.east.verizon.net) |
| 07:56.01 | xlnc2002 | hi there, i'm having trouble setting up an extension, can anyone help pls? .. the AsterikInfo (status) says the extension's host is "UNSPECIFIED". Where does one specify the host of a (SIP) extension? I would have throught it would just default to the PBX server IP#... thanks |
| 07:56.41 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 07:56.47 | xlnc2002 | i missed out - this is setting up extension on the PBX server |
| 07:56.58 | angryuser | good morning;) |
| 07:57.31 | xlnc2002 | hiya, angryuser |
| 07:57.54 | angryuser | i am searching for a voip phone compatible asterisk, with line status of other accounts |
| 07:58.36 | WAudette | I am attempting to get Overhead Paging running while useing * 1.2.13. It looks like my sound card is configured. I have added exten => *52,1,Dial(console/dsp) and exten => *52,2,Hangup() extensions_custom.conf. Is there a way to pump say an mp3 to see if I can hear it. I am running CentOS 4.4. |
| 08:00.06 | xlnc2002 | angryuser, isn't this something the Flash OPerator Panel can do? it monitors line status.. |
| 08:01.43 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 08:01.59 | angryuser | xlnc2002: yes fop can do this, but it is difficult to change habits of people;) |
| 08:02.02 | Strom_M | angryuser, polycom ip430/501/601 |
| 08:02.40 | xlnc2002 | <PROTECTED> |
| 08:03.25 | Strom_M | xlnc2002, explain your question again |
| 08:03.34 | xlnc2002 | so you want a harware phone device...Snom has individual LEDs model 320 configurable for varous "lines" |
| 08:03.48 | xlnc2002 | besides haveing some excellent Codec support. |
| 08:04.37 | xlnc2002 | Strom, thanks.. I am setting up (Trixbox 2.0RC) an extension on the PBX server. A simple SIP extension. |
| 08:04.39 | *** join/#asterisk [hC] (n=hardcore@S0106000fb51cc225.vf.shawcable.net) |
| 08:04.48 | angryuser | Strom_M: thanks |
| 08:04.56 | Strom_M | oh christ |
| 08:05.00 | Strom_M | can't you read the topic? |
| 08:05.09 | Strom_M | ~trixbox |
| 08:05.19 | jbot | extra, extra, read all about it, trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 08:05.20 | xlnc2002 | however, asterixinfo reports the extension's host is Unregistered |
| 08:05.48 | xlnc2002 | ok , sure thing. .. :) |
| 08:05.49 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 08:06.44 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 08:07.21 | xlnc2002 | okay folks, sorry for the intrusion.... bye for now..cheers |
| 08:09.01 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 08:09.48 | angryuser | Strom_M:how polycom ip430/501/601 manage to monitor lines?(asterisk managed inplemented?) |
| 08:10.04 | angryuser | *manager |
| 08:10.07 | Strom_M | angryuser, sip present support |
| 08:10.09 | Strom_M | er |
| 08:10.10 | Strom_M | presence |
| 08:10.13 | Strom_M | no AMI stuff |
| 08:10.19 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:11.15 | angryuser | Strom_M: so will i have a status, like sip/205 busy, Sip/206 free? |
| 08:12.15 | Strom_M | i believe you'll be able to see the status on the telephone set |
| 08:15.32 | *** join/#asterisk foRza (n=forza@firewall.hikt.no) |
| 08:15.57 | angryuser | Strom_M: i need status of sip client to avoid blind transfers |
| 08:16.33 | Strom_M | what do you mean? |
| 08:16.33 | brookshire | angryuser: and do what with them? |
| 08:16.45 | brookshire | Strom_M: ! |
| 08:16.45 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:16.59 | Strom_M | brookshire! |
| 08:17.10 | Strom_M | what's new? |
| 08:17.11 | angryuser | Strom_M: nothing more, status of sip clients on buttons;) |
| 08:17.22 | Strom_M | angryuser, you can do that with a polycom set |
| 08:17.35 | angryuser | Strom_M: thank you for help |
| 08:18.26 | brookshire | Strom_M: not much.. should be in bed |
| 08:19.14 | Strom_M | are you at the office? |
| 08:19.17 | brookshire | no |
| 08:19.20 | brookshire | at my house |
| 08:19.23 | Strom_M | ah |
| 08:19.30 | Strom_M | i'm in bed ;) |
| 08:19.36 | brookshire | i'm close! |
| 08:19.48 | brookshire | it's just around the corner begging for me |
| 08:20.04 | Strom_M | I spent all of today battling a crazy-ass fever |
| 08:20.23 | clyrrad | Strom_M: join the club :s |
| 08:20.23 | brookshire | everyone is sick |
| 08:20.27 | brookshire | i'm not! |
| 08:20.29 | brookshire | muahahaha |
| 08:20.38 | clyrrad | not yet :) |
| 08:20.42 | brookshire | Vitamin C |
| 08:21.08 | brookshire | i haven't been sick in a long long time |
| 08:21.10 | Strom_M | this is the first time I can remember being sick in at least a year |
| 08:21.11 | brookshire | i changed my diet |
| 08:21.13 | Strom_M | maybe two |
| 08:21.27 | brookshire | and i haven't been sick since |
| 08:21.30 | Strom_M | cool |
| 08:33.54 | *** join/#asterisk drone1 (n=kova@tech.quentris.be) |
| 08:34.17 | drone1 | Hi u all |
| 08:34.41 | drone1 | could any one help me with gtalk connection? |
| 08:35.16 | drone1 | I got my asterisk connected to gtalk .. but when I make a call there's no audio |
| 08:37.14 | *** part/#asterisk foRza (n=forza@firewall.hikt.no) |
| 08:37.47 | brookshire | :( |
| 08:38.11 | brookshire | wish i could help you, i've never played around with gtalk |
| 08:38.21 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 08:38.28 | brookshire | so are you sure it's on the gtalk end? |
| 08:38.49 | drone1 | how do you mean? |
| 08:39.14 | drone1 | I've made a test call with SIP ... works fine |
| 08:39.15 | brookshire | how are you connected to asterisk on the other end? |
| 08:39.34 | brookshire | okay.. |
| 08:44.52 | drone1 | there's way too little doc on this subject |
| 08:47.38 | *** join/#asterisk sasch (n=sasch@82.107.30.102) |
| 08:48.01 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:48.48 | sasch | fro pstn line what is the best codec ??? |
| 08:49.11 | sasch | for remove eco .... because i have a tdm400p and with telecom line I have a lot of echo |
| 08:49.40 | sasch | this is correct ?? |
| 08:49.42 | sasch | disallow=all |
| 08:49.42 | sasch | allow=ulaw |
| 08:49.42 | sasch | allow=alaw |
| 08:49.50 | sasch | in my sip.conf |
| 08:50.27 | brookshire | sasch: that will not help your echo problems |
| 08:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:51.11 | brookshire | http://kb.digium.com/entry/1/1/ |
| 08:55.19 | sasch | in asterisk 1.2 at start of call i have a echo but after echo go out |
| 08:55.29 | sasch | now with asterisk 1.4 echo stay all dial |
| 08:56.09 | sasch | <brookshire> thanks for link |
| 08:57.58 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 09:02.43 | *** join/#asterisk Dibbler_Lap (n=Dibbler@212.248.196.179) |
| 09:04.18 | *** join/#asterisk phj- (i=phj@111-168-233-83.relakks.com) |
| 09:04.19 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 09:06.24 | phj- | hi, i was wondering if its possible to terminate a hanging call from the asterisk console? |
| 09:06.27 | zoa | http://news.asteriskguru.com/1/3295/2007/1/9/Slot_car_racing_with_VoIP |
| 09:06.29 | zoa | this is funny |
| 09:06.38 | *** join/#asterisk jaike (n=jaike@125.5.144.90) |
| 09:08.40 | perd | hah |
| 09:08.46 | perd | phone nerds gone wild |
| 09:10.18 | *** join/#asterisk bdheeman (n=bsd@59.144.243.79) |
| 09:10.31 | perd | that needs more technical details though, zoa |
| 09:10.31 | *** part/#asterisk bdheeman (n=bsd@59.144.243.79) |
| 09:15.11 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:15.27 | zoa | yeah i didnt write that |
| 09:15.30 | zoa | it was posted by someone |
| 09:17.10 | frawd | hello, anyone know how the alarms work in zaptel for analog cards? |
| 09:17.51 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 09:23.28 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 09:24.49 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 09:26.14 | monsted | eep, just added a third PRI to a customer who was whining about dropped calls with their 60 channels... minutes later all 90 are in use - good capacity planning? |
| 09:26.44 | zoa | :) |
| 09:26.45 | Ahrimanes | very |
| 09:28.28 | Ahrimanes | mm "not my problem" |
| 09:29.28 | monsted | they might've been a little bit more ahead of thing when they started adding more branches to the setup :) |
| 09:29.48 | Ahrimanes | monsted, still @tdc? |
| 09:32.24 | monsted | yeah |
| 09:32.45 | Ahrimanes | monsted, ok, involved in wholesale traffic stuff? |
| 09:33.16 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 09:33.18 | monsted | nope, hosted voip |
| 09:33.37 | Ahrimanes | ah |
| 09:33.47 | Ahrimanes | we're competitors then... hehe |
| 09:34.16 | angryuser | is there any tuto how 'hints' (aster BLF) work? |
| 09:34.49 | monsted | Ahrimanes: i do the network side of things for 20 hosted cisco ccm and nortel cs1000 setups for medium and large customers |
| 09:35.14 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 09:35.45 | Ahrimanes | monsted, ok, I do a homebrew webinterface for a bunch of small and medium customers ;) |
| 09:36.48 | *** join/#asterisk jm|work (n=jamie@dilbert.jamiem.com) |
| 09:37.14 | monsted | Ahrimanes: for who? |
| 09:37.27 | monsted | your own company? |
| 09:38.01 | Ahrimanes | monsted, segtel |
| 09:46.28 | zoa | anybody from digium online ? |
| 09:46.56 | BrokenNoze | Anyone used ALERT_INFO with 1.4? |
| 09:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:51.45 | frawd | hi again, i have one small doubt: on a "zap show channel x", what is the "Hookstate" parameter supposed to tell? The state of the line (cable connected or not), or the state of a call (currently in communication or not)? |
| 09:53.52 | sasch | <brookshire> i resolve echo channel |
| 09:54.53 | queuetue | When asterisk can't find a sound file, why doesn't it show an error on the console? |
| 09:55.23 | queuetue | It does show in the logs, though - wish I had bothered to look there a few hours ago. :) |
| 09:56.02 | Ahrimanes | queuetue, what have you set verbose to on the console? |
| 09:57.02 | Gido-E | queuetue check logger.conf |
| 09:57.36 | queuetue | Ahrimanes: A whole bunch of v's - 12 or so, so verbose 12, I guess. |
| 09:58.07 | Ahrimanes | queuetue, ok.. usually when debugging i do "set verbose 99" on the console.. much more info :) |
| 09:58.20 | queuetue | Ahrimanes: Ok, good to know. |
| 09:58.23 | *** join/#asterisk drone1 (n=kova@tech.quentris.be) |
| 09:59.27 | drone1 | could someone explain: during a gtalk communication, 'gtalk show channels' says 'no gtalk channels in use' |
| 10:00.18 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 10:02.13 | *** join/#asterisk kay2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 10:02.20 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) |
| 10:05.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:07.31 | drone1 | did anyone here manage to get gtalk working? |
| 10:07.38 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 10:10.43 | sasch | for transfer a call with my sip phone |
| 10:10.52 | sasch | is key # + trasfer number ??? |
| 10:12.26 | *** join/#asterisk speedwagon (n=ariel_@dsl-20-177.cofs.net) |
| 10:13.13 | Ahrimanes | sasch, if it's enabled in res_features yes |
| 10:14.08 | sasch | atxfer => *2 ; Attended transfer |
| 10:14.24 | sasch | this in the features.conf |
| 10:14.27 | Ahrimanes | look for blind transfer as well |
| 10:14.32 | sasch | ok |
| 10:14.36 | Ahrimanes | usually # or #1 afair |
| 10:14.57 | sasch | blindxfer => # |
| 10:15.10 | sasch | but in sip.conf i edit any line ?? |
| 10:15.15 | Ahrimanes | nope |
| 10:15.33 | Ahrimanes | hitting # on your phone should play a voiceprompt saying transfer |
| 10:15.38 | Ahrimanes | then enter the number |
| 10:16.30 | *** join/#asterisk GiantPickle (n=GiantPic@S0106006008bd147d.gv.shawcable.net) |
| 10:19.05 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:21.03 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 10:21.50 | angryuser | i am using aster 1.4 with external voip provider, but when aster does a dns search like myprovider.sip.com it has 2 ip adresses, aster takes first one, any way to change it so asterisk finds automaticly good ip to register to? |
| 10:22.16 | Ahrimanes | "good" ? |
| 10:22.50 | angryuser | hm, correct ip to register to;) |
| 10:23.17 | angryuser | auto swith between ip's |
| 10:23.33 | Ahrimanes | load balance or just if the first one is down? |
| 10:24.25 | angryuser | dns lookup = 2 ip's, try to register to first one if not success another one |
| 10:25.42 | angryuser | just tired to change ip's manually |
| 10:26.19 | *** join/#asterisk kupsi (n=sipunan@210.213.101.34) |
| 10:27.06 | sasch | i had features.conf |
| 10:27.17 | sasch | but transfer don't run |
| 10:27.20 | kupsi | !ping |
| 10:28.24 | Ahrimanes | angryuser, hm not sure how far the domain support in asterisk is |
| 10:28.42 | Ahrimanes | sasch, is res_features enabled in modules.conf ? |
| 10:29.05 | sasch | i look |
| 10:29.05 | angryuser | il look |
| 10:29.17 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:29.19 | angryuser | ;) |
| 10:29.47 | sasch | http://pastebin.ca/311911 |
| 10:30.43 | Ahrimanes | hm |
| 10:33.54 | kupsi | hello everyone, currently i administrate a local network with large number of clients. I'm looking for a VOIP solution, can asterisk do the job? I'm planning to host my asterisk server on a SunFire x4200 server but it doesn't come with a sound card, is it possible to install asterisk on it? thanks in advance... |
| 10:35.24 | Ahrimanes | kupsi, sure asterisk can do it.. soundcard is only needed for local calling on the server.. |
| 10:35.26 | poller | How are you planning on receiving/making outside calls? |
| 10:35.53 | Ahrimanes | kupsi, "large number og clients" <- how many is that approx? |
| 10:35.56 | kupsi | the server will simply act as a pbx? |
| 10:35.57 | Ahrimanes | mjoeh |
| 10:36.02 | kupsi | Ahrimanes: around 500 |
| 10:36.05 | kupsi | :D |
| 10:36.25 | Ahrimanes | kupsi, should be just fine.. |
| 10:36.35 | kupsi | the server will only route voip calls... is that possible? |
| 10:36.41 | Ahrimanes | sure |
| 10:36.46 | sasch | <Ahrimanes> can help me with transfer |
| 10:36.54 | kupsi | since i don't have a sound card on my server |
| 10:36.58 | kupsi | ok thanks |
| 10:37.05 | Ahrimanes | sasch, try load res_features.so from the console |
| 10:37.26 | kupsi | is it hard to configure asterisk? |
| 10:37.54 | Ahrimanes | kupsi, not really.. you might look at http://www.asterisknow.org/ |
| 10:38.26 | sasch | i had but don't run |
| 10:38.51 | Ahrimanes | sasch, what does it say? |
| 10:39.15 | sasch | i add line load => res_features.so |
| 10:39.19 | sasch | in featerus.conf |
| 10:39.21 | *** join/#asterisk gripner (n=leif@195.178.169.154) |
| 10:39.23 | sasch | and i restart asterisk |
| 10:39.24 | gripner | got some wierd problems i cant seem to get ontop of. I have sip phone connected to my server, the server is behind a fw with port 5060 forwarded to the server. The sip phon registers ok nd i can make calls and i hear the person i speak to but what is said in the sip phone i cant hear. also when calling voicemail i get wrong password error. any ideas ? |
| 10:39.30 | BrokenNoze | Anyone help me with an ALERT_INFO problem? |
| 10:39.41 | sasch | but when i call a number and i digit # (after) don't transfer |
| 10:40.33 | Ahrimanes | sasch, ah maybe your dtmf mode settings on the phone and asterisk are not the same |
| 10:40.55 | kupsi | Ahrimanes: thanks a lot. |
| 10:41.00 | Ahrimanes | gripner, your asterisk is behind nat? |
| 10:42.03 | Ahrimanes | gripner, http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions has a lot of info on this |
| 10:42.42 | gripner | it is behind a nat, and i followed instruktions to forward pots and update the sip_nat.conf |
| 10:43.18 | Ahrimanes | gripner, this is a typical NAT one way audio problem |
| 10:43.39 | sasch | <Ahrimanes> i post my sip.conf |
| 10:44.24 | *** join/#asterisk SimoAmi (n=simoami@user-1087vl2.cable.mindspring.com) |
| 10:44.28 | SimoAmi | hi there |
| 10:44.46 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 10:46.18 | sasch | dtmfmode = rfc2833 |
| 10:46.23 | sasch | in [general] |
| 10:46.33 | SimoAmi | I wonder if someone can remember that asterisk computer processing background sound ? |
| 10:47.46 | Ahrimanes | sasch, you need to check dmtf settings in sip.conf with what the settings is on the sip phone |
| 10:49.59 | sasch | http://pastebin.ca/311921 |
| 10:50.04 | sasch | this is my sip.conf |
| 10:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:53.03 | kupsi | what's the latest stable version of asterisk? is it 1.4 or 1.2? |
| 10:53.58 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 10:53.58 | Nobbie | heya =) |
| 10:53.59 | kupsi | :D |
| 10:54.22 | gripner | humm been reading some, do u HAVE to use the STUN stuff if your sip clients is behind a nat firewall and your asterisk server is behind another nat firewall? |
| 10:54.23 | Nobbie | what different SIP packets/events do asterisk send to a client busy doing a SIP REGISTER, when nat is enabled, as opposed to disabled |
| 10:58.21 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 10:58.44 | zoa | Nobbie: nothing diffferent |
| 10:59.01 | zoa | it just remembers other return addresses |
| 10:59.09 | zoa | and thus will send the same packets to different ports / ips |
| 10:59.16 | *** join/#asterisk ShaneAu (n=ShaneAu@124-168-12-120.dyn.iinet.net.au) |
| 10:59.21 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 10:59.49 | Ahrimanes | gripner, in general, client AND asterisk behind different NATs will be a big headache |
| 11:00.06 | ShaneAu | Hi all! :D |
| 11:00.18 | Ahrimanes | sasch, what's your sip phone setup to? |
| 11:00.57 | ShaneAu | I'm having an annoying problem, this is the error in the log when I attempt to make an internal call |
| 11:00.58 | ShaneAu | WARNING[4916] chan_sip.c: username mismatch, have <501>, digest has <500> |
| 11:01.29 | ShaneAu | This is when I try to make a call from ext 500 -> 501 |
| 11:01.45 | ShaneAu | I just get a engaged tone, and that error in my logs. |
| 11:01.55 | ShaneAu | Any ideas? |
| 11:03.03 | CtRiX | check your client config or your sip.conf |
| 11:04.08 | ShaneAu | Ok thanks, another thing I did not mention though is if I make a call from 501 -> 500 it works fine, both phones are setup the same on the client side |
| 11:04.19 | *** part/#asterisk jaike (n=jaike@125.5.144.90) |
| 11:10.39 | ShaneAu | Everything seems to be fine, :S. I'm really out of ideas, that why I've came here. |
| 11:10.52 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:12.51 | *** join/#asterisk slima (i=slima@syriusz.slnet.uu3.net) |
| 11:12.55 | Makenshi | if only there were ipv6 support :-( |
| 11:12.56 | Ahrimanes | ShaneAu, looks like 500 has authuser set to 501 or something like that |
| 11:13.18 | BrokenNoze | Anyone know how I can Call SipAddHeader from the Manager API??? |
| 11:13.55 | gripner | do i have to open the RTP ports in the firewall protecting the asterisk server AND the firewall protectiong the sip phones? |
| 11:16.29 | ShaneAu | Ahrimanes, In users.conf you mean? |
| 11:16.56 | ShaneAu | For 500, there is no "authuser" defined, or any of the other ext's. |
| 11:18.01 | Ahrimanes | ShaneAu, on the phone |
| 11:18.09 | ShaneAu | Ahr. |
| 11:18.24 | ShaneAu | It's correct |
| 11:18.32 | ShaneAu | It is registered |
| 11:18.43 | ShaneAu | If it was wrong, it would not be able to register, correct? |
| 11:18.45 | Ahrimanes | well it's somehow sending an md5 hash with the wrong username in |
| 11:18.52 | penguinFunk | hey guys, is there a nice easy way to find out which codecs are being negotiated, i.e syslog or asterisk CLI command? |
| 11:19.02 | ShaneAu | Ok |
| 11:19.25 | Ahrimanes | penguinFunk, sip show channels shows what codecs are being used |
| 11:19.52 | penguinFunk | thanks Ahrimanes :) |
| 11:30.54 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 11:33.02 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 11:34.06 | kupsi | seems like Ahrimanes is an asterisk god.... :D |
| 11:41.51 | Ahrimanes | heeh |
| 11:45.17 | BrokenNoze | anyone deal with the asterisk website, I can't register for the forums |
| 11:45.35 | penguinFunk | ok got another one for you... I have setup a bunch of users assigned to a caller group = 1 |
| 11:46.01 | penguinFunk | now how can i bind a sip user to the group so that when i dial this sip user, all phones on the caller group ring |
| 11:46.17 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:47.21 | Bhaal | Anyone know of a free Linux skype<->asterisk module yet? Im surfing around but found nothing yet... Anyone know of anything? |
| 11:48.50 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 11:49.37 | zoa | there is something |
| 11:49.42 | zoa | but it requires X |
| 11:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:50.40 | penguinFunk | nevermind got it sorted now :) |
| 11:50.51 | zoa | http://www.asteriskguru.com/archives/image-vp239187.html |
| 11:50.55 | zoa | its not really free |
| 11:50.59 | zoa | but there is a free demo |
| 11:51.10 | e-ddie | Ahrimanes: hi man. how's life? |
| 11:51.21 | Ahrimanes | hey e-ddie, pretty good, you? |
| 11:51.25 | e-ddie | same here |
| 11:56.50 | Ahrimanes | e-ddie, hows foniris? |
| 11:57.28 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 11:57.59 | e-ddie | Ahrimanes: fine... |
| 11:58.30 | e-ddie | Ahrimanes: still working for the same company you started at when you left? |
| 11:59.48 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 12:03.13 | penguinFunk | can anyone please tell me.. whats syntactically or logically wrong with this: Dial(SIP/101&SIP/102&SIP/103&SIP/104&SIP/105&SIP/106&SIP/107&SIP/109) ? |
| 12:04.39 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 12:04.45 | Ahrimanes | e-ddie, yup, good pay and others to do the vagtordning ;) |
| 12:04.59 | e-ddie | heheh |
| 12:05.13 | e-ddie | sounds nice |
| 12:06.00 | *** join/#asterisk Hello2007 (n=wardeh20@mail.splendor.net) |
| 12:06.07 | Hello2007 | hwllo everybody |
| 12:06.19 | Hello2007 | does asterisk supprt h264? |
| 12:08.29 | Ahrimanes | Hello2007, i believe there's preliminary support for h264 calls |
| 12:08.43 | Ahrimanes | e-ddie, you still the only one there, or? |
| 12:09.08 | penguinFunk | how can I manually register a sip user using the CLI ? |
| 12:09.11 | e-ddie | yeah |
| 12:09.23 | Ahrimanes | penguinFunk, from asterisk cli? |
| 12:09.24 | zapp-branigan | hi what is the diference between iax and iax2 trunked? |
| 12:09.25 | Ahrimanes | e-ddie, ok |
| 12:09.47 | Ahrimanes | zapp-branigan, trunking has less overhead for many channels between 2 servers |
| 12:10.17 | zapp-branigan | only i must place trunk=yes |
| 12:10.17 | penguinFunk | ive created a sip user called '100'. I want it to refer to everyone in the office. but how can i manuall register it? |
| 12:10.39 | zapp-branigan | only i must place trunk=yes in the iax.conf ? |
| 12:10.43 | penguinFunk | yeh Ahrimanes: asterisk CLI |
| 12:11.08 | Ahrimanes | penguinFunk, use queues or just make an extension like exten => 200,1,Dial(SIP/1&SIP/2...) |
| 12:11.18 | Ahrimanes | zapp-branigan, yes, and have a timing source... |
| 12:11.20 | penguinFunk | exten => 100,1,Answer() |
| 12:11.20 | penguinFunk | exten => 100,2,Dial(SIP/101&SIP/102&SIP/103&SIP/104&SIP/105&SIP/106&SIP/107&SIP/109) |
| 12:11.23 | penguinFunk | i have that |
| 12:11.35 | penguinFunk | but it also has a sip entry in sip.conf... shall i remove the sip entry ? |
| 12:11.38 | *** join/#asterisk MCBoY (n=Hiiii@ADSL-138-163.myt.mu) |
| 12:11.40 | zapp-branigan | but the sound is the same? |
| 12:11.48 | Ahrimanes | zapp-branigan, yes |
| 12:11.59 | Ahrimanes | penguinFunk, yes, sip entry is not needed |
| 12:12.16 | zapp-branigan | i want to route 120 calls can i use that ? |
| 12:12.39 | Ahrimanes | zapp-branigan, if you can enable trunking at both ends, yes |
| 12:12.48 | penguinFunk | thank you very much Ahrimanes |
| 12:12.53 | zapp-branigan | thanks for all penguinFunk |
| 12:12.56 | Ahrimanes | penguinFunk, np |
| 12:12.57 | SimoAmi | how to play a sound file in the dialplan and move on to the next command without waiting for a digit |
| 12:13.52 | Ahrimanes | SimoAmi, Playback() |
| 12:14.19 | SimoAmi | ok, it's a 2 minutes background sound |
| 12:14.53 | SimoAmi | it's not supposed to play all 2 minutes, but just until to remote transaction is performed |
| 12:14.57 | zapp-branigan | the anothers g729a in internet work fine ? |
| 12:15.24 | zapp-branigan | http://kvin.lv/pub/Linux/Asterisk/built-for-asterisk-1.2/ |
| 12:15.42 | zapp-branigan | or work better the digium codec |
| 12:16.30 | zapp-branigan | and the another is the intel codel ipp |
| 12:16.31 | Ahrimanes | SimoAmi, hm not sure theres anything standard for that.. you could use music on hold somehow i guess |
| 12:16.56 | SimoAmi | yes, I was thinking about that |
| 12:16.57 | Ahrimanes | zapp-branigan, depends on regulations on the g729a codec patents in your country/region i guess |
| 12:17.14 | zapp-branigan | spain ? |
| 12:17.18 | SimoAmi | you got it, it's some kind of music on hold |
| 12:17.42 | Ahrimanes | zapp-branigan, ok, i dont know what the regulations are there.. i'd buy the license and get the module from digium.. works like a charm for me |
| 12:18.02 | zapp-branigan | ok |
| 12:18.22 | *** join/#asterisk radclifff (n=radcliff@ua-83-227-171-92.cust.bredbandsbolaget.se) |
| 12:18.50 | Ahrimanes | SimoAmi, yeah, so you could start musiconhold, do your transaction and do SetMusicOnHold(none) |
| 12:18.55 | *** join/#asterisk apardo (n=apardo@87.217.144.192) |
| 12:19.00 | *** join/#asterisk atif_ (n=atif@202.92.17.40) |
| 12:19.15 | Ahrimanes | SimoAmi, allthough i'm not entirely sure if MusicOnHold() will return control to the dialplan |
| 12:19.23 | atif_ | hello all, can some one tell me if packetization support is merged in 1.4.0, e.g. allow= g729:10, g729:20 etc |
| 12:20.32 | SimoAmi | I think it will. and SetMusicOnHold(none) is what stops it |
| 12:20.53 | SimoAmi | just reading the wiki right now |
| 12:21.05 | Ahrimanes | SimoAmi, test it out, and please tell me if it works.. interesting way to handle this |
| 12:21.54 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:22.46 | SimoAmi | ok, one moment |
| 12:22.52 | SimoAmi | I'm on it |
| 12:24.17 | Ahrimanes | :) |
| 12:28.06 | Hello2007 | Ahrimanes : do you have any link that talk about asterisk+h264 support? |
| 12:29.15 | Ahrimanes | Hello2007, hm will check, but i have made h264 calls through asterisk around 5-6 months ago with asterisk trunk |
| 12:29.52 | Ahrimanes | Hello2007, http://www.voip-info.org/wiki/view/SIP+Video+Phones search for h264 on that page |
| 12:30.43 | *** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 12:31.32 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 12:31.57 | Hello2007 | does it support it by default,or you need a pacth? |
| 12:33.18 | *** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 12:33.28 | *** join/#asterisk wm4k (n=cfairey@194.164.236.240) |
| 12:33.56 | *** join/#asterisk PeppiX (n=PeppiX@85-18-66-24.ip.fastwebnet.it) |
| 12:34.57 | Ahrimanes | Hello2007, as far as i can tell it should be in 1.4 standard.. try show video codecs on the cli |
| 12:35.34 | Hello2007 | any recommanded video codec for asterisk? |
| 12:35.53 | Ahrimanes | Hello2007, i've used h.263 a lot, works quite well, mostly depending on the phones you use though |
| 12:36.20 | Hello2007 | k, thanks man |
| 12:36.23 | Ahrimanes | np |
| 12:38.43 | *** join/#asterisk santibiotico (n=santi@247.Red-88-15-142.dynamicIP.rima-tde.net) |
| 12:38.44 | santibiotico | hi |
| 12:47.42 | santibiotico | does anybody know how to restore the conversation after trying to transfer a call (attended transfer) |
| 12:49.04 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 12:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:50.54 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:51.32 | Ahrimanes | santibiotico, the other end doesnt pick up or hangs up while you're attending, the original call should come back to you |
| 12:51.32 | santibiotico | yes |
| 12:51.32 | santibiotico | i know |
| 12:51.33 | santibiotico | but in case the other end doesn't hang up |
| 12:51.36 | santibiotico | i.e.: |
| 12:51.40 | santibiotico | i transfer a call |
| 12:52.15 | santibiotico | the person who i transfer the call says he doesn't want to speak to that person |
| 12:52.24 | santibiotico | but doesn't hang up |
| 12:52.33 | Ahrimanes | why wouldnt he hang up? |
| 12:52.36 | creativx | transfer him back then |
| 12:52.41 | creativx | return the problem |
| 12:52.52 | Ahrimanes | the call hasnt been transferred yet.. |
| 12:52.57 | santibiotico | ok |
| 12:53.02 | santibiotico | the example is not correct |
| 12:53.05 | creativx | then you dont need to transfer him :-) |
| 12:53.09 | santibiotico | as my english is not very fluent |
| 12:53.11 | creativx | hehe |
| 12:53.20 | creativx | feel free to rephrase:) |
| 12:53.21 | santibiotico | i didn't want to say "hang up" |
| 12:54.20 | santibiotico | ok.. |
| 12:54.28 | santibiotico | i transfer a call to somebody |
| 12:54.40 | santibiotico | then that person answers the phone |
| 12:54.54 | santibiotico | and he says he doesn't want to speak to that person right now |
| 12:54.55 | santibiotico | ok? |
| 12:54.58 | Ahrimanes | yep |
| 12:55.07 | santibiotico | the normal procedure is that person to hang up the phone |
| 12:55.23 | santibiotico | then i would restore the original call |
| 12:55.25 | santibiotico | ok?? |
| 12:55.28 | Ahrimanes | yes |
| 12:55.31 | santibiotico | well |
| 12:55.33 | santibiotico | imagine now |
| 12:55.43 | santibiotico | that the person who i transfer the call to |
| 12:56.02 | santibiotico | answers the phone, says: i don't want to speak to that person right now |
| 12:56.05 | *** join/#asterisk Mr_Jingle (n=Me@adslgva0491.worldcom.ch) |
| 12:56.13 | santibiotico | but remains with the phone in his hand |
| 12:56.15 | Mr_Jingle | Hello hello |
| 12:56.25 | santibiotico | and doesn't hang up |
| 12:56.39 | santibiotico | i would not restore the original communication |
| 12:56.40 | Ahrimanes | santibiotico, then i'd use call parking or a multiline phone.. and make a seperate call to that person |
| 12:57.44 | santibiotico | is there any procedure to restore the control? |
| 12:57.50 | santibiotico | apart from using call parking |
| 12:58.07 | Mr_Jingle | Question. Does anyone has experience in delphi programming? I'm looking for a free opensource sipstack or activeX control for |
| 12:58.07 | Ahrimanes | hm i'm note sure |
| 12:59.26 | Ahrimanes | Mr_Jingle, http://www.ictrnid.org.uk/index.html?softlib.html ? - a google search for delphi sip stack gives lots.. |
| 12:59.46 | Mr_Jingle | Ok I already tried that. |
| 12:59.55 | Mr_Jingle | lol |
| 13:00.07 | Mr_Jingle | But I'm experiencing some bugs with it |
| 13:00.51 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 13:00.51 | Ahrimanes | ok |
| 13:01.44 | Mr_Jingle | thanks Ahrimanes :) |
| 13:02.57 | Hello2007 | can asterisk do video codec conversion?i found that it work in passthrough mode by default? |
| 13:03.11 | zoa | Mr_Jingle: we use reciprocate with delphi |
| 13:03.20 | Ahrimanes | Hello2007, i don't believe video transcoding is supported |
| 13:03.31 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 13:04.10 | Ahrimanes | Hello2007, "core show translation" on the CLI will show you what formats asterisk can convert between |
| 13:04.21 | Mr_Jingle | zoa: really? |
| 13:06.53 | zoa | yes |
| 13:07.56 | angryuser | it is possible to define a Sip peer's group in sip.conf? |
| 13:08.28 | Ahrimanes | angryuser, "group" ? |
| 13:09.18 | Hello2007 | Ahrimanes: core is not a command??? |
| 13:09.47 | Ahrimanes | Hello2007, what asterisk version are you running? |
| 13:10.23 | Hello2007 | 1.2.7.1 |
| 13:10.55 | Ahrimanes | Hello2007, ah ok |
| 13:11.05 | Ahrimanes | show translation maybe then |
| 13:11.31 | Hello2007 | yes show translation return info about voice codecs only , not video codecs |
| 13:11.39 | angryuser | yes i would like to make a script for outgoing peer selection, if Groupcount >2 then select another peer to dial out |
| 13:12.38 | Ahrimanes | Hello2007, right, so video transcoding is not implemented.. which is probably for the best.. would be rather cpu heavy |
| 13:13.23 | Hello2007 | :-) |
| 13:13.26 | Ahrimanes | angryuser, usually this is done in the dialplan.. dont think you can set variables from sip.conf |
| 13:13.35 | Ahrimanes | Hello2007, but there is work being done for this |
| 13:13.53 | Hello2007 | is it possible to do video transcoding, or some codec need a licence like g729 in voice? |
| 13:14.04 | Ahrimanes | Hello2007, http://www.asterisk.org/node/96 |
| 13:14.17 | Ahrimanes | Hello2007, there's a mailinglist about this |
| 13:14.34 | Ahrimanes | Hello2007, i think it has some answers.. but a lot of patent questions are unanswered |
| 13:14.45 | Hello2007 | k ,thanks |
| 13:14.50 | angryuser | Ahrimanes: ok i will search for more info |
| 13:15.07 | monsted | set port name 1/7 APM TB HN205860 |
| 13:15.10 | monsted | err |
| 13:15.30 | monsted | hov :) |
| 13:15.31 | *** join/#asterisk AstaWerksDotCom (n=doug@63.161.96.170) |
| 13:15.40 | *** join/#asterisk tset (n=tset@S010600e029958636.vs.shawcable.net) |
| 13:15.50 | Ahrimanes | monsted, irc != prompt? |
| 13:16.24 | monsted | CatOS, irssi... same same but different |
| 13:19.04 | Ahrimanes | :) |
| 13:19.36 | Ahrimanes | monsted, except.. once you type enable or the like and enter passwords on irc, so much work changing passwords awaits |
| 13:21.09 | monsted | mmm, trusted ssh keys |
| 13:21.26 | Ahrimanes | ah yes.. they finally got ssh a few years back ;) |
| 13:25.27 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 13:26.36 | e-ddie | i prefer telnet |
| 13:27.00 | e-ddie | makes it a little more interresting |
| 13:27.19 | Ahrimanes | haha, leaves that extra bit of danger in the everyday |
| 13:27.25 | e-ddie | exactly |
| 13:27.38 | e-ddie | actually i think telnet might be more secure in some years |
| 13:27.44 | e-ddie | because noone expects you to use it |
| 13:27.44 | flujan | Hi guys, I am trying to stream a wav file and I am having this message: "format_wav.c:161 check_header: Not in mono 2". Do I need to convert the file to another format? |
| 13:27.59 | e-ddie | as long as you disable ssh and ushc |
| 13:28.00 | e-ddie | such |
| 13:28.13 | *** join/#asterisk ShadowTech (n=jerespet@c-66-176-202-207.hsd1.fl.comcast.net) |
| 13:28.42 | Ahrimanes | flujan, mono 8khz |
| 13:29.37 | flujan | thanks Ahrimanes |
| 13:29.48 | Ahrimanes | e-ddie, not too sure about that |
| 13:29.56 | e-ddie | heheh |
| 13:30.00 | Ahrimanes | e-ddie, port scanners are widely used ;) |
| 13:30.17 | e-ddie | Ahrimanes: port scanner, wtf is that ?? :) |
| 13:30.49 | tzafrir | search engines for the BSD port colelctions |
| 13:30.57 | *** join/#asterisk guilherme_jorge (n=guilherm@200-170-201-134.core01.spo.ifx.net.br) |
| 13:31.02 | Ahrimanes | tzafrir, hehe /usr/ports/security |
| 13:31.22 | Ahrimanes | ah |
| 13:31.35 | Ahrimanes | today |
| 13:31.59 | Ahrimanes | e-ddie, remind me to add a short list of random ip's on arrownet's range to my scan list ;) |
| 13:32.21 | guilherme_jorge | hello all, I'ld like to do a load sharing with 2 asterisk servers. Is there any software to do this, or I have use generic softs to do this, for example, IPVS and heart beat... |
| 13:32.54 | Ahrimanes | guilherme_jorge, hm well heartbeat would be for failover.. simple loadsharing could be done with round-robin dns |
| 13:33.01 | e-ddie | Ahrimanes: heheh |
| 13:33.09 | *** join/#asterisk af_ (n=af@ip-131-134.sn2.eutelia.it) |
| 13:37.34 | Gido-E | _build_general_config: misdn.conf: "method=standard" (section: general) invalid or out of range. Please edit your misdn.conf and then do a "misdn reload". |
| 13:37.50 | Gido-E | is this normal? I have more of these messages. |
| 13:40.42 | *** join/#asterisk mmoreno80 (n=mmoreno8@200.123.180.33) |
| 13:42.28 | mmoreno80 | Hi! |
| 13:42.42 | mmoreno80 | There is a call timeout? |
| 13:42.49 | *** join/#asterisk NirS (n=Nir@84.94.163.104.cable.012.net.il) |
| 13:43.05 | Bhaal | zoa: Yeah, saw that... Im looking for something thats free, its a home thing and I dont have the cash atm... |
| 13:43.08 | Ahrimanes | mmoreno80, explain? |
| 13:43.20 | Bhaal | zoa: re: skype <-> asterisk |
| 13:43.21 | mmoreno80 | Ahrimanes: because TIMEOUT(type) is a channel timeout, an I want a call timeout. |
| 13:43.32 | mmoreno80 | s/an/and |
| 13:43.49 | Ahrimanes | mmoreno80, ah, when you execute dial() you can set an absolute timeout |
| 13:44.01 | mmoreno80 | Ahrimanes: Ok, thanks. |
| 13:46.14 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:48.33 | zoa | Bhaal: there is nothing else |
| 13:49.20 | monsted | Bhaal: i do skype/asterisk with a windows box running skype, a usb/fxs box and an fxo/sip gateway to asterisk |
| 13:50.21 | monsted | (yeah it's ugly, but it works and i even get peoples names via DTMF) |
| 13:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:50.40 | monsted | err, via the caller id thingie that is |
| 13:51.44 | *** join/#asterisk linlin (i=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 13:53.06 | *** join/#asterisk RahaiL (n=Rahail@209-19-88-240.detroit.mi.D-Conn.net) |
| 13:53.35 | RahaiL | got question I have friend who have daiup speed its only 16kb 2kB |
| 13:53.49 | RahaiL | what you guiess recomend to use ilBc or g729 |
| 13:53.50 | tzafrir | Bhaal, don't use skype |
| 13:54.11 | tzafrir | bah, he left |
| 13:55.22 | *** join/#asterisk DerPraktikant (n=Tgu@pD95DE87D.dip.t-dialin.net) |
| 13:55.37 | DerPraktikant | hi |
| 13:55.51 | tzafrir | RahaiL, gsm is also marginally good enough fopr that. or speex. And if you're really desparate: adpcm |
| 13:55.56 | *** join/#asterisk af_ (n=af@ip-131-134.sn2.eutelia.it) |
| 13:55.58 | DerPraktikant | does anyone of u use t-online with asterisk? |
| 13:56.27 | DerPraktikant | i got my server running and i can get incoming calls, but i cant call out |
| 13:57.04 | DerPraktikant | the error is: Forbidden - wrong password on authentication for INVITE to '"032226260403" <sip:92@192.168.0.141>;tag=as4dfd2cf8' |
| 13:57.11 | RahaiL | tzafrir |
| 13:57.31 | *** join/#asterisk gripner (n=leif@195.178.169.154) |
| 13:57.43 | RahaiL | so you are saying 15kpbs gsm is good enough |
| 13:57.49 | atif_ | can some one please tell me if 1.2.14 supports variable packetization .... e.g. g729:10, g729:20 etc |
| 13:57.55 | gripner | can you set delay on voicemail answering per etension? |
| 13:57.59 | DerPraktikant | ah sorry this is the right error: chan_sip.c:9856 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"032226260403" <sip:032226260403@tel.t-online.de>;tag=as5f74f66b' |
| 13:58.42 | DerPraktikant | does anybody knows what i am doing wrong or got experience with such things |
| 13:58.45 | tzafrir | gsm is 13kbps, right? |
| 13:58.56 | RahaiL | i am not sure |
| 13:59.14 | tzafrir | anyway, indeed too marginal |
| 13:59.17 | DerPraktikant | 12,6 k |
| 14:01.21 | RahaiL | tzfrir which one got better sound quality |
| 14:01.25 | RahaiL | gsm or ilbc |
| 14:01.26 | DerPraktikant | the asterisk does the registration but then it gives in error when it wants to dial?! |
| 14:01.28 | *** join/#asterisk Skarmeth (n=Skarmeth@201009061013.user.veloxzone.com.br) |
| 14:02.32 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 14:04.30 | mmoreno80 | How I especify more than one argument to agi EXEC? |
| 14:04.42 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:05.01 | mmoreno80 | For example: dial(SIP/foo,100). |
| 14:05.28 | DerPraktikant | u want to dial 2 people or more at same time?` |
| 14:05.46 | mmoreno80 | DerPraktikant: No. I want especify 2 arguments to dial. |
| 14:06.04 | jeremy_g | mmoreno80:try that && |
| 14:06.18 | jeremy_g | thats a stab in the dark though |
| 14:06.18 | AstaWerksDotCom | quit |
| 14:06.26 | mmoreno80 | jeebusroxors: Ok. |
| 14:07.18 | DerPraktikant | Dial,SIP/91&SIP/92|30|r for example |
| 14:07.19 | *** join/#asterisk zirman (i=zirman@ip194.207.107.216.seg.net) |
| 14:08.38 | mmoreno80 | DerPraktikant: But that example is for agi? |
| 14:09.23 | DerPraktikant | u can even write dial(SIP/100&SIP/101,100) |
| 14:10.01 | DerPraktikant | it look different but is the same |
| 14:10.27 | DerPraktikant | does nobody has an idea to my problem above? :( |
| 14:10.30 | jeremy_g | do you have anything for say for vlan trunking? |
| 14:10.43 | jeremy_g | two voip phones on a diffrent vlan |
| 14:11.33 | mmoreno80 | DerPraktikant: Ok. |
| 14:11.51 | gripner | wich codec is asterisk useing by default to sip phones? |
| 14:12.34 | DerPraktikant | md5 for passwords and µlaw / alaw for the voice |
| 14:12.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:12.43 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:13.11 | DerPraktikant | u can even use sha for password , but its ab bit difficult to configure |
| 14:14.01 | *** join/#asterisk DirtyD (n=DigiD@ool-18bddad8.dyn.optonline.net) |
| 14:14.10 | DirtyD | hiya hiya. |
| 14:14.13 | DerPraktikant | hi |
| 14:14.27 | DerPraktikant | maybe u can help me... |
| 14:14.35 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 14:14.58 | DerPraktikant | i got an athentitcation error when i try to call out over the sip provider |
| 14:15.15 | DerPraktikant | registration goes fast and well |
| 14:15.34 | DirtyD | which provider? |
| 14:15.39 | DerPraktikant | but when i try to dial asterisk gives the error that the password is wrong |
| 14:15.44 | DerPraktikant | t-online :( |
| 14:15.53 | DirtyD | hmm |
| 14:16.17 | DerPraktikant | i post the extensions mom |
| 14:16.46 | NirS | hello all |
| 14:16.50 | NirS | how is everybody doing ? |
| 14:17.29 | DerPraktikant | [tonline_out] |
| 14:17.29 | DerPraktikant | exten => _0.,2,Dial(SIP/${EXTEN}@032226260403,45,Ttr) |
| 14:17.29 | DerPraktikant | exten => _0.,1,Set(CALLERID(name)=032226260403) |
| 14:17.29 | DerPraktikant | <PROTECTED> |
| 14:17.29 | DerPraktikant | exten => _0.,3,Hangup |
| 14:17.35 | jeremy_g | DerPraktikant:what are they using for auth |
| 14:17.49 | DerPraktikant | md5 i guess |
| 14:17.50 | DirtyD | http://www.ip-phone-forum.de/archive/index.php/t-94808.html |
| 14:17.53 | DirtyD | does this help? |
| 14:18.01 | *** join/#asterisk Aurs (n=Aurs@81.191.112.190) |
| 14:18.12 | DirtyD | What's that German? |
| 14:18.48 | DerPraktikant | no sry the scripts on this side dont function |
| 14:19.15 | DerPraktikant | the post autor doenst post the finaly working version |
| 14:19.42 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:20.02 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 14:20.03 | penguinFunk | hey guys, is there anyway of using logic in sip.conf? or is the dial plan in extensions.conf the only way to do it |
| 14:20.05 | b11d | morning lads |
| 14:20.40 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:20.45 | jeremy_g | does * support tls? |
| 14:21.03 | penguinFunk | the reason being... I want to use alaw when the dialled number is _9. but i want to use G729 any other time is there a way to do this? |
| 14:21.05 | DerPraktikant | in the sip.conf u can only configure ur phones, server and users |
| 14:21.26 | Ahrimanes | jeremy_g, i remember some patches in the bugtracker.. but stock asterisk doesnt.. requires sip over tcp afaik |
| 14:21.46 | penguinFunk | but since coders are defined in sip.conf and dial plans are defined in extensions.conf, how can i implement this logic ? |
| 14:21.51 | Ahrimanes | penguinFunk, hm, dont think so.. |
| 14:22.03 | jeremy_g | Ahrimanes:yup,sip over tcp is a pre-req for that. |
| 14:22.09 | DerPraktikant | maybe u can use a user whis hast allow=g729 and one with allow=alaw and link back from the extensions.conf to this user like u do by outbound calls |
| 14:22.13 | tzafrir | penguinFunk, what kind of logic? |
| 14:22.18 | jeremy_g | Ahrimanes:chan_sip3 might have tcp for sip |
| 14:22.21 | Ahrimanes | penguinFunk, only way is to have the other end require g729 or alaw.. but then * will probably transcode |
| 14:22.48 | Ahrimanes | jeremy_g, yeah.. hope work is progressing well on that :) |
| 14:23.04 | *** join/#asterisk yitzhakbg (n=yitzhakb@IGLD-83-130-227-253.inter.net.il) |
| 14:23.09 | jeremy_g | Ahrimanes: :) |
| 14:23.22 | Ahrimanes | tzafrir, choosing codecs based on dialled extension |
| 14:23.42 | yitzhakbg | Hi. Can anybody help do an initial setup? |
| 14:23.44 | DerPraktikant | oenguinFunk maybe this way : _9.,1,Dial(SIP/${EXTEN}@userinSIP.CONFwithalaw) |
| 14:23.50 | tzafrir | choose a different peer |
| 14:24.05 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:24.05 | tzafrir | yitzhakbg, ask a specific question... |
| 14:24.23 | Ahrimanes | tzafrir, wouldnt * just transcode? |
| 14:24.23 | yitzhakbg | I don't seem to be making the initial connection |
| 14:24.43 | Ahrimanes | ah of course if allow=ulaw,g729 it would switch yes |
| 14:24.52 | yitzhakbg | i've got all the debugging on, yet I can't see what's going on? |
| 14:24.56 | tzafrir | Ahrimanes, have several peer entries to the same target, each with different codec setting. Ugly, but works |
| 14:25.10 | Ahrimanes | tzafrir, yep |
| 14:25.28 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:25.30 | tzafrir | yitzhakbg, nither can we ;-) |
| 14:25.34 | yitzhakbg | Tzafrir, did we see you Mon. night at the restaurant with Mark spencer? |
| 14:25.37 | Ahrimanes | tzafrir, Set(CODEC=g729) in dialplan would be cool though |
| 14:26.36 | Ahrimanes | Set(CODEC(voice)=ulaw) and Set(CODEC(video)=h263) |
| 14:26.40 | Ahrimanes | <PROTECTED> |
| 14:26.40 | tzafrir | yup, that's me again |
| 14:26.49 | Ahrimanes | oops sorry |
| 14:27.19 | tzafrir | Ahrimanes, there's actually some work to rewrite the whole Dial command |
| 14:27.19 | yitzhakbg | Tsafrir, Can I send u an e-mail? |
| 14:27.41 | tzafrir | yitzhakbg, maybe pastebin some relevant output |
| 14:27.45 | Ahrimanes | tzafrir, oh to include codec choice, etc? |
| 14:27.46 | tzafrir | ~pb |
| 14:27.48 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 14:28.42 | yitzhakbg | OK, will do. One moment |
| 14:28.58 | tzafrir | yitzhakbg, what do you try to do? |
| 14:30.00 | *** join/#asterisk oQPa (n=uawename@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 14:30.07 | *** join/#asterisk kuto (n=kuto@58.69.158.114) |
| 14:30.28 | yitzhakbg | I'm connecting to a machine on the LAN through the X-Lite SIP phone. Seems to register, but I can't see the call connecting |
| 14:30.42 | yitzhakbg | I have to learn how to use pastebin |
| 14:31.06 | [TK]D-Fender | yitzhakbg : Copy & paste. Its not rocket science |
| 14:31.07 | tzafrir | before that: do you see it registered? |
| 14:31.30 | tzafrir | in the asterisk CLI, zap show peers |
| 14:31.38 | tzafrir | Do you see its IP address? |
| 14:31.42 | tzafrir | duh |
| 14:31.45 | tzafrir | sip show peers |
| 14:32.05 | yitzhakbg | I saw it registered from the X-Lite log. What do I tell Asterisk to display registration? |
| 14:32.06 | penguinFunk | tzafrir: I basically want to use alaw when users dial '_9.' but want to use G729 when users dial any other number... is there a way to do this? |
| 14:32.31 | yitzhakbg | you mean sip show peers? I'm not using zap |
| 14:32.44 | Ahrimanes | tzafrir, ip adresses on zap... do elaborate ;) |
| 14:33.18 | [TK]D-Fender | penguinFunk: He just said up top.... make 2 differnt peer entries. |
| 14:34.04 | yitzhakbg | I'm going to paste the peers on pastebin. One sec... |
| 14:34.15 | *** join/#asterisk Godsey (n=jason@pdpc/supporter/sustaining/Godsey) |
| 14:34.37 | Godsey | is there anything special about cisco callmanager and it's sip implementation? |
| 14:34.43 | b11d | yeah |
| 14:34.44 | b11d | ccm sucks |
| 14:34.50 | b11d | ok, maybe it doesnt.. i dont know.. |
| 14:34.54 | Godsey | my telco is offering sip but only to callmanager customers |
| 14:35.05 | yitzhakbg | It's at http://pastebin.ca/312005 |
| 14:35.12 | jeremy_g | Godsey:$$ |
| 14:35.24 | Godsey | wondering if I can change my vendor string and just tell them I have callmanager :) |
| 14:35.45 | jeremy_g | Godsey:Lolz |
| 14:35.47 | [TK]D-Fender | yitzhakbg: That tells us virtually nothing. pastebin your entire sip.conf, and then another with your extensions.conf |
| 14:36.03 | yitzhakbg | OK, will do |
| 14:36.12 | jeremy_g | yitzhakbg:do not include the passwords |
| 14:36.16 | tzafrir | yitzhakbg, seems to be registered |
| 14:36.20 | penguinFunk | okay thanks tzafrir / [TK]D-Fender |
| 14:36.29 | yitzhakbg | I don't mind. It's only on the LAN |
| 14:36.44 | jeremy_g | yitzhakbg:cute!! |
| 14:37.33 | tzafrir | next: what do you see on the CLI trace? |
| 14:37.41 | tzafrir | what version of Asterisk is it? |
| 14:38.30 | yitzhakbg | extensions.conf is at http://pastebin.ca/312011 |
| 14:38.57 | [TK]D-Fender | YAY * GUI! |
| 14:39.29 | b11d | :( |
| 14:39.55 | jeremy_g | [TK]D-Fender:you like it |
| 14:40.05 | [TK]D-Fender | jeremy_g: to DEATH! |
| 14:40.08 | yitzhakbg | Pardon me. First time I used pastebin. I think extensions.conf is at http://pastebin.ca/312014 |
| 14:40.16 | jeremy_g | yitzhakbg: :d |
| 14:40.18 | jeremy_g | :D |
| 14:40.45 | [TK]D-Fender | GUI = ass. |
| 14:40.52 | jeremy_g | yitzhakbg:pardon me =>reminds me of oliver twist |
| 14:41.03 | tzafrir | so: core set verbose 3 |
| 14:41.25 | [TK]D-Fender | I think I'll jsut quietly step away from this now.... |
| 14:41.32 | tzafrir | Do you see anything when you try to call? |
| 14:41.50 | yitzhakbg | sip.conf is at http://pastebin.ca/312015 |
| 14:42.24 | yitzhakbg | one sec... |
| 14:42.44 | Katty | morning. |
| 14:42.46 | tzafrir | yitzhakbg, mind you, if this is the 1.4 with the gui, the set up may be through users.conf (if you used the "basic" interface) |
| 14:43.43 | yitzhakbg | No. I don't see anything and verbose is set to 10 |
| 14:43.50 | [TK]D-Fender | Katty: Mew. |
| 14:43.56 | Katty | i think my shoulder is out of place this morning. |
| 14:44.01 | yitzhakbg | Yes. I'm using 1.4 Did I have to do something else? |
| 14:44.02 | Katty | it feels all..needing to crack |
| 14:44.25 | yitzhakbg | What's this about users.conf |
| 14:44.39 | [TK]D-Fender | Katty: Slam it into a wall like Mel Gibson in Lethal Weapon :) |
| 14:44.49 | Katty | hmm. |
| 14:44.54 | Katty | maybe i'll just visit the doctor instead. |
| 14:44.58 | tzafrir | yitzhakbg, anyway, you don't have any allow or disallow |
| 14:45.04 | yitzhakbg | I'm a newbie. Setup the new Asterisk GUI, but didn't understand some of the questions, so I went back to trying from the book |
| 14:45.20 | tzafrir | add a 'allow=any' to the last sip.conf entry to allow any codec there |
| 14:45.28 | yitzhakbg | Where does allow or disallow go? |
| 14:45.38 | yitzhakbg | one sec. |
| 14:45.40 | tzafrir | sorry: allow=all |
| 14:45.46 | Katty | so my boss came to me this morning and said...gee, you don't have any linux books. |
| 14:45.47 | tzafrir | which codecs may be used |
| 14:45.58 | Katty | and i said yeah, and i'd really like a few, since i sorta have to work on linux. |
| 14:46.05 | Katty | so he said okay, much to my surprise. |
| 14:46.16 | Katty | and now i'm looking for recommendations. |
| 14:46.23 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:46.38 | yitzhakbg | reload sip file now? |
| 14:47.05 | tzafrir | sip reload , or simply: reload |
| 14:47.13 | Katty | [TK]D-Fender: recommendations? |
| 14:47.27 | mercestes | Katty: I recommend more Mercestes love. |
| 14:47.31 | penguinFunk | tzafrir: how is that to work though? you mean 2 peer entries in sip.conf for each user ? |
| 14:47.46 | Katty | mercestes: i dun think that's a book title. |
| 14:47.50 | yitzhakbg | how to reload sip file or should restart Asterisk? |
| 14:47.53 | tzafrir | penguinFunk, as I said, it is a lame method, but it will work. |
| 14:47.55 | [TK]D-Fender | Katty: I suck at Linux :) Wouldn't know whats best. |
| 14:48.05 | Katty | :< |
| 14:48.09 | Katty | k, i'll enquire elsewhere |
| 14:48.13 | penguinFunk | im not sure i completely understand what you said |
| 14:48.29 | mercestes | Katty: I am in a book. |
| 14:48.35 | Katty | which asterisk book are we all recommending this week? |
| 14:48.40 | Katty | ~book |
| 14:48.42 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:48.42 | [TK]D-Fender | Katty: However the O'Reilly series of books is highly regarded. I'd say look for one for your distro, one general for Linux overall, and any specifics that you use. |
| 14:48.57 | [TK]D-Fender | Katty: The one Ture Book! |
| 14:49.00 | tzafrir | penguinFunk, or get someone (you?) to code this into chan_sip (codec selection from a channel var), if this is not already coded somewhere... |
| 14:49.02 | [TK]D-Fender | MY PRECIOUS!!!!! |
| 14:49.04 | penguinFunk | two entries in sip.conf > one with type=friend one with type=peer > both have same sip username |
| 14:49.05 | mercestes | The one book, to RULE THEM ALL! |
| 14:49.05 | [TK]D-Fender | True* |
| 14:49.15 | Katty | and in the darkness.. |
| 14:49.19 | Katty | uhh |
| 14:49.20 | Katty | i mean |
| 14:49.21 | [TK]D-Fender | Disconnect them! |
| 14:49.21 | mercestes | bind them. |
| 14:49.23 | mercestes | :D |
| 14:49.25 | Katty | yes!Q |
| 14:49.27 | mercestes | I like binding. |
| 14:49.28 | Katty | disconnect them! |
| 14:49.30 | tzafrir | dnsmasq them |
| 14:49.34 | *** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com) |
| 14:49.42 | mercestes | And in the darkness, chown them. >.> |
| 14:49.46 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 14:49.51 | kuto | Katty: freepbx is running on CentOS, its a better alternative |
| 14:49.53 | yitzhakbg | you there tsafrir? |
| 14:50.07 | tzafrir | tzafrir, but yes, I'm here |
| 14:50.19 | mercestes | kuto: What did you just ssay? |
| 14:50.21 | Katty | kuto: let's not start propaganda, mkay? |
| 14:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:50.36 | penguinFunk | the codec selection based on dialled extensions is for both incoming and outgoing calls |
| 14:50.43 | kuto | lols |
| 14:50.46 | yitzhakbg | I intentionally placed the name of a non-existant sound file in the extensions.conf to make it bomb |
| 14:51.11 | yitzhakbg | I just tried after having done resatrt gracefully. Still see nothing |
| 14:51.32 | tzafrir | it will not bomb. It will simply give a little warning that it cannot find that file and move on to the next phase |
| 14:51.38 | tzafrir | next line |
| 14:52.08 | [TK]D-Fender | kuto: I'd offer you some crack, but I can see you've got an ample supply already ;) |
| 14:52.08 | tzafrir | If you want to print something to the CLI, use NoOp |
| 14:52.12 | yitzhakbg | No warning. I think X-lite is calling OK. I see the IP of the server pop up |
| 14:52.19 | tzafrir | NoOp(got here) |
| 14:52.35 | [TK]D-Fender | tzafrir: You're parience at this is quite remarkable... |
| 14:52.40 | yitzhakbg | what do u mean NoOp. Is that a command? |
| 14:53.10 | tzafrir | yitzhakbg, make it something longer. a longer sound file, Echo, etc. |
| 14:54.13 | yitzhakbg | By what u mean to change the name of the sound file? What exactly do I have to do after every change? restart asterisk? |
| 14:54.31 | tzafrir | extensions reload or: reload |
| 14:54.34 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:54.58 | yitzhakbg | Now what? rplace the sound file? With what? |
| 14:55.19 | tzafrir | do you have the extra sounds installed? |
| 14:56.03 | yitzhakbg | I've got all the 1.4 sounds in /var/... |
| 14:56.09 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:56.17 | tzafrir | hmmm... I know a long sound file: http://karlsbakk.net/fun/asterisk-installation.wav ;-) |
| 14:57.37 | yitzhakbg | I keep on seeing: "Really destroying SIP dialog '7f29a60f0ae2b8365b5a8ba46e9eb822@127.0.1.1' Method: NOTIFY" |
| 14:57.40 | [TK]D-Fender | Katty: Ew.... No, you are really better off with straight Debian. |
| 14:57.45 | tzafrir | lyrics-louie-louie is also quite long |
| 14:57.53 | yitzhakbg | I'm not working locally. Does that have to bother me? |
| 14:58.07 | Katty | [TK]D-Fender: but ubuntu is pretty |
| 14:58.13 | *** join/#asterisk Ryushin (n=Ryushin@windwalker.openinnovations.com) |
| 14:58.24 | tzafrir | yitzhakbg, well, why don't you try listening to the sound on the phone? |
| 14:58.29 | e-ddie | Katty: pretty, how? |
| 14:58.41 | [TK]D-Fender | Katty: Its just Gnome... you can do that really easy just as you are |
| 14:58.43 | e-ddie | that brownish color is pretty ugly, imo |
| 14:58.44 | Katty | e-ddie: visually pleasing. |
| 14:58.53 | [TK]D-Fender | e-ddie: Hear hear |
| 14:58.54 | yitzhakbg | I don't hear on the phone. Just a moment |
| 14:59.16 | e-ddie | [TK]D-Fender: hear what? |
| 14:59.45 | Katty | speaking of hear, i outta start on my dialplan again. |
| 15:00.27 | [TK]D-Fender | e-ddie: Just raising a voice agains the BROWN. |
| 15:01.06 | [TK]D-Fender | Brown is a horrid colour. its teh colour of SHIT, and NO it does not work on suits, corduroy, or much else! A fasion FAILURE! |
| 15:01.07 | mercestes | <PROTECTED> |
| 15:01.16 | yitzhakbg | tsafrir, I'm lost. What do u mean listen to the sound on the phone. Also, I don't have lyrics-louie... |
| 15:01.21 | Katty | i ask dumb questions every day |
| 15:01.35 | Katty | and i'm about to ask fender to help me setup this auto answer thingy |
| 15:01.39 | tzafrir | use Echo |
| 15:01.53 | tzafrir | this will run an echo test |
| 15:01.55 | mercestes | bu tyour katty, your special. I'm just the dark overlord, mercestes. I need to know if *I* am allowed a stupid question today..:D |
| 15:02.08 | Katty | so i'm just katty |
| 15:02.09 | Katty | a girl |
| 15:02.12 | Katty | who's supposed to be dumb?! |
| 15:02.16 | Katty | what are you trying to say?! |
| 15:02.40 | tzafrir | mercestes, you just asked a dumb question. |
| 15:02.53 | mercestes | Katty: :D I said you were special. :) That's what I meant to say. |
| 15:03.32 | mercestes | Ok: So I'll ask another one. Is there a standard way to say, offer a user the ability to "Continue to hold or press 69 to leave a voicemail" while in a queue without loosing their place in the queue? |
| 15:04.33 | mercestes | like you call some places and they go, "You are the 375th person in line, the average hold time is 3 days. You may continue to hold, or press # to leave a voiecmail, and someone will not return your call." I wanna do that, and I can using a timeout but with a timeout it would basically reorder them in the queue. |
| 15:04.45 | *** join/#asterisk UlbabraB (n=salama@88-149-155-155.f5.ngi.it) |
| 15:04.51 | Katty | ^_- |
| 15:04.55 | b11d | heh 69 |
| 15:04.58 | Katty | someone just called me to inform me they changed their password. |
| 15:05.32 | Katty | as if i somehow need to keep track of all passwords inside this building. |
| 15:05.41 | mercestes | Katty: You should have gone, "Ok, just lte me know what it is so I can update the list." lol Bet they give it to you. |
| 15:05.53 | yitzhakbg | tzafrir, all I hear is a little beep. I think it's local to the SIP phone |
| 15:05.56 | Katty | i'm sure they would. |
| 15:06.39 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:06.39 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:07.47 | Katty | in sip.cfg what does 'class' refer to? |
| 15:08.00 | Katty | example: voIpProt.SIP.alertInfo.2.class="somenumber" |
| 15:08.41 | mercestes | so, has anyone done this yet? |
| 15:08.43 | Katty | is it just an identifiying number? |
| 15:08.48 | b11d | im up for anything once |
| 15:09.06 | Katty | oh dear. |
| 15:09.14 | penguinFunk | try anything twice |
| 15:09.14 | b11d | ohh yeahhh |
| 15:09.19 | b11d | really.. three times.. |
| 15:09.30 | mercestes | Katty: this sets a sip header called "alertINfo" which has to do with ring behavior, or other phone alter behaviors, such as a blinking light, a chirp, a phantom ring, a full ring, or a phaser blast to the head. |
| 15:09.39 | mercestes | Nah, just twice. |
| 15:09.48 | mercestes | Just in case you got it wrong the first time. |
| 15:10.20 | mercestes | :D |
| 15:10.22 | Katty | oooh la la. |
| 15:10.31 | yitzhakbg | tzafrir there? |
| 15:10.38 | Katty | now explain, again,...in katish terms... |
| 15:10.43 | Katty | i get sip headers now. |
| 15:10.47 | mercestes | Katty: It makes it ring special. |
| 15:10.51 | Katty | i'm going to setup a sip header named 'page' |
| 15:11.01 | Katty | and when the sip.cfg sees page... |
| 15:11.09 | Katty | alertinfo.1.value="page" |
| 15:11.31 | Katty | it's going to se.rt.ringer="phasernoise" |
| 15:11.34 | Katty | but! |
| 15:11.43 | Katty | there's this line that does not parse. |
| 15:12.08 | tzafrir | so in an echo test you don't hear yourself? |
| 15:12.08 | Katty | voIpProt.SIP.alertInfo.1.class="number" |
| 15:12.09 | mercestes | Well, alert info doesn't have to ring tho. It also has to do with what a sip notify does which, usually, reboots the phone. |
| 15:12.09 | tzafrir | yitzhakbg, so in an echo test you don't hear yourself? |
| 15:12.15 | Katty | and i see that number later on in everything |
| 15:12.20 | Katty | se.rt.classnumber.type |
| 15:12.25 | Katty | se.rt.classnumber.timeout |
| 15:12.31 | mercestes | erm, let me open up my sip.cfg |
| 15:12.33 | Katty | is it just a reference to another place in time and space?! |
| 15:12.36 | mercestes | you owe me a queue config tho..:P |
| 15:12.46 | Katty | i dunno how to queue yet |
| 15:12.54 | Katty | i'm queueless. |
| 15:13.44 | Katty | mercestes: ref: http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
| 15:14.13 | yitzhakbg | Echo test? Let me look up the instruction |
| 15:14.21 | Katty | i like echo test. |
| 15:14.43 | `Sauron | Katty: You just like hearing yourself |
| 15:15.02 | mercestes | wel, first, ignore the ipmid reference, ....that's deprecated to be included in sip.cfg |
| 15:15.21 | tzafrir | Do you see that extension of the 'Echo' being reached in the CLI trace? |
| 15:15.24 | mercestes | Katty: Also in this context the "alertinfo" is a "ring type" that is basically a "don't ring, just answer" ring tone. |
| 15:16.01 | mercestes | For the .class...I dunno what it refers to specifically, I haven't had to touch it |
| 15:16.28 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 15:16.31 | Katty | yesyes, all good. |
| 15:16.33 | Katty | very nice. |
| 15:16.36 | Katty | </borat> |
| 15:17.02 | mercestes | The class is an "index" and then under se.rt.# you set the parameters for the ring tone of that index. |
| 15:17.16 | mercestes | so the .class is for custom ring tones not already found in sip.cfg |
| 15:17.23 | mercestes | so if you wanted to make a phaser blast....there you go |
| 15:18.27 | Katty | that's for ringer. |
| 15:18.39 | Katty | but there's a whole bunch of se.rt.number variables. |
| 15:18.49 | Katty | name, type, timeout, ringer... |
| 15:18.51 | [TK]D-Fender | Katty: Don't mess with any of those. |
| 15:19.02 | danp | are "avoided initial deadlock" messages bad? |
| 15:19.06 | Katty | [TK]D-Fender: i just wanna comprehend it. |
| 15:19.11 | Katty | danp: usually yes. |
| 15:19.12 | mercestes | Don' tlisten to hom. mess..mess.... |
| 15:19.21 | danp | any tips on tracking them down? |
| 15:19.25 | mercestes | Don't listen to him. Mess..messs |
| 15:19.39 | mercestes | So, can anyone help me with my q? |
| 15:19.52 | Katty | if i wasn't queueless, i'd love to. |
| 15:20.19 | [TK]D-Fender | Katty: <alertInfo voIpProt.SIP.alertInfo.1.value="" voIpProt.SIP.alertInfo.1.c |
| 15:20.20 | [TK]D-Fender | lass="" voIpProt.SIP.alertInfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.cl |
| 15:20.22 | [TK]D-Fender | ass="4" voIpProt.SIP.alertInfo.3.value="Auto Answer" voIpProt.SIP.alertInfo.3.cl |
| 15:20.23 | [TK]D-Fender | ass="3"/> |
| 15:20.25 | mercestes | Katty: You can be in my queue anytime you wish. |
| 15:20.51 | mercestes | I won' teven assign you a timeout value. |
| 15:21.27 | mercestes | I *would* like to offer you the option to continue to hold, or press # to leave a general voicemail tho if someone could help me do that. |
| 15:21.34 | [TK]D-Fender | Katty: then under <patterns> much lower down ensure that : <AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer"/> |
| 15:21.36 | mercestes | without you loosing your place in line. |
| 15:21.36 | [TK]D-Fender | <PROTECTED> |
| 15:21.37 | [TK]D-Fender | t.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/> |
| 15:22.29 | [TK]D-Fender | Katty: Please note the unfortunate line breaks my client added and adjust accordingly |
| 15:22.30 | Katty | oh ah. |
| 15:22.42 | Katty | so the class does point somewhere else. |
| 15:23.04 | Katty | class="number" goes to the se.rt.number.type |
| 15:23.14 | [TK]D-Fender | Katty: And to USE it : exten => 31,1,SIPAddHeader(Alert-Info: Ring Answer) |
| 15:23.15 | Katty | so you could have a bazillion enteries. |
| 15:23.22 | yitzhakbg | How can I tell if I'm reaching the context I intend to use? |
| 15:23.30 | mercestes | Katty: I bet yo ucould. Yo ushould try it. |
| 15:23.31 | [TK]D-Fender | Katty: 31,2,Macro(stdexten,SIP/31,0) |
| 15:23.38 | Katty | 31,1,SIPAddHeader(Alert-Info: Sales Department) |
| 15:23.47 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:23.49 | Katty | [TK]D-Fender: mew? |
| 15:23.53 | Katty | stdexten? |
| 15:24.04 | [TK]D-Fender | Katty: No, don't mess the the type like that . It will have to match an XML tag. |
| 15:24.06 | mercestes | Katty: /[macro-stdexten :) |
| 15:24.06 | yitzhakbg | Isn't there some kind of like a breakpoint or watchpoint to inform whether it got there? |
| 15:24.20 | [TK]D-Fender | Katty: No proactical point. |
| 15:24.27 | [TK]D-Fender | practical* |
| 15:24.35 | Katty | but! |
| 15:24.40 | danp | here's an example of the deadlock stuff i'm seeing: http://pastie.caboo.se/32324 |
| 15:24.41 | Katty | if you make class="5" |
| 15:24.57 | [TK]D-Fender | Katty: You are just deptermining HOW you want it to reing, not what you want the phone to SAY when you call. You'd just muck with CallerID for that. |
| 15:24.58 | danp | any clues as to what would be causing that and how to fix it? |
| 15:24.59 | Katty | and then set se.rt.5.type="sales stuff" |
| 15:25.08 | [TK]D-Fender | Katty: Again, no point. |
| 15:25.18 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:25.25 | yitzhakbg | Can someone help a newbie setup? |
| 15:25.27 | Katty | nono |
| 15:25.28 | Katty | no |
| 15:25.31 | Katty | not what i'm trying to do |
| 15:25.34 | Katty | i want a blast group. |
| 15:25.38 | [TK]D-Fender | Katty: You are only really messing with 2 kinds of AA. Immediate AA (no ring indication or anything), and "Ring Ansewr". |
| 15:25.43 | Katty | to page a group of sales people. |
| 15:25.51 | mercestes | Katty: Oh, then...yea, leave that alone. |
| 15:26.14 | mercestes | katty: Just use a sipAddHeader("Auto Answer") and then do a Page(Sip/Tom&Sip/dick&Sip/harry) |
| 15:26.15 | [TK]D-Fender | Katty: that is *'s job, not Polycom's. You're just preparing the phone to be ABLE to auto-answer. You don't tell it 5 ways it can do it! |
| 15:26.31 | Katty | oh |
| 15:26.36 | [TK]D-Fender | Katty: Much like mercestes jsut said./ |
| 15:26.37 | Katty | okay, that works too |
| 15:26.39 | mercestes | :D |
| 15:26.45 | mercestes | yay, I get a gold star. |
| 15:26.59 | mercestes | now I just need to know how to do my queue debauchery thing. |
| 15:27.09 | Katty | stand back! i'm gonna try it! |
| 15:27.26 | [TK]D-Fender | Katty: And actually, adding a global header (if applicable to all the devices added) means you don't need to do the extra macro & context crap you saw in the WIKI example./ |
| 15:28.16 | danp | do you guys set call-limit for polycom phones? |
| 15:28.24 | danp | would that have anything to do with my deadlock stuff? |
| 15:28.53 | a1fa | D-Fender > * |
| 15:28.59 | danp | i'm trying to get any clue as to what my problem is |
| 15:29.30 | Katty | but it still doesn't matter which class number i use. |
| 15:29.48 | Katty | value="auto answer" voIpProt.SIP.alertinfo.2.class="anynumberiwantas long as it matches in ipmid.cfg" |
| 15:30.13 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 15:30.17 | danp | the classes are defined in sip.cfg; look for a tag called RING_ANSWER for an example |
| 15:30.30 | mercestes | YOu should not have an ipmid.cftg |
| 15:30.32 | mercestes | ignore that. |
| 15:30.34 | [TK]D-Fender | Katty: I suggest you take my samples as they are from my WORKING config. ipmid was discontinued 3 major releases ago :) |
| 15:30.37 | danp | (those tag names are just for your information, the class numbers they use when setting stuff up is what really matters) |
| 15:30.40 | mercestes | put everythign in sip.cfg. and update your firmware. |
| 15:30.42 | mercestes | no ipmid.cfg |
| 15:30.43 | Katty | okay. |
| 15:31.08 | tzafrir | yitzhakbg, let's see: do you know that the calls actually start? |
| 15:31.08 | Katty | so the second bit...goes into sip.cfg now |
| 15:31.13 | danp | i'm using 2.0.1 and all i had to add to my site phone config was a mapping from Ring Answer to class 4 i believe |
| 15:31.30 | yitzhakbg | tsafrir, can u help me pls? |
| 15:31.31 | tzafrir | What do you see on show dialplan getting_started |
| 15:32.04 | yitzhakbg | I'm asking how I can see some debug info, like if it hts the context or not? |
| 15:32.04 | mmoreno80 | I have a question: There is a way to hangup but keeping the channel? |
| 15:33.12 | Katty | [TK]D-Fender: where at under patterns does this go? |
| 15:33.18 | Katty | [TK]D-Fender: under callprogress? |
| 15:33.22 | Katty | [TK]D-Fender: on it's own? |
| 15:33.25 | tzafrir | show dialplan context_name shows exactly that context (exactly what Asterisk thinks it is right now) |
| 15:33.33 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:33.41 | mercestes | I need a queue guru. |
| 15:33.43 | Katty | [TK]D-Fender: under ringer? |
| 15:33.43 | yitzhakbg | tzafrir, How can I tell if my context is being reached? |
| 15:33.46 | mercestes | ....hey..that rhymed. |
| 15:33.54 | mercestes | I'm a poet and I wasn't even aware of it. |
| 15:34.25 | tzafrir | You should see that in the CLI trace, if the core debug is at least 3 |
| 15:34.38 | [TK]D-Fender | Katty: It should already exist about 1/2 way down |
| 15:34.48 | [TK]D-Fender | Katty: just ensure that it matches like I pasted |
| 15:34.49 | Katty | it's set as class 3 |
| 15:35.14 | mmoreno80 | Or, there is another way to hangup not using AUTOHANGUP and HANGUP? |
| 15:35.24 | Katty | yeah it's set. |
| 15:35.25 | Katty | all the same. |
| 15:35.51 | yitzhakbg | tzafrir, I'm not getting anything in the cli trace |
| 15:35.52 | Katty | i'm gonna give it a shot now (= |
| 15:35.54 | [TK]D-Fender | Katty: So up top : <alertInfo voIpProt.SIP.alertInfo.1.value="" voIpProt.SIP.alertInfo.1.class="" voIpProt.SIP.alertInfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4" voIpProt.SIP.alertInfo.3.value="Auto Answer" voIpProt.SIP.alertInfo.3.class="3"/> |
| 15:36.16 | Katty | there's only the first one. |
| 15:36.18 | tzafrir | Is the SIP client still registered? |
| 15:36.20 | [TK]D-Fender | Katty: For which you'd have to note that I added a class from where it used to only have 1, AND.... |
| 15:36.20 | Katty | i added the third one. |
| 15:36.30 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 15:36.48 | Katty | but all three are listed down at the bottom below patterns. |
| 15:37.15 | [TK]D-Fender | <AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer"/> |
| 15:37.17 | tzafrir | it is a friend, so it is both a peer and a user. You'll see the context to which it goes in the sip users list (but also its password) |
| 15:37.21 | [TK]D-Fender | <RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/> |
| 15:37.22 | tzafrir | sip list users |
| 15:37.32 | [TK]D-Fender | Katty: THESE 2 lines should be below ringers |
| 15:37.39 | Katty | [TK]D-Fender: yes, they are. |
| 15:37.42 | [TK]D-Fender | Katty: Alertinfo stuff is way at the top |
| 15:37.43 | Katty | [TK]D-Fender: but only 3 is above at the top |
| 15:37.53 | Katty | [TK]D-Fender: 1, and 3...ring answer isn't at the top |
| 15:37.54 | yitzhakbg | tzafrir, Yofi! finally got some debug feedback. It says "There is no existence of 'getting-started' context" What now? |
| 15:37.56 | [TK]D-Fender | Katty: Substitute with my line |
| 15:38.01 | Katty | [TK]D-Fender: but i don't need ring answer, do i? |
| 15:38.20 | Katty | i just want auto answer. |
| 15:38.26 | tzafrir | yitzhakbg, is it "getting-started" or "getting_started" (- or _ ?) |
| 15:38.45 | mmoreno80 | Please, any idea? |
| 15:38.48 | [TK]D-Fender | Katty: Its a POLITENESS thing.... do you just want to start blaring over their phone without beeping first? |
| 15:38.57 | Katty | yes ;P |
| 15:39.03 | Katty | i see what you're saying tho |
| 15:39.04 | [TK]D-Fender | Katty: Either way you're setting up a frameworks to support BOTH. |
| 15:39.15 | Katty | k, let me add that in too |
| 15:39.20 | [TK]D-Fender | Katty: Thats evil... like unannounced eavesdropping... |
| 15:39.20 | yitzhakbg | tzafrir, I'm embarresed. Now I got feedback. |
| 15:39.20 | Katty | what about visual? |
| 15:39.29 | [TK]D-Fender | Katty: Dunno about that one. |
| 15:39.32 | Katty | k |
| 15:39.38 | [TK]D-Fender | Katty: Start with these 2 |
| 15:39.59 | yitzhakbg | Here's the output: [ Context 'getting_started' created by 'pbx_config' ] |
| 15:39.59 | yitzhakbg | <PROTECTED> |
| 15:39.59 | yitzhakbg | <PROTECTED> |
| 15:39.59 | yitzhakbg | <PROTECTED> |
| 15:39.59 | yitzhakbg | -= 1 extension (3 priorities) in 1 context. =- |
| 15:40.16 | [TK]D-Fender | Katty: exten => 31,1,SIPAddHeader(Alert-Info: Ring Answer) |
| 15:40.23 | Katty | hold up (= |
| 15:40.26 | [TK]D-Fender | Katty: exten => 41,1,SIPAddHeader(Alert-Info: Auto Answer) |
| 15:40.44 | [TK]D-Fender | Katty: Those are the 2 foramts to shoose your ringtype before you do your dial. |
| 15:40.58 | tzafrir | and is that the context of the SIP phone's user? |
| 15:40.59 | yitzhakbg | tzafrir, how can I tell if it's reaching there? |
| 15:41.10 | tzafrir | it is a friend, so it is both a peer and a user. You'll see the context to which it goes in the sip users list (but also its password) |
| 15:41.31 | yitzhakbg | u mean on the SIP phone? |
| 15:41.37 | tzafrir | sip list users |
| 15:41.47 | *** join/#asterisk Dandre (n=testdan@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:42.14 | danp | hmm, i wonder if Monitor was causing my problems |
| 15:42.16 | yitzhakbg | tzafrir, no such command. Is it changed in 1.4? |
| 15:42.30 | danp | i was recording every call but i took it out last night to see if it helped today |
| 15:42.52 | tzafrir | any 1.4 user here? do I have a typo? sip show users? |
| 15:43.43 | zoa | any chinese people here ? |
| 15:43.53 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 15:44.01 | yitzhakbg | tzafrir, right it was sip show, not sip list |
| 15:44.16 | in-pt | tzafrir: sip list users |
| 15:44.18 | tzafrir | good. At least not that |
| 15:44.44 | yitzhakbg | tzafrir, Can I show the list here? its about 4 lines |
| 15:44.50 | tzafrir | so I guess that this is a 1.2 system |
| 15:45.00 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:45.03 | tzafrir | the line for that phone's user |
| 15:45.18 | *** join/#asterisk xnon (n=xnon@200.82.223.85) |
| 15:45.19 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 15:45.28 | brettnem | hello all.. long time no talk |
| 15:45.30 | yitzhakbg | output from sip show users |
| 15:45.33 | yitzhakbg | *CLI> sip show users |
| 15:45.33 | yitzhakbg | Username Secret Accountcode Def.Context ACL NAT |
| 15:45.33 | yitzhakbg | yitzhakbg 1234 getting_started No RFC3581 |
| 15:46.04 | mmoreno80 | (again) there is a way to hangup keeping the channel? |
| 15:46.27 | yitzhakbg | tzafrir, U asked: "the line for that phone's user" How to do? |
| 15:46.35 | tzafrir | so it seems to go to the context getting_started , indeed |
| 15:46.55 | brettnem | hey anyone know how to detect call progress tones (SIT) on a SIP channel? |
| 15:46.59 | yitzhakbg | tzafrir, how can u tell? |
| 15:47.26 | Katty | [TK]D-Fender: hmm. |
| 15:48.03 | Katty | [TK]D-Fender: that's hot |
| 15:48.28 | Katty | [TK]D-Fender: but the ring answer... |
| 15:48.36 | Katty | [TK]D-Fender: for some reason it just rang and rang |
| 15:48.41 | Katty | [TK]D-Fender: and didn't pick up after one. |
| 15:48.49 | brettnem | Polycom phone? |
| 15:50.06 | danp | did you reboot the phone? |
| 15:50.21 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 15:50.33 | danp | i'm just saying :P |
| 15:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:50.47 | Katty | wrong class number. |
| 15:51.35 | Katty | [TK]D-Fender: auto answer is my new hero. |
| 15:51.37 | danp | that'll do it |
| 15:51.46 | Katty | [TK]D-Fender: it gives me a warm fuzzy feeling inside. |
| 15:51.57 | danp | i'm using ring answer at this deployment i'm trying to debug |
| 15:52.07 | danp | it works well...i might take the timeout down a little |
| 15:52.10 | [TK]D-Fender | Katty: So both modes tested and happy? |
| 15:52.17 | danp | it takes just a hair too long to come off hook |
| 15:52.29 | Katty | auto answer does, ring answer had the wrong class number...phones are rebooting slowly right now. |
| 15:52.48 | [TK]D-Fender | Katty: Creepy isn't it? |
| 15:53.09 | Katty | both answer now. |
| 15:53.22 | Katty | now it's time to setup my fancy schmancy ring tone! |
| 15:55.36 | [TK]D-Fender | Katty: This part has proven tricky, esp with custom tones. |
| 15:56.01 | Katty | it's just like a number. |
| 15:56.08 | Katty | ringtype=numberinpolycomphone |
| 15:56.46 | Katty | i've got customized ringtones for everything else already...when its my extension, it plays a wav file that says incoming call from angie |
| 15:56.50 | Katty | surely it can be anymore complicated than that |
| 15:57.10 | hoobastooba | i am getting deadlock errors after every time I start Monitoring. http://pastebin.ca/312081 |
| 15:57.17 | hoobastooba | can anyone tell me why? |
| 15:57.49 | brettnem | hey anyone ever delt with detecting call progress tones (tri tones) on SIP channels? |
| 15:58.09 | Strom_M | brettnem, do you mean SIT tones? |
| 15:58.36 | brettnem | Strom_C: yes, SIT tones |
| 15:58.45 | brettnem | Strom_C: Over SIP |
| 15:59.02 | brettnem | Strom_C: transmitted in 183 SDP |
| 15:59.11 | Strom_M | brettnem, the vast majority of the time, the call never supervises if you're going to a recording with SIT tones |
| 15:59.56 | brettnem | Strom_M: I have a lot of calls going to these numbers.. I'm definately getting 183 with SDP and an inband SIT |
| 16:00.29 | brettnem | Strom_M: I don't ever get a 200, but I don't want to pass that kind of call to the user |
| 16:00.43 | Strom_M | I smell.....a telemarketer |
| 16:01.15 | brettnem | Strom_M: It's a business. at least they are non-profit. |
| 16:01.41 | Strom_M | brettnem, so only pass the call to the user if it supervises, and make note of any call that tears down without supervising. |
| 16:02.08 | brettnem | Strom_M: How do you control the call before it supervises? |
| 16:02.26 | Strom_M | what do you mean? |
| 16:02.29 | brettnem | Strom_M: Btw, using amd2 right now.. it detects SIT as answering machines |
| 16:02.39 | *** join/#asterisk killfill (n=killfill@freebsd.cl) |
| 16:02.52 | zoa | what is SIT ? |
| 16:03.15 | brettnem | zoa: tri-tone.. special information tone. <beep> <beep> <beep> we're sorry, the number you reached has been disconnected |
| 16:03.23 | zoa | aha |
| 16:03.41 | *** join/#asterisk lorinc (n=ang@caracas-2007.adsl.interware.hu) |
| 16:03.51 | Strom_M | brettnem, you're better off going based on supervision, not tones, because a lot of recordings that don't supervise also don't play SIT. |
| 16:04.44 | tzanger | hmm |
| 16:04.47 | tzanger | I wonder what's changed now |
| 16:05.02 | brettnem | how do I handle the call based on supervision |
| 16:05.13 | tzanger | send a call to unlimitel (SIP), get back busy, asterisk sees it as busy and goes to play busy to my norstar (PRI) and the norstar just hangs up |
| 16:05.56 | Strom_M | tzanger, are you supervising and sending busy tones, or are you just sending a BUSY message? |
| 16:05.56 | hoobastooba | i get that avoided deadlock after every time a call starts the Monitor command. What could I have done wrong? |
| 16:06.41 | tzanger | Strom_M: I am using priindication=outofband for the connection to the norstar, and sending busy via Busy() in the dialplan, so I am betting that I'm sending a PRI hangup with a causecode of busy (whatever that is) and terminating the call |
| 16:06.51 | Katty | hmm. |
| 16:06.57 | Katty | [TK]D-Fender: if i try to dial more than one phone.... |
| 16:07.06 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 16:07.07 | Katty | [TK]D-Fender: the second SIP/number gives me a 500 internal server error |
| 16:07.22 | [TK]D-Fender | Katty: Using Dial, or Page? |
| 16:07.32 | Katty | dial. |
| 16:07.37 | Katty | that's the problem, isn't it |
| 16:07.44 | [TK]D-Fender | Katty: Shouldn't do that... every phone then tries to fight for the call... |
| 16:07.54 | [TK]D-Fender | Katty: Yeah, mass callouts should use Page |
| 16:08.18 | [TK]D-Fender | Katty: Single Dials are a great way to go "get the hell off your ass!" and freak people out. |
| 16:09.03 | Katty | i see, i see. |
| 16:10.00 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net) |
| 16:10.27 | *** join/#asterisk Kyler (n=Kyler_La@74-132-227-26.dhcp.insightbb.com) |
| 16:11.04 | [TK]D-Fender | Katty: Also think about nifty shit like letting poeple record "reminder messages" and cron up a .call file to AA themselves :) |
| 16:11.12 | Katty | [TK]D-Fender: what's this SIP/numberx1 thingy |
| 16:11.20 | [TK]D-Fender | Katty: Where? |
| 16:11.31 | Katty | [TK]D-Fender: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page |
| 16:11.33 | Katty | example 2 |
| 16:11.46 | Kyler | I need to limit the amount of time spent in app_dictate(). I've been looking at timeout settings but it's not obvious that provides what I need. |
| 16:12.10 | [TK]D-Fender | Katty: Oh, thats jsut actually PART of the the guy's device name because he's redarded and didn't know how to set up his line keys :) |
| 16:12.23 | *** join/#asterisk karmatronic (n=boumkar@84.77.155.231) |
| 16:12.31 | Katty | k |
| 16:13.02 | [TK]D-Fender | Katty: Actually... nix that, but he's still retarded ;) Not forPolycom |
| 16:13.21 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:13.21 | [TK]D-Fender | Katty: Simple and fun now? |
| 16:13.22 | Katty | [TK]D-Fender: "here's how i got this to work for my polycom phones" <- that looks polycomish to me |
| 16:13.44 | Katty | [TK]D-Fender: that *96...the 96 is the extension? |
| 16:13.49 | *** part/#asterisk karmatronic (n=boumkar@84.77.155.231) |
| 16:13.56 | [TK]D-Fender | Katty: Thats the second part of that example... |
| 16:13.58 | Katty | [TK]D-Fender: that catch all is confusing me |
| 16:14.24 | *** join/#asterisk santibiotico (n=santi@247.Red-88-15-142.dynamicIP.rima-tde.net) |
| 16:14.25 | [TK]D-Fender | Katty: He's just using that macro to add the header to each dial. Actually.... you might HAVE to.. I'm not sure... |
| 16:14.26 | santibiotico | hi |
| 16:14.37 | santibiotico | i'm having trouble with dtmf |
| 16:14.54 | Katty | [TK]D-Fender: that set(TIMEOUT(digit)=5 means they have 5 seconds of talk time? |
| 16:14.55 | [TK]D-Fender | Katty: Does adding the header, then Page(whack'o'SIP'shere) actually AA all the phones? |
| 16:14.56 | b11d | get a restrining order |
| 16:14.59 | santibiotico | i've checked all dtmf parameters, but i might be forgetting something |
| 16:15.04 | Katty | [TK]D-Fender: lemme see |
| 16:15.23 | santibiotico | whenever i try to make a call through sip, dtmf is not working |
| 16:15.43 | santibiotico | however, if i try calling through zaptel, dtmf is working |
| 16:15.43 | *** join/#asterisk ruzulfnag (n=irc4u@90-227-16-110-no130.tbcn.telia.com) |
| 16:15.50 | [TK]D-Fender | Katty: That guy's example is to use a single little IVR to either let you page a targeted phone, or "*" to page all. |
| 16:15.58 | santibiotico | i'm using dtmfmode=rfc2833 in sip.conf |
| 16:16.05 | Katty | [TK]D-Fender: "no application 'page' |
| 16:16.14 | [TK]D-Fender | Katty: Must fix!@ |
| 16:16.24 | Katty | [TK]D-Fender: mew? |
| 16:16.26 | [TK]D-Fender | Katty: You still on that old release or is this the new box? |
| 16:16.30 | Katty | new box |
| 16:16.34 | [TK]D-Fender | Katty: 1.4? |
| 16:16.36 | Katty | yes |
| 16:16.40 | santibiotico | is there any other parameter i should look for when confguring dtmf? |
| 16:16.41 | [TK]D-Fender | Katty: HRM |
| 16:16.52 | [TK]D-Fender | Katty: Check your source for app_page |
| 16:17.17 | santibiotico | i'm sure it's not a problem of the phone, as with other asterisk server and the same phone config, dtmf is workng |
| 16:17.28 | santibiotico | any idea?? |
| 16:17.39 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 16:18.26 | Katty | [TK]D-Fender: i have app_page.c in /asterisk/asterisk-1.4.0/apps |
| 16:18.36 | Katty | [TK]D-Fender: should there be an app_page.conf file? |
| 16:18.41 | [TK]D-Fender | Katty: Do you see the .so in your modules folder? |
| 16:18.48 | [TK]D-Fender | Katty: nope. |
| 16:18.51 | Katty | no |
| 16:18.56 | Katty | that's the only app_page file on the machine |
| 16:18.57 | [TK]D-Fender | Katty: no configuration needed |
| 16:19.13 | [TK]D-Fender | Katty: Do you see a compiled .so in your source folder? |
| 16:19.53 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 16:19.54 | Katty | no |
| 16:19.58 | [TK]D-Fender | Katty: Should be under "apps |
| 16:20.04 | Katty | [TK]D-Fender: most of them have .c, .o, and .so files |
| 16:20.08 | Katty | [TK]D-Fender: but app page does not. |
| 16:20.38 | *** join/#asterisk zogulus (n=dale@host-87-74-94-64.bulldogdsl.com) |
| 16:20.40 | Katty | it didn't compile right, did it. |
| 16:20.42 | [TK]D-Fender | Katty: Somehow got left out. Someone more experienced than I can probably answer you quickly from here now that we got this far. |
| 16:21.10 | stephane_ | jour |
| 16:21.28 | santibiotico | anyone helping me with dtmf, plz? |
| 16:21.33 | Katty | [TK]D-Fender: they additional files are created when you compile tho, right? |
| 16:22.07 | [TK]D-Fender | Katty: Yup, part of the compile process. I am not a Linux programmer, so I can't ay much more than this.... |
| 16:22.14 | Katty | k |
| 16:24.06 | undrdawg | why does dialing 14435551212 fail with Call Failed: Not Found |
| 16:24.07 | undrdawg | in kphone |
| 16:24.31 | [TK]D-Fender | undrdawg: Means there is no match in your dialplan. |
| 16:24.33 | Katty | [TK]D-Fender: josh said it's because i don't have zap. |
| 16:24.44 | [TK]D-Fender | Katty: EEK! |
| 16:24.48 | Katty | [TK]D-Fender: and meetme needs zap too, and meetme didn't compile right either |
| 16:24.53 | undrdawg | you happen to know the code to dial in the US with voipuser.org? |
| 16:25.00 | [TK]D-Fender | Katty: Yeah, you need a timing source, because its effectively a Meetme |
| 16:25.03 | undrdawg | i think they're in uk |
| 16:25.10 | Katty | [TK]D-Fender: call me dumb.... |
| 16:25.15 | [TK]D-Fender | Katty: No it wouldn't have |
| 16:25.15 | Katty | [TK]D-Fender: zap doesn't mean zaptel tho, right? |
| 16:25.29 | Katty | [TK]D-Fender: i'm confuzzled about zap. |
| 16:25.31 | [TK]D-Fender | Katty: Minor oversight. It didn't have Meetme blatantly labeled on it :) |
| 16:25.58 | brettnem | Strom_M: Hey sorry, I had a phone call. :/ So how do I handle the call based on supervision? Call progress is immeditelly sent to the endpoint |
| 16:26.10 | [TK]D-Fender | undrdawg: I'm sure they'e a basic SIP provider like everyone else. |
| 16:26.13 | Katty | [TK]D-Fender: now the million dollar question, where and what is zap....... |
| 16:26.23 | Katty | [TK]D-Fender: and how do i get it on here, so i can recompile everything |
| 16:26.26 | [TK]D-Fender | Katty: You know.... Zaptel..... |
| 16:26.44 | Katty | [TK]D-Fender: ah ha! |
| 16:26.48 | Katty | [TK]D-Fender: i just got zaptel on here |
| 16:26.51 | Katty | [TK]D-Fender: yesterday in fact. |
| 16:26.58 | Strom_M | brettnem, you could generate call files, which dial a specific extension only if they answer |
| 16:26.59 | Katty | [TK]D-Fender: should i just...recompile? |
| 16:27.04 | [TK]D-Fender | Katty: Recompile and rebuild in the right order and all should be good. |
| 16:27.13 | undrdawg | they claim to let you make a few calls for free |
| 16:27.16 | Katty | [TK]D-Fender: there's a right order? :< |
| 16:27.30 | [TK]D-Fender | Katty: Normally libpri, zaptel, then Asterisk |
| 16:27.39 | *** join/#asterisk beehive (n=michael@pool-71-246-201-56.washdc.fios.verizon.net) |
| 16:27.42 | Katty | k |
| 16:27.45 | undrdawg | i'd pay for real service if it'd let me make a call to see if the call is clear |
| 16:27.59 | brettnem | Strom_M: Hmm, I'm not sure how that would work in a call center environment.. I'd have to think on that.. call files seems messy |
| 16:28.16 | undrdawg | right now i kinda got screwed a bit by a service that doesnt have service in my area so i have to wait for the void on my CC |
| 16:28.26 | Strom_M | brettnem, are the calling users dialing the phone manually? |
| 16:28.30 | brettnem | Strom_M: It'd be nice to totally disable inband progress |
| 16:28.35 | Katty | [TK]D-Fender: i guess that means i need libpri too |
| 16:28.35 | brettnem | Strom_M; well no |
| 16:28.36 | Katty | [TK]D-Fender: joy. |
| 16:28.42 | undrdawg | you have any ideas [TK]D-Fender? |
| 16:28.53 | brettnem | Strom_M: Calls are made with Originate API |
| 16:28.55 | brettnem | we |
| 16:28.57 | brettnem | er |
| 16:28.59 | brettnem | AMI |
| 16:29.35 | santibiotico | arggg i'm getting crazy hehehe any help with dtmf over sip¿ |
| 16:29.55 | brettnem | santibiotico: use RFC2833 |
| 16:29.58 | brettnem | and be happy |
| 16:30.33 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net) |
| 16:31.35 | santibiotico | brettnem hehe it is what i am using right now |
| 16:31.36 | santibiotico | and i'm having problems |
| 16:31.36 | Katty | [TK]D-Fender: wait... |
| 16:31.36 | Katty | [TK]D-Fender: pri? am i even using pri? |
| 16:33.07 | [TK]D-Fender | Katty: An ounce of prevention ;) |
| 16:33.15 | mmoreno80 | There is a war to access channel's variables when the channel is closed with hangup? |
| 16:33.22 | mmoreno80 | s/war/way |
| 16:33.29 | Katty | [TK]D-Fender: i don't have libpri on my machine |
| 16:33.35 | [TK]D-Fender | undrdawg: Yeah... stop being cheap and just choose somebody, or ask them for free service and cross your fingers. |
| 16:33.40 | Katty | [TK]D-Fender: actually, i don't even have zaptel cards in here yet ;) |
| 16:33.48 | [TK]D-Fender | Katty: FIX !@ |
| 16:33.52 | Katty | i can't |
| 16:33.57 | [TK]D-Fender | Katty: No matter! |
| 16:33.58 | Katty | the zaptel cards are in the other box, in use. |
| 16:34.17 | [TK]D-Fender | Katty: I mean just download libpri! Doesn't matter if you don't think you'll need it NOW. |
| 16:35.15 | Katty | [TK]D-Fender: i love how there are instructions on how to get it on the wiki |
| 16:35.23 | Katty | [TK]D-Fender: s/are/aren't/ |
| 16:36.08 | Katty | how do you guys get libpri on debian? libpri-dev? |
| 16:36.11 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:39.58 | mercestes | So..... |
| 16:40.30 | mercestes | In my queues, I want to add an option that says, "Press 1 to continue to hold, or press # to leave a general voicemail." What is the best way to do that??? |
| 16:41.30 | brettnem | mercestes: I think you just set a context in the queue definition for those extensions |
| 16:41.55 | [TK]D-Fender | mercestes: Not happening. You press something to LEAV the queue, not CONTINUE to hold. |
| 16:42.09 | [TK]D-Fender | mercestes: So its "continue to hold, or press "X" for VM) |
| 16:42.16 | Katty | well. |
| 16:42.18 | Katty | i feel better now. |
| 16:42.21 | Katty | this calls for a cookie. |
| 16:42.50 | [TK]D-Fender | ! |
| 16:43.16 | brettnem | haha.. just don't 1 a valid extension? :) |
| 16:44.20 | [TK]D-Fender | brettnem: smooth...... |
| 16:44.27 | brettnem | ;) |
| 16:44.38 | [TK]D-Fender | brettnem: But liable to piss off teh paranoid :) |
| 16:44.55 | brettnem | well.. good.. ;) haha |
| 16:45.04 | brettnem | we're out to get those paranoid types you know |
| 16:45.15 | [TK]D-Fender | brettnem: 1... 1... 1... 1... 1... 1... 1... 1... (fingers fall off due to RSI). Lawsuit, etc.... |
| 16:45.30 | brettnem | oh, well when you put it that way |
| 16:45.56 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 16:46.23 | anonymouz666 | When I call from SIP/100 to Zap/1 and get a tone to dial to any number... is there a way to get this number on CDR? |
| 16:46.49 | brettnem | reset cdr? |
| 16:46.51 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 16:47.09 | anonymouz666 | ?? |
| 16:47.44 | [TK]D-Fender | anonymouz666: No, nor should you. You shouldn't just give them raw line tone like that |
| 16:47.46 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:48.04 | [TK]D-Fender | anonymouz666: Dial(Zap/1/1234567) |
| 16:48.12 | anonymouz666 | CDR is alway from SIP/100 to Zap/1 and that's correct. If I call from SIP/100 to 999 through Zap/1 i don't get this 999 on cdr |
| 16:49.13 | *** join/#asterisk burnproof (n=jsharryp@124.106.206.155) |
| 16:49.59 | anonymouz666 | oh sorry I am talking about the flash operator panel... you drag the external 1 (zap/1) and drop on SIP/100... so you get the tone to dial on SIP/100... but there is no way to record the number that you will call |
| 16:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:51.29 | DirtyD | Hi. |
| 16:51.44 | anonymouz666 | then you call to 999 when you hang up, the cdr will be sip/100 -> zap/1 |
| 16:52.23 | [TK]D-Fender | anonymouz666: Well that clearly jsut won't cut it and you're up a creak... |
| 16:52.50 | [TK]D-Fender | anonymouz666: Need a better way to initiate calls. |
| 16:52.53 | DirtyD | Is it possible to interface a SIP/Device with a Desktop agent? Such as CRM Software? An example of something I'm looking to do is when a Sip extension rings, customer information will display on the user's desktop. |
| 16:53.28 | hoobastooba | anyone here recording files to ramdisk? I am trying to make my ramdisk 1GB, but it doesnt work... anything larger than ramdisk_size=16000 fails |
| 16:53.31 | [TK]D-Fender | DirtyD: Jsut before doing your Dial, tell your dialplan to do some sort of script that will do your screen-pop. |
| 16:53.40 | [TK]D-Fender | DirtyD: This script has nothing to do with * however. |
| 16:54.14 | DirtyD | TK, thanks.. I kinda understand what to do. Thanks. |
| 16:54.51 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 16:55.03 | anonymouz666 | [TK]D-Fender but If I get the DTMF digits I could have the number |
| 16:56.24 | [TK]D-Fender | anonymouz666: Except you CAN'T. * will not just spy on DTMF and collect. It will never know how many to expect or when to stop. IVR's would screw you. |
| 16:56.33 | [TK]D-Fender | anonymouz666: So this is another case of TFB. |
| 16:57.27 | anonymouz666 | TFB? |
| 16:57.34 | wunderkin | heh heh |
| 16:59.11 | undrdawg | anyone ever use vonage? |
| 16:59.31 | DirtyD | Asterisk can't detect DTMF? Oh no, how will I ever use asterisk to take credit card payments. |
| 16:59.34 | undrdawg | i noticed a promo for a free phone card that doesnt really become active until tomorrow |
| 16:59.51 | undrdawg | i was wondering if i could use kphone or the like to dial with vonage |
| 16:59.59 | [TK]D-Fender | ~tfb |
| 17:00.02 | jbot | rumour has it, tfb is Too #&^$ing bad.... |
| 17:00.18 | anonymouz666 | heh heh |
| 17:00.39 | [TK]D-Fender | DirtyD: Please read in context...... |
| 17:01.16 | [TK]D-Fender | DirtyD: It means that the 10 million DTMFS you send AFTER your dial have no impact on the # * was TOLD to dial initially. |
| 17:01.25 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net) |
| 17:01.42 | [TK]D-Fender | undrdawg: You just keep looking for pain... it will find you... |
| 17:01.55 | undrdawg | i need to call someone |
| 17:02.00 | wunderkin | in the butt |
| 17:02.02 | undrdawg | its free trial i can cancel heh |
| 17:02.09 | wunderkin | and you need vonage for that? |
| 17:02.30 | undrdawg | no |
| 17:02.35 | undrdawg | i wanted phonehog minutes |
| 17:02.41 | undrdawg | for signing up free and then cancelling |
| 17:02.59 | undrdawg | 500 minute phone card for free |
| 17:03.35 | hoobastooba | Global charged me 10cents per. |
| 17:03.39 | undrdawg | i used to do this all the time with aol ads and stuff |
| 17:03.40 | hoobastooba | errr. wrong window |
| 17:03.48 | undrdawg | and get free calling minutes |
| 17:03.51 | hoobastooba | but now you know. |
| 17:04.04 | hoobastooba | that is how much they charged me for DID numbers ;-) |
| 17:04.17 | anonymouz666 | who is using 1.4 in production enviroment ? |
| 17:05.07 | undrdawg | oh noes |
| 17:05.14 | undrdawg | screw that im not buying hardware :P |
| 17:06.01 | undrdawg | some of them you can just get a stupid PO box and whore out your address to spammers for free minutes |
| 17:06.09 | *** join/#asterisk alamantia (n=Anthony@65.4.24.231) |
| 17:06.15 | [TK]D-Fender | undrdawg: Look at all the running around you doing being a cheap-ass. God, just get Skype and pay their 10$ charge and abuse away. |
| 17:06.21 | mercestes | [TK]D-Fender: So do the queues accept DTMF while playing music on hold?? |
| 17:06.30 | [TK]D-Fender | mercestes: Yes |
| 17:06.57 | undrdawg | A: its $15 |
| 17:07.02 | undrdawg | B: skype sucks |
| 17:07.12 | [TK]D-Fender | mercestes: Set"context=" in the queue def and that will define the exist context in which you should have a matching single digit entry for place you'd like them to go as they leave. |
| 17:07.17 | undrdawg | C: other phone co took the damn money out of my card so i cant |
| 17:07.34 | *** join/#asterisk waverly360 (n=waverly@209.149.58.214) |
| 17:07.40 | mercestes | omgzorz. You rock. |
| 17:07.42 | [TK]D-Fender | undrdawg: How much did they take? |
| 17:09.23 | [TK]D-Fender | lunch time... BBIAF |
| 17:10.55 | mercestes | l8s |
| 17:10.56 | undrdawg | well $25 |
| 17:11.00 | mercestes | Heya, Brettnem, long time no hear |
| 17:11.07 | undrdawg | not loads, but it was a prepaid CC that i have to pay money to put money on |
| 17:15.24 | frawd | anonymouz666: i'm using 1.4 in production |
| 17:16.32 | frawd | anonymouz666: but i'm kind of stupid |
| 17:20.57 | frawd | while i'm alone talking, can anyone tell me why the "hookstate" of an analog line shown with "zap show channel x" doesn't show the real hook state, but the "connected state" (cable plugged in or not)?? |
| 17:21.08 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 17:21.16 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
| 17:22.28 | *** join/#asterisk Skarmeth (n=Skarmeth@201009061013.user.veloxzone.com.br) |
| 17:24.21 | a1fa | D-Fender > * |
| 17:24.37 | a1fa | frawd : 1.4 > 1.2 |
| 17:29.56 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:31.55 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 17:31.58 | jmls | howdy ! |
| 17:32.10 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 17:32.23 | jmls | wanted to know if there was anyone using 1.4 with realtime queues and queue members |
| 17:32.41 | Kyler | Is it possible to get the channel ID from "sip show channels" or something similar? I can't even get a usable ID from "show channels". |
| 17:32.53 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:34.57 | [TK]D-Fender | frawd: Hook-state has nothing to do with being PLUGGED IN. |
| 17:35.04 | *** join/#asterisk shinux__ (n=shinux@196.220.30.223) |
| 17:36.43 | penguinFunk | anonymouz666: why do you ask? |
| 17:36.55 | penguinFunk | we're are using it in production as of 1 hour ago |
| 17:37.06 | penguinFunk | we're using it in production as of 1 hour ago even |
| 17:37.07 | penguinFunk | lol |
| 17:39.45 | a1fa | DFend! |
| 17:39.54 | Kyler | Finally figured out how to hangup a SIP channel from the CLI. |
| 17:39.55 | Kyler | 1. Type "soft hangup " hit [tab]. |
| 17:39.57 | Kyler | 2. Grab the channel names that result. |
| 17:39.59 | Kyler | 3. Backspace over "soft hangup ". |
| 17:40.00 | Kyler | 4. Execute "show channel chanid" for each of the channels discovered in step #2. |
| 17:40.02 | Kyler | 5. Execute "soft hangup chanid" for the ones I want to kill. |
| 17:41.45 | a1fa | cool |
| 17:41.54 | a1fa | and if you only figured out not to paste to channel, that would be nice |
| 17:42.37 | Kyler | "paste"? Who pasted? |
| 17:42.58 | a1fa | you did |
| 17:43.10 | Kyler | Not here. Not today. |
| 17:43.18 | a1fa | you just did> |
| 17:43.41 | Kyler | Let's be real clear. You're claiming that you watch me "cut and paste" some text into this channel? |
| 17:44.14 | Kyler | From where do you think I cut/pasted it? |
| 17:45.20 | a1fa | i dont know.. it doesnt matter anymore |
| 17:46.18 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 17:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:52.01 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 17:52.27 | *** join/#asterisk sjwilliamson2007 (i=swilliam@209.42.110.46) |
| 17:56.19 | *** part/#asterisk Dandre (n=testdan@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:57.10 | Dr-Linux | mercestes: around? :) |
| 17:59.17 | Dr-Linux | My problem is that, i can't transfer the call to the agi(something.agi). It executes the file and crash, while i can transfer the call to any other extension. |
| 17:59.39 | Dr-Linux | here is my Debug logs: http://pastebin.ca/312187 |
| 18:01.48 | *** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 18:02.29 | clyrrad | Dr-Linux: I had an issue like this with an AGI before - I found that it was an issue with my code - I had a syntax error, another time I used the echo command for debugging and that caused the problem too |
| 18:02.40 | caio1982 | does someone here uses asterisk with UTF-8 (conf files, databases, doesnt matter) ? |
| 18:02.55 | caio1982 | (i meant, does it work, asterisk supports utf-8?) |
| 18:03.17 | Dr-Linux | clyrrad: aww but it doesn't work with my any Agi, |
| 18:03.35 | Dr-Linux | clyrrad: when i dial directly the agi, it just works fine |
| 18:03.55 | Dr-Linux | but when i transfer the call to the agi, it crashes |
| 18:04.02 | *** join/#asterisk wm4k (n=chris@194.164.236.240) |
| 18:04.08 | *** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:04.12 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:04.24 | Dr-Linux | clyrrad: is there any solution for this. Or do you suggest me any other way for Transfer ? |
| 18:04.28 | clyrrad | Dr-Linux: the EXACT same code is executed each time? It does not branch out to other code from an if? |
| 18:04.54 | clyrrad | like if (transfer) blah blah blah if (direct) do something else? |
| 18:04.57 | *** join/#asterisk yitzhakbg (n=yitzhakb@IGLD-83-130-77-225.inter.net.il) |
| 18:05.05 | clyrrad | im refering to your AGI code |
| 18:05.16 | Dr-Linux | clyrrad: well, i have about 8 AGI's , and they work just fine since long |
| 18:05.45 | Dr-Linux | clyrrad: but now i want my CSR should trnasfer the caller back to the agi IVR |
| 18:05.59 | Dr-Linux | so in this case agi doesn't work |
| 18:07.02 | clyrrad | Dr-Linux: im not sure why its crashing on you - I remember pulling my hair out when mine was doing that - it ended up being the way I coded the AGI |
| 18:07.12 | *** join/#asterisk wacky_ (n=wacky@modemcable188.232-131-66.mc.videotron.ca) |
| 18:07.15 | wacky_ | heya :) |
| 18:07.27 | wacky_ | Is there a way to connect Asterisk to the JACK audio connection kit, in some way ? |
| 18:07.33 | Dr-Linux | clyrrad: my problem is only with tranfser |
| 18:07.45 | Dr-Linux | clyrrad: is there any laternative way to transfer the call? |
| 18:07.47 | clyrrad | Dr-Linux: yep I know |
| 18:07.50 | *** join/#asterisk crich1999 (n=crich@port-212-202-210-130.dynamic.qsc.de) |
| 18:08.22 | Dr-Linux | clyrrad: the other way you mean? :) |
| 18:08.27 | clyrrad | Dr-Linux: I normally transfer all calls on the actual phone - using its transfer button. Have not done it with any AGI |
| 18:09.01 | Dr-Linux | clyrrad: hhm.. yeah i can transfer the call everywhere but not to the AGI |
| 18:09.14 | beehive | Hello folks. I have some very technical oriented questions. Our service provider needs answers on how Asterisk deals with RTP. Any takers? |
| 18:09.18 | clyrrad | your trying to transfer from the phone to the AGI? |
| 18:10.02 | [TK]D-Fender | Dr-Linux: And if you call your AGI normally w/o transfer? |
| 18:10.29 | Dr-Linux | [TK]D-Fender: that works just fine |
| 18:10.34 | [TK]D-Fender | beehive: Just ask, and see what you egt. |
| 18:10.48 | [TK]D-Fender | get* |
| 18:10.51 | Dr-Linux | [TK]D-Fender: not only one agi but i tried the same, on many agi's also on 2 different servers |
| 18:11.21 | beehive | ok.. is Asterisk 'SIP RFC 3261 Compliant' My answer is no. |
| 18:11.30 | [TK]D-Fender | Dr-Linux: I'm referring SPECIFICALLY to the one that crashed. Have you proved that it only crashes on a transferred call? |
| 18:12.51 | wacky_ | would it be hard for Asterisk to connect to Gstreamer or the JACKit.. |
| 18:13.05 | wacky_ | I'd like to use Asterisk for on-air calls, in an Internet radio station... |
| 18:13.06 | Dr-Linux | [TK]D-Fender: yes, and i'm trying to resolve this issue since before x-mas , i often asked here but no solution |
| 18:13.16 | Dr-Linux | [TK]D-Fender: i changed the agi file, same happend |
| 18:13.23 | Dr-Linux | i changed the server, same happend |
| 18:13.34 | [TK]D-Fender | wacky_: Seriously doubt it. |
| 18:14.06 | [TK]D-Fender | Dr-Linux: I said the SAME AGI. as in call it from a transfer = crash, and then immediately restart and try to call it direct = crash? |
| 18:14.11 | Dr-Linux | [TK]D-Fender: anthm told me to debug it and give me logs, he suspected it's MASQU.... problem. |
| 18:14.50 | Dr-Linux | [TK]D-Fender: no, directly always works |
| 18:15.13 | Dr-Linux | transfer never works if the transfering to an agi |
| 18:15.29 | beehive | <PROTECTED> |
| 18:16.12 | Hmmhesays | [TK]D-Fender did you get my multipage faxes yesterday? |
| 18:16.49 | [TK]D-Fender | Hmmhesays: Nope. Think its my fault though |
| 18:17.24 | Dr-Linux | [TK]D-Fender: is there any other way to transfer the call back to the agi? |
| 18:17.35 | Dr-Linux | since blind transfer doesn't work |
| 18:17.46 | [TK]D-Fender | Dr-Linux: Not sure of a way to trick it.... |
| 18:18.28 | Dr-Linux | [TK]D-Fender: i'm not sure if it's bug or needs some configuration |
| 18:18.38 | Dr-Linux | i'm sure it's CHANNEL issue |
| 18:19.09 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 18:19.20 | TripleFFFF | anyone got the voicemail /myapp thing to use templates or html ? |
| 18:19.45 | Dr-Linux | [TK]D-Fender: this guy has a little similiar problem: http://lists.digium.com/pipermail/asterisk-dev/2005-July/013862.html |
| 18:20.57 | [TK]D-Fender | Dr-Linux: Ancient news, and the only thing in common is a failed transfer. Nothing to do with AGI even. |
| 18:21.19 | Dr-Linux | hhm.. |
| 18:22.09 | beehive | Does asterisk support E.164 numbers with a + in the front? |
| 18:23.35 | ruzulfnag | hello all, does anybody have any suggetions for this: |
| 18:23.54 | ruzulfnag | triggered by a http-GET, I want asterisk to dial a specific phonenumber and ask the user to press a key... |
| 18:24.09 | ruzulfnag | then I want the http-get to return the value of the key that was pressed |
| 18:24.59 | ruzulfnag | I'm thinking of hacking something using the manager-API, but maybe there is some other way? |
| 18:26.03 | *** part/#asterisk wm4k (n=chris@194.164.236.240) |
| 18:27.50 | beehive | Another item on the voip provider interop questionnair: List all SIP messages used. |
| 18:27.58 | beehive | anyone have a list? |
| 18:32.26 | *** part/#asterisk wacky_ (n=wacky@modemcable188.232-131-66.mc.videotron.ca) |
| 18:32.32 | *** join/#asterisk wm4k (n=wm4k@194.164.236.240) |
| 18:32.38 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 18:32.57 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:33.23 | [TK]D-Fender | beehive: I think you'll need to go to the WIKI for all this stuff... |
| 18:33.24 | [TK]D-Fender | ~wikis |
| 18:33.26 | jbot | wikis is probably http://www.voip-info.org |
| 18:35.14 | *** join/#asterisk oQPa (n=uawename@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 18:37.07 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 18:38.27 | *** join/#asterisk mmoreno80 (n=mmoreno8@200.123.180.33) |
| 18:38.54 | TripleFFFF | i guess not |
| 18:39.09 | mmoreno80 | There is a way to get channel variables from a hanged channel? |
| 18:39.55 | TripleFFFF | hmm the fact its hung is meant locked and not acceible i assume |
| 18:40.56 | danp | this system i'm working on seems to be doing a lot better since i took out the calls to Monitor |
| 18:41.38 | perd | anyone have the cisco 7912 sip firmware? |
| 18:42.32 | mmoreno80 | TripleFFFF: But if I set AUTOHANGUP and then take the var ANSWEREDTIME, for example, there is a error. So, I need access to the channel. |
| 18:42.47 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 18:42.56 | brettnem | What is AUTOHANGUP? |
| 18:43.06 | perd | it's off the hook |
| 18:43.18 | perd | or is that on the hook, i cant get slang right these days. |
| 18:43.20 | *** join/#asterisk Grnd-Wire (n=groundwi@71-217-127-175.tukw.qwest.net) |
| 18:43.39 | TripleFFFF | yo |
| 18:43.41 | Grnd-Wire | good morning! |
| 18:43.45 | TripleFFFF | HANG= FUCKED |
| 18:44.07 | TripleFFFF | if your door is locked.. you can break a window..to get in but still cant open door |
| 18:44.24 | mmoreno80 | brettnem: http://www.voip-info.org/wiki/view/set+autohangup |
| 18:44.25 | brettnem | so is AUTOHANGUP= AUTOFUCKEDUP ? |
| 18:44.28 | TripleFFFF | so its unreliable once locked |
| 18:44.34 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 18:47.06 | brettnem | Hmm when was AUTOHANGUP added? |
| 18:47.24 | brettnem | I've been "gone" for a while. :) |
| 18:48.11 | mmoreno80 | brettnem: Is agi command. |
| 18:48.41 | brettnem | huh? |
| 18:48.57 | Grnd-Wire | Has anyone integrated Asterisk with a Merlin Legend/Magix over a T1/PRI connection? (like a Digium board) |
| 18:49.36 | brettnem | couldn't be too hard... |
| 18:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:50.34 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:50.40 | *** join/#asterisk ManxPower (n=manxpowe@68.113.119.198) |
| 18:51.30 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-174-126.red.bezeqint.net) |
| 18:51.51 | yitzhakbg | tzafrir, are you back? |
| 18:52.16 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:53.12 | Grnd-Wire | brettnem: I'm sure it's not hard - just wondering if someone has done it, so they can tell me if there are any limitations they have identified.. I would definately set it up as a PRI, so all of the caller ID info would pass.. |
| 18:54.26 | brettnem | it should work fine provided everyone follows the protocols |
| 18:54.43 | Grnd-Wire | hehe.. Do you have any experience with Digium's T1 boards yourself? |
| 18:54.55 | brettnem | it'll probably not work at all, work without callerid, or work just fine |
| 18:55.06 | brettnem | yes, I do.. |
| 18:55.13 | brettnem | but i don't particularly like them |
| 18:56.04 | Grnd-Wire | brettnem: This is the sort of information I'm looking for! Tell me what it is you don't like, and what brand do you like instead? |
| 18:56.41 | brettnem | well, I got turned off by all of the digium boards quite a while ago for poor resource utilization, echo, and overall quality problems |
| 18:57.10 | brettnem | I use Sangoma boards instead. I've never had any problems with sangoma hardware.I think they are just built better |
| 18:57.25 | brettnem | but that's my opinion |
| 18:57.51 | Grnd-Wire | brettnem: So Sangoma has hardware echo prevention as well? |
| 18:57.59 | brettnem | yes |
| 18:58.35 | Grnd-Wire | hmm.. and the pricing is pretty competitive - In fact, I think Sangoma is less expensive.. |
| 18:59.00 | Grnd-Wire | Do they have support for getting things setup, like Digium does? |
| 18:59.19 | brettnem | I'm sure they have some support. However, if you can't setup the T1 card. You probably can't setup Asterisk. so you may want to consider a consultant. |
| 18:59.22 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-4-30.lsanca.dsl-w.verizon.net) |
| 19:00.05 | brettnem | Then again, if you can setup asterisk, you can probably handle the T1 card as well. I wouldn't be too intimidated |
| 19:00.08 | brettnem | http://www.sangoma.com/main/products/hardware/cards |
| 19:00.19 | Grnd-Wire | heh.. Well, I can setup Asterisk, and I'm quite familiar with T1's in both data and voice utilizations.. I've just never made Asterisk use a T1 board.. Only Digium FXO ports.. |
| 19:00.35 | brettnem | oh, then you'll be fine |
| 19:00.55 | brettnem | they've got some pretty good documentation on it.. read up before you decide |
| 19:01.08 | Grnd-Wire | oh that's happy! and they have PCI-E/X cards! So you can use them with newer servers.. |
| 19:01.29 | brettnem | The Digium cards *are* easier to install.. But that's mainly because someone won't allow competitor drivers commited to the trunk. :) |
| 19:02.03 | brettnem | Grnd-Wire: Sangoma has been making T1 hardware for computers for like 10-15 years I think. They know what they are doing. |
| 19:02.03 | Grnd-Wire | tee hee.. |
| 19:02.32 | Grnd-Wire | Yah! Well I know you can buy Soekris boxes with Sangoma hardware integrated with the unit.. |
| 19:02.49 | brettnem | yes, I have one in my drawer here actually. :) |
| 19:03.11 | Qwell[] | brettnem: "Won't allow"? |
| 19:03.16 | brettnem | it's a pretty little think |
| 19:03.19 | Qwell[] | brettnem: When they file a disclaimer, we'll put them in |
| 19:03.22 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 19:03.35 | Qwell[] | until then, *WE* would be in violation, if we were to do so :) |
| 19:03.45 | brettnem | Qwell: :) I'm not really a part of it. And I've been gone for quite some time. |
| 19:04.32 | Grnd-Wire | brettnem: <drool> Can I borrow it!? I need to buy one of these boards for testing.. and to build the proof-of-concept that I'll use to sell this to a customer that wants to link two of their Magix systems together. |
| 19:05.09 | brettnem | I'm not sure how it'll work with T1 boards.. The one I have I think is analog only. |
| 19:05.18 | brettnem | but I think they have models suitable for T1 |
| 19:05.27 | brettnem | you're proof of concept will work |
| 19:05.32 | brettnem | it's really quite trival actually |
| 19:05.42 | brettnem | or trivial as the case may be |
| 19:06.01 | brettnem | I don't know if I'd waste my time getting it to work on soekris tho |
| 19:06.10 | brettnem | it's pretty, but do you really need that form factor? |
| 19:06.12 | Grnd-Wire | ya.. I'm heading over there now.. I'm wondering if someone has a good Linux distro that is built around ramdrives, or at least running with as little drive access as possible.. |
| 19:06.30 | brettnem | see, that's what I'm saying about wasting time on soekris. |
| 19:06.37 | Grnd-Wire | brettnem: hmm - Everything I'm doing is Mini-ITX, so that would actually be ok |
| 19:06.46 | Grnd-Wire | yeah - and that's the one reason I haven't deployed them already. :D |
| 19:06.52 | brettnem | Just get a real pc, get whatever nice stable distro out there and you'll be fine.. should be up in a few hours. :) |
| 19:07.18 | mercestes | Grnd-Wire: Check out astlinux for a ramdrive distro fo rasterisk |
| 19:07.38 | Grnd-Wire | mercestes: Are you serious? oooh! |
| 19:07.39 | Kyler | Grnd-Wire: Have you considered an IDE flash drive? I use one in a colo machine. |
| 19:07.46 | mercestes | Yea it's largely read only |
| 19:07.50 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 19:08.12 | brettnem | astlinux is kinda a pain if you want to do "anything else" from what I know about it.. HOwever, it'll make you happy with based asterisk setup like what you've descried |
| 19:08.49 | Grnd-Wire | Kyler: I've got one in use right now running OpenBSD, and I get all sorts of weird IDE bus errors.. but that could just be the OBSD implementatin, and the fact it's trying to talk to it at UDMA 2 :P |
| 19:09.02 | Grnd-Wire | Thanks for the advice on astlinux - I'm going to go look at that! |
| 19:09.20 | brettnem | it's been around for a while.. it should make you happy |
| 19:09.45 | Grnd-Wire | Well I like to be happy. :) |
| 19:10.24 | Grnd-Wire | So does anyone know about the SLAtrunk functions in * v1.4 ?? I've read the changelog, but it's very very vague. I'm not sure where to get documentation on what it does, or how it works.. ? |
| 19:11.02 | *** join/#asterisk RoyK (n=roy@ti211310a080-13838.bb.online.no) |
| 19:11.18 | brettnem | Grnd-Wire: Did you read bugnotes on #7701? |
| 19:11.43 | brettnem | oh |
| 19:11.47 | brettnem | there arn't any. :) |
| 19:13.19 | Grnd-Wire | no - I guess that's what I'm asking for, is a starting place for that.. It seems to me the only Asterisk documentation that really exists is all of the people who spend time documentation stuff they've made work.. and if it weren't for them, I wouldn't know nearly as much as I know! I certainly plan on documenting my Magix<->Asterisk experimentation, since I'm not seeing anything identical in the "Legacy Interfacing" sections. |
| 19:13.38 | file | ignore the SLA Stuff forn ow |
| 19:14.04 | brettnem | sounds like the voice of reason to me |
| 19:14.26 | Grnd-Wire | Is it still too broken to be useful? I guess what I'm mostly interested in.. is HOW is it going to work once it's ready to use? |
| 19:14.26 | Grnd-Wire | It is even too early to get a handle on that? |
| 19:14.56 | file | okay - the SLA stuff right now is not what you expect and not what people want, therefore it's going to be rewritten by an individual who has used key systems and know how people expect them to work |
| 19:15.38 | Grnd-Wire | aha! That's pretty cool.. ok.. I won't even bother asking "how long", because I know you couldn't have an answer.. even though I still want to. :D |
| 19:20.40 | Strom_C | file: want me to ship you the 1A2 frames I have rotting away in a garage? :) |
| 19:22.18 | file | Strom_C: :P |
| 19:23.57 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 19:24.34 | jtexter3 | If I'm using native formats (wav, format_mp3) for music on hold, how can I get it to cycle, like when using mpg123 rather than always starting hold from the beginning of the file? |
| 19:26.01 | ManxPower | jtexter3: I think I saw a resolved bug that indicated the starting sound file was not random. |
| 19:26.25 | Qwell[] | ManxPower: he wants more than that |
| 19:26.35 | jtexter3 | Qwell: Exactly |
| 19:26.35 | Qwell[] | he wants it to continue to stream when no calls are on hold |
| 19:26.42 | Qwell[] | it's not possible with files |
| 19:26.50 | ManxPower | Ah! |
| 19:26.58 | jtexter3 | I was afraid you were going to say that |
| 19:26.58 | ManxPower | Nope, not possible with native. |
| 19:27.26 | Qwell[] | Strom_C: You just volunteered to do the SLA stuff :P |
| 19:27.28 | jtexter3 | So, I need to set it up to call an external app that just keeps it streaming? |
| 19:27.39 | Strom_C | Qwell[]: I don't know C |
| 19:27.41 | *** join/#asterisk unik-rados (n=rados@c-68-62-71-76.hsd1.mi.comcast.net) |
| 19:27.48 | Qwell[] | pfft, minor detail |
| 19:27.52 | Strom_C | haha |
| 19:36.31 | *** join/#asterisk masonf (n=masonf@dungle.vineyard.net) |
| 19:36.34 | DirtyD | Anyone familiar with how Asterisk, MGCP and NCS are coming along? |
| 19:36.35 | *** part/#asterisk mmoreno80 (n=mmoreno8@200.123.180.33) |
| 19:38.25 | masonf | anyone heard of a zaptel card screetching for about a second every once in a while? |
| 19:39.02 | *** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-48f27bd45b34a138) |
| 19:41.48 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 19:42.24 | markit | hi, I'm unable to understand what means "overlap dialing" (mISDN)... what is in practice? |
| 19:42.51 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:42.51 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:43.16 | *** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net) |
| 19:44.36 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 19:46.23 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 19:47.57 | RoyK | One question - I want to run this SIP/SIP gateway, preferably multihomed, replacing two other ones. The problem is - will this one be able to reply to requests from both IPs with the correct source IP? or even - to the right gateway? I can't see how this can be done. Should I use a default gateway per nic? |
| 19:50.34 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:50.34 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:51.03 | zzirhc | RoyK: I guess that can be fixed by setting up the routing on the box correctly. www.lartc.org |
| 19:52.16 | RoyK | zzirhc: any specific link? there's quite a lot at lartc |
| 19:52.26 | zzirhc | RoyK: http://www.lartc.org/howto/lartc.rpdb.html |
| 19:53.02 | zzirhc | RoyK: and also http://www.lartc.org/howto/lartc.rpdb.multiple-links.html |
| 19:53.18 | sjwilliamson2007 | hey did they change RDNIS in 1.4? |
| 19:53.23 | zzirhc | RoyK: assuming that I understood your problem correctly |
| 19:54.19 | sjwilliamson2007 | I don't seem to get this information anymore, ie ${RDNIS} would give me a value in 1.2, not in 1.4 I get nothing |
| 19:55.45 | RoyK | zeeesh: box has two nics, one at x.x.x.1 and one at x.x.y.1 (for reference). if a client on the net contacts x.x.x.1, it should reply with the correct IP, through the correct gateway. same with x.x.y.1 |
| 19:56.42 | RoyK | zeeesh: so I guess split routes is what i'm looking for |
| 19:57.14 | *** join/#asterisk casterman (n=susafder@83.214.0.170) |
| 19:59.51 | masonf | when I get get run zttest I get numbers down to 97. Could it be my machine is too slow? |
| 20:00.39 | RoyK | masonf: see http://karlsbakk.net/fun/dirty-advice.txt for reference |
| 20:02.21 | RoyK | is 1.4 stable yet? |
| 20:03.37 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:04.54 | masonf | RoyK: could you be a bit more descreet with the file name? |
| 20:05.21 | masonf | RoyK: Thanks! now the noise stopped |
| 20:05.42 | RoyK | hehe |
| 20:06.18 | masonf | RoyK: Was my question too RTFM for you or where you just having a good time |
| 20:06.21 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:08.01 | jart | all i do is play the blues |
| 20:08.26 | RoyK | masonf: just having fun. there's generally no reason to run zttest without initial problems. is audio bad? is that because of zap? etc. etc. |
| 20:13.39 | masonf | RoyK: yes. I get a schreech that lasts about half a second every once in a while |
| 20:13.40 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 20:14.34 | *** join/#asterisk JoeDeveloper (n=jdevel@www.airlinksystems.com) |
| 20:14.45 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:15.47 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 20:17.06 | JoeDeveloper | What kind of things should I check for if my SpeechCreate() is returning an error? |
| 20:17.31 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 20:17.46 | Asteriskmonkey | off topic, but does anyone have experice with hylafax? |
| 20:17.46 | *** join/#asterisk entelechy (i=user@72.54.40.206) |
| 20:18.01 | TheCops | Asteriskmonkey yup |
| 20:18.28 | Asteriskmonkey | cool, I seem to have it going but its mangling my email attachemnts to 64 bytes any ideas where to look whats causeing that? |
| 20:19.12 | TheCops | Asteriskmonkey, sorry I don't use this feature in Hylafax, only paging/faxing from a web-app. sorry |
| 20:19.53 | TheCops | Asteriskmonkey, look at your fax2file converter |
| 20:20.50 | Asteriskmonkey | will do thanks :) just need a place to start looking thanks |
| 20:21.10 | Asteriskmonkey | the tifs are fine, just email is mangles :P |
| 20:23.57 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:24.33 | *** part/#asterisk wm4k (n=wm4k@194.164.236.240) |
| 20:24.55 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 20:31.31 | RoyK | masonf: what sort of hardware? ht enabled? |
| 20:34.52 | masonf | X100P |
| 20:34.56 | masonf | ht? |
| 20:36.15 | *** join/#asterisk zogulus (n=dale@host-87-74-115-144.bulldogdsl.com) |
| 20:36.17 | *** join/#asterisk panitaxx (n=iaponte@201.243.154.170) |
| 20:36.51 | RoyK | masonf: hyperthreading. it's known to fsckup digium cards or drivers |
| 20:36.59 | zogulus | hello folks, anyone know of alternative queue impls for Asterisk? |
| 20:37.11 | RoyK | zogulus: app_icd |
| 20:37.20 | [TK]D-Fender | masonf: pastebin "cat /proc/interrupts" |
| 20:37.39 | masonf | its freebsd |
| 20:37.42 | zogulus | RoyK: cheers I'll take a look |
| 20:37.43 | RoyK | zogulus: but I don't know if it's been ported to 1.2/1.4 |
| 20:38.05 | Katty | [TK]D-Fender: i wiped the box. |
| 20:38.42 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 20:38.42 | [TK]D-Fender | Katty: Whee! |
| 20:38.47 | Katty | aye. |
| 20:38.50 | Katty | it's gonna be clean! |
| 20:38.50 | mercestes | Katty: Is it all clean now? |
| 20:38.58 | Katty | mercestes: yesh. |
| 20:39.00 | mercestes | yay! |
| 20:39.04 | [TK]D-Fender | masonf: Dunno about Zaptel on BSD, sorry |
| 20:42.14 | *** join/#asterisk unsucht (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 20:42.32 | unsucht | what is the point behind priorities over 100? |
| 20:42.48 | unsucht | like when a dialplan skips from 6 to 102 |
| 20:43.35 | unsucht | anyone? |
| 20:44.39 | mercestes | unsucht: Well, it's a deprecated option known as "priority jumping" in which, when encountering an error, asterisk would take the current priority and add 100. |
| 20:44.39 | markit | unsucht: is a "trick" to have a jump in the dialplan in certain conditionz... i.e. like "if busy, jumps to priority + 101" |
| 20:44.58 | *** join/#asterisk rpm (n=russell@66.119.170.34) |
| 20:45.20 | unsucht | you say depreciatiing does that mean it will be gone eventually? |
| 20:45.21 | masonf | [TK]D-Fender: http://pastebin.com/856199 |
| 20:45.21 | mercestes | unsucht: So what you are probably seeing is exten => blah,1,Dial(Sip/blah) exten => blah,2,Voicemail(ublah@blah) exten => blah,102,Voicemail(ublah@blah). |
| 20:45.26 | Marty-OTT | mansonf: I just paid M. Solodev to upgrade Zaptel for BSD |
| 20:45.29 | Marty-OTT | FreeBSD mind you |
| 20:45.41 | Marty-OTT | my contribution to Open Source |
| 20:46.01 | mercestes | unsucht: This means that if you run into an error or priority 1 in the dial then play voicemail isntead of being stupid and going "Syntax Error or file not found." or something. |
| 20:46.40 | mercestes | unsucht: I didn' tsay deprecating. I said deprecated. You have to specify -j somewhere to enable it again. the prefered method is the use of ${DIALSTATUS} |
| 20:47.47 | *** join/#asterisk tacubo (n=tacvbo@189.137.15.33) |
| 20:47.52 | unsucht | ok, thanks, one more question: let say i have some external php script not related to asterisk and I want to dial an exention when some condition is met, is this possible |
| 20:47.57 | *** join/#asterisk queuetue (n=scott@MTRLPQ02-1177745854.sdsl.bell.ca) |
| 20:48.04 | Marty-OTT | Hey... recently... I just configure Asterisk with a Blacklist to block numbers |
| 20:48.11 | mercestes | unsucht: Yes. |
| 20:48.12 | unsucht | like lets say a fire alarm or something |
| 20:48.14 | Marty-OTT | But I need to do it on a per-destination-nmber bases |
| 20:48.16 | Marty-OTT | basis |
| 20:48.32 | *** join/#asterisk ta^3 (n=tacvbo@189.137.15.33) |
| 20:48.33 | unsucht | would it work to just us the $agi object in that script? |
| 20:48.33 | zogulus | am I right in saying that it wouldn't be possible to write a queue using AGI? |
| 20:48.50 | Marty-OTT | In other Words... if number 555-1212 dial *60 and blocks 333-4444, well, 333-4444 is block to dial 555-1212 but can get to everyone else |
| 20:49.20 | mercestes | unsucht: possibly. |
| 20:49.27 | Marty-OTT | In order to do this, I'm going to have to create table to store the info when people do *60 with source and destination number - can I do that with AstDB? Thinnking of using MySQL |
| 20:49.29 | unsucht | is there a better way? |
| 20:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:50.52 | Marty-OTT | Asterisk seems very flexible in letting you use AGI to use Perl, for example to manage calls... but I'm not there yet |
| 20:51.15 | Marty-OTT | so, anyone, can I create my own table in AstDB? If so... .. hmm... lmme check wiki |
| 20:52.11 | Strom_C | yes you can |
| 20:52.44 | Strom_C | Set(DB(users/${CALLERID(num)}/${EXTEN})=1) or something |
| 20:53.20 | Marty-OTT | Strom_c: so what is users? Is that the table I would create? |
| 20:53.34 | Marty-OTT | I don't have my head completely wrapped around AstDB - I'm used to SQL |
| 20:53.42 | Strom_C | brb, phone |
| 20:53.42 | *** join/#asterisk karmatronic (n=karmatro@84.77.155.231) |
| 20:53.45 | Marty-OTT | ok |
| 20:55.19 | Strom_C | ok |
| 20:55.19 | Marty-OTT | ok |
| 20:55.19 | wunderkin | not ok |
| 20:55.19 | Strom_C | basically, the astdb is a simple berkeley db |
| 20:55.19 | Strom_C | the data is organizesd as key/value pairs |
| 20:55.19 | Marty-OTT | which I've never had any experience with.. |
| 20:55.19 | Strom_C | keys can have subkeys |
| 20:55.24 | Marty-OTT | phone.. just a sec. |
| 20:55.27 | Strom_C | all keys must be in a family |
| 20:55.28 | Strom_C | ok |
| 20:57.06 | Marty-OTT | back |
| 20:57.35 | Marty-OTT | ok,.. that's what I find weird.. I.e. when I entered a number to be blocked yesterday... the number as the key .. but I would have expected the number to be the value |
| 20:57.43 | *** join/#asterisk entelechy (i=user@72.54.40.206) |
| 20:57.57 | Marty-OTT | so the family was blacklist, the key was 5551212 and the value was 1 |
| 20:58.01 | Marty-OTT | That was confusing |
| 20:58.04 | JoeDeveloper | Anyone using lumenvox know what I can check/look for if my speechCreate() function is giving me an ERROR=1? |
| 20:58.32 | entelechy | hi - i'm new to asterisk and trying to sort out dialtone / PSTN providers... anyone know a web site with maybe a side by side comparison of different providers rates and services? |
| 20:58.56 | Strom_C | Marty-OTT: well, you can check if a key exists fairly easily |
| 20:59.08 | Marty-OTT | phone again... :} |
| 21:00.02 | *** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com) |
| 21:00.05 | rpm | does asterisk lock the astdb? or will it not get mad at me if i try to open it externally? |
| 21:00.10 | b11d | nah |
| 21:00.14 | b11d | i rsync mine every 30 seconds |
| 21:00.19 | b11d | no complaints |
| 21:00.31 | rpm | good stuff. |
| 21:00.34 | b11d | cant speak to editing it live though.. then asterisk might bitch |
| 21:00.41 | rpm | yeah |
| 21:02.19 | entelechy | im surprised nobody here has a provider to plug? over on #freeswitch i asked the same question and immediately had PM's from several individuals proffering their services. |
| 21:02.57 | sjwilliamson2007 | so does anyone know why my ${RDNIS} and ${CALLERIDNUM} variables don't seem to work on my test 1.4 asterisk setup, they work on my production 1.2 |
| 21:03.36 | wunderkin | because you werent paying attention to the deprecation message |
| 21:03.39 | mercestes | If I were in the commercial sector I would gladly offer my services. |
| 21:03.49 | sjwilliamson2007 | wunderkin, ah, |
| 21:03.52 | Strom_C | sjwilliamson2007: because those were deprecated as of 1.2 and removed as of 1.4 |
| 21:04.16 | sjwilliamson2007 | Strom_C, wunderkin, where is the info on this? |
| 21:04.27 | Strom_C | changes.txt, upgrade.txt |
| 21:04.33 | wunderkin | ${CALLERID(rdnis)} ? maybe? and ${CALLERID(num)} ... show function callerid... and that |
| 21:04.36 | *** join/#asterisk waverly360 (n=waverly@209.149.58.214) |
| 21:05.48 | [TK]D-Fender | entelechy: Thats because most of the people involved with freeswitch are carrier types, so most of them ARE businesses that hope to use it. |
| 21:05.49 | entelechy | mercestes: maybe you could recommend a decent provider with flat rate nationwide and good international rates? im a total n00b to this, was hoping to find a side by side comparison of different providers somewhere |
| 21:06.04 | sjwilliamson2007 | i just see something about a callerid change, but not specific info in upgrade.txt |
| 21:06.09 | [TK]D-Fender | entelechy: Here's you'll find mostly normal USERS |
| 21:07.42 | entelechy | [TK]D-Fender: this is a good thing, im sure i will be getting down and dirty with the nuts and bolts and asterisk.conf files, but first im stuck with a telephony implementor who just doesnt happen to have a preferred provider they work with... they tend to do LAN setups with FXO->PSTN and voip for LAN only |
| 21:07.42 | sjwilliamson2007 | for the record, CALLERID(<all|num|RDNIS...>) |
| 21:08.22 | [TK]D-Fender | entelechy: Which is what any normal business WOULD do... |
| 21:08.50 | entelechy | [TK]D-fender: well, we already have a T1 here, so we really dont have any physical lines to attach to a FXO |
| 21:08.57 | [TK]D-Fender | entelechy: But if you're talking USA, then VoicePulse seems less assy than most. |
| 21:09.19 | [TK]D-Fender | entelechy: Ditch the T1 for ADSL, and get a partial PRI out of it :) |
| 21:09.32 | sjwilliamson2007 | wunderkin, Strom_C thanks for the pointer |
| 21:09.35 | *** join/#asterisk ta^3 (n=tacvbo@189.137.15.33) |
| 21:09.41 | [TK]D-Fender | entelechy: End cost would probably be a fair bit less |
| 21:10.32 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 21:10.32 | Katty | does it really matter where you put libpri, zaptel, and asterisk when you get ready to compile them? |
| 21:10.32 | Strom_C | no |
| 21:10.39 | Katty | good. |
| 21:10.40 | Strom_C | you can put them into /home/suzie/dogballs/catsex/ if you wanted to |
| 21:10.47 | Katty | that'd be hot. |
| 21:10.57 | b11d | well.. most of the time it doesnt matter |
| 21:10.59 | b11d | sometimes it certainly does |
| 21:11.05 | Strom_C | but if you have root access, why not just put it in /usr/src? |
| 21:11.13 | b11d | im a /usr/local/src guy myself :) |
| 21:11.15 | Katty | is that what usr src is there for? |
| 21:11.20 | Katty | ;) |
| 21:11.21 | Strom_C | duh |
| 21:11.21 | b11d | /usr/src is system shit |
| 21:11.30 | [TK]D-Fender | Katty: Highly advised to use /usr/src |
| 21:11.34 | Katty | [TK]D-Fender: why? |
| 21:11.35 | b11d | local packages should go to /usr/local/src |
| 21:11.39 | [TK]D-Fender | Katty: and symlink the asterisk & zaptel folders |
| 21:11.40 | b11d | system sources stay in /usr/src |
| 21:11.46 | b11d | but im a BSD guy |
| 21:11.48 | Katty | [TK]D-Fender: mew? |
| 21:11.53 | sjwilliamson2007 | b11d right |
| 21:11.55 | Katty | [TK]D-Fender: i've only done symlinks once. |
| 21:11.55 | [TK]D-Fender | Katty: because many add-in scripts expect them to be there and we want a sane install |
| 21:12.02 | [TK]D-Fender | Katty: Mew. |
| 21:12.03 | Katty | oh ah. |
| 21:12.14 | b11d | meh.. this is how systems become "messy" |
| 21:12.16 | [TK]D-Fender | Katty: We're about to DOUBLE your experience! Time for a raise! |
| 21:12.19 | Katty | oh god. |
| 21:12.22 | Katty | GOD NO |
| 21:12.40 | [TK]D-Fender | 5 dolla! 5 dolla! |
| 21:12.43 | entelechy | [TK]D-Fender: hmmm... well... we're contractually stuck with the T1 provider... and either way, even if we switched to ADSL, theres still the matter of getting someone to provide dialtone/PSTN/phone # for any new lines that arent already served over the T1 (more contractual nonsense) |
| 21:12.44 | Katty | keep your money - i want my sanity! |
| 21:12.48 | b11d | 5 dolla love you long time |
| 21:13.06 | entelechy | voicepulse looks good but they look maybe consumer related? we need like 10 lines, 10 fixed phone numbers |
| 21:13.09 | [TK]D-Fender | entelechy: Fine, like I said, KEEP the T1, just cange to PRI signalling, and divide the channels :) |
| 21:13.21 | b11d | if PRI is an option from his vendor, that'd work |
| 21:13.22 | Katty | i'd sure like me a pri. |
| 21:13.31 | Katty | instead of this channel bank stuff. |
| 21:13.39 | Katty | silly t1 analog line crap |
| 21:13.40 | b11d | yeah im glad I went to PRI instead of doing that |
| 21:13.51 | [TK]D-Fender | Katty: Last I recall.... you HAVE one. Your idiot boss just has you using that stupid channel bank only to go into TDM cards! |
| 21:13.55 | b11d | what kind of rates are you guys seeing for PRI out there? |
| 21:14.00 | b11d | im paying $450/mo for PRI |
| 21:14.02 | Katty | [TK]D-Fender: it's nota pri, it's a t1 |
| 21:14.05 | b11d | $1000 installation.. which was waived. |
| 21:14.22 | [TK]D-Fender | Katty: Close enough to have changed with 1 phones call, and 1 card.... |
| 21:14.33 | [TK]D-Fender | b11d: Usuall 0-500 |
| 21:14.38 | Katty | [TK]D-Fender: i think we just need a t1 and a pri |
| 21:14.40 | *** join/#asterisk jyme (n=jp@66.230.172.198) |
| 21:14.47 | b11d | 0 would be cool |
| 21:14.49 | [TK]D-Fender | Katty: Doubt you need T1... just PRI.... |
| 21:14.50 | sjwilliamson2007 | T1 servive over SDSL with pri signalling |
| 21:14.57 | Katty | [TK]D-Fender: oh trust me, we need both |
| 21:15.06 | b11d | PRI implies T1.. |
| 21:15.07 | sjwilliamson2007 | *service |
| 21:15.08 | Katty | [TK]D-Fender: really really |
| 21:15.08 | [TK]D-Fender | Katty: You host services locally? |
| 21:15.09 | entelechy | [TK]D-Fender: well, actually, I'd love to use a PRI interface myself, except the provider currently uses a cisco 2496/8 to break out our phone lines off the T1... im pretty sure they wont let us plug into the T1 directly, when we complete our migration to the asterisk system they will be switching out the 2496 for some other box that requires we use "Sipconnect" |
| 21:15.30 | Katty | [TK]D-Fender: we're sharing the t1 with 8 phone lines......it's god awful slow. |
| 21:15.39 | cpm | Ewww |
| 21:15.42 | cpm | just get another T1 |
| 21:15.44 | [TK]D-Fender | entelechy: You could do COMPLETELY without that other gear.... |
| 21:16.17 | [TK]D-Fender | Katty: So split voice/data? Convert to partial PRI, and get an ADSL. Cheapr & faster... |
| 21:16.20 | *** part/#asterisk JoeDeveloper (n=jdevel@www.airlinksystems.com) |
| 21:16.27 | sjwilliamson2007 | http://en.wikipedia.org/wiki/Digital_Signal_1 |
| 21:16.37 | jyme | hi, i am trying to setup a sangoma wanpipe card to replace a broken digium care. however the T1 isnt coming up properly. The only error i get is: http://pastebin.ca/312395 |
| 21:16.39 | Katty | [TK]D-Fender: i think i'll stick with getting an asterisk box up by myself first. |
| 21:16.45 | jyme | any thoughts? |
| 21:16.58 | [TK]D-Fender | Katty: Good idea :) The rest can wait a little bit |
| 21:17.08 | cpm | Katty, how many folks? |
| 21:17.11 | Katty | but hot dog! we having paging! |
| 21:17.13 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
| 21:17.35 | Katty | i've been calling people sporadically throughout the day without warning |
| 21:17.43 | Katty | especially the ones i have video cameras on throughout the building |
| 21:18.01 | Katty | it's been reeeeeeeeeall good ^_^ |
| 21:18.02 | [TK]D-Fender | Katty: "I'm waitching youuuuuuuuu!!!!!!" |
| 21:18.47 | sjwilliamson2007 | so is it true that T1 (&DS1) no longer refer to the signalling, but just a set of services? and can be signalled using other forms of SDSL? |
| 21:18.57 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 21:19.02 | entelechy | [TK]D-Fender I know, that would entail getting a PRI itnerface card and plugging the T1 directly into the box. Unfortunately the provider (Cbeyond, a reseller of sprint i believe) doesnt allow that as I think they couldnt support their own management interface for the separate lines that we get from them... |
| 21:19.05 | `Sauron | DS1 isstill signaling |
| 21:19.21 | sjwilliamson2007 | `Sauron cool |
| 21:19.22 | Strom_C | DS1 is the framing and whatnot, T1 is the actual physical interface spec |
| 21:20.03 | [TK]D-Fender | entelechy: Supremely shitty telco, and being bound by a long term contrac as you say you are, this whole setup fall under the category of "unfortunate" |
| 21:20.52 | *** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 21:21.06 | Dr-Linux|home | perd: around ? :) |
| 21:21.24 | entelechy | [TK]D-Fender yeah it is *definitely* unfortunate, I agree 100%. I'd rather NOT get the 6 lines we're currently getting from Cbeyond, they use a minute plan and it sucks, but we're locked in contractually for another year... so I wanna at least make sure now that we're migrating to asterisk that any new lines we get are from *anyone else but them* :) |
| 21:22.38 | entelechy | we need like 10 more lines/phone #'s within the next 2 weeks |
| 21:22.48 | [TK]D-Fender | entelechy: EEK. |
| 21:23.21 | [TK]D-Fender | entelechy: Tieing yourself to a VoIP provider isn't a great deal either...... |
| 21:23.24 | entelechy | and im the lucky one stuck researching what providers might suffice since our asterisk implementor has no suggestions to offer for providers |
| 21:23.38 | monsted | use several providers? |
| 21:23.42 | entelechy | [TK]D-Fender well the ideas was , of course , not to get contractually locked in too tightly with any VOIP providers either at this point |
| 21:23.55 | [TK]D-Fender | entelechy: Might be better to pay the startup of a new partial PRI from someone else. I wouldn't want to run a business on voip for termination/origination. |
| 21:24.13 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 21:24.24 | Marty-OTT | strom_c: you there? |
| 21:24.28 | Strom_C | yes |
| 21:24.35 | Marty-OTT | awesome! :) |
| 21:24.36 | Marty-OTT | <PROTECTED> |
| 21:24.42 | Marty-OTT | back on this... friggin phone |
| 21:24.45 | Marty-OTT | anyways... |
| 21:25.17 | Marty-OTT | so yeah, that key/value pair thing. I would have expected the VALUE to be like a number 5551212 but when I blocked the nubmer, the key was 5551212 and the value was "1" |
| 21:25.19 | entelechy | [TK]D-Fender: you wouldnt want to run a business on voip termination/origination? why? you suggest getting a 2nd PRI, wouldnt we still be in the same situation? |
| 21:25.29 | Marty-OTT | If you could shed some light on that - it would be great! |
| 21:25.32 | Strom_C | Marty-OTT: that's because i didnt finish explaining the theory |
| 21:25.37 | Strom_C | you ran off for a phone call |
| 21:25.48 | Marty-OTT | yeah... i know... sorry about that. Really appreciate your help |
| 21:26.00 | [TK]D-Fender | entelechy: Don't get locked in for more than a year, and make sure they let you use your equipment. |
| 21:26.15 | [TK]D-Fender | entelechy: Think Quality of service, and uptime guarantees. |
| 21:26.16 | *** join/#asterisk nays85 (i=nays85@shell.thehostbusters.com) |
| 21:26.39 | entelechy | [TK]D-Fender yes... we absolutely require a provider with a decent SLA |
| 21:27.18 | entelechy | and we are looking to go month to month if possible... there seem to be a few providers who offer that |
| 21:27.35 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 21:27.43 | [TK]D-Fender | entelechy: a tad extreme. Single year is acceptable... |
| 21:28.33 | CunningPike | ~seen serge-v |
| 21:28.57 | jbot | i haven't seen 'serge-v', CunningPike |
| 21:29.11 | CunningPike | ~seen serge |
| 21:29.14 | jbot | i haven't seen 'serge', CunningPike |
| 21:29.16 | *** join/#asterisk mdruedal (n=mdruedal@port812.ds1-ro.adsl.cybercity.dk) |
| 21:29.43 | Dr-Linux|home | anynone using agi? |
| 21:29.43 | entelechy | [TK]D-Fender true. as long as they dont have some kind of hidden minutes cap on their "unlimited" packages, as well as having a SLA with a decent then a 1 year contract would be acceptable |
| 21:31.01 | *** join/#asterisk SimoAmi (n=simoami@user-1087vl2.cable.mindspring.com) |
| 21:32.28 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 21:34.45 | SimoAmi | is anyone familiar with the dialplan command Backgroud()? |
| 21:35.46 | CunningPike | Man, that asterisk.org website does that to me every time |
| 21:37.15 | *** join/#asterisk alamantia (i=Anthony@nat/digium/x-ae8e3fa57bbb40fb) |
| 21:37.39 | *** part/#asterisk E-bola (i=bola@rbii-valhalla.mrseb.co.uk) |
| 21:37.42 | *** join/#asterisk BSDTech (n=RNeese@pool-71-118-41-36.lsanca.dsl-w.verizon.net) |
| 21:39.04 | b11d | i hate that new website |
| 21:39.07 | b11d | so much |
| 21:39.10 | b11d | but whatever.. |
| 21:39.36 | `Sauron | It's awfully... "cute" |
| 21:40.00 | terrapen | anybody ever used Juniper's media gateways? |
| 21:40.23 | *** part/#asterisk BSDTech (n=RNeese@pool-71-118-41-36.lsanca.dsl-w.verizon.net) |
| 21:41.11 | b11d | you could cut a roast on that website. |
| 21:41.59 | terrapen | i'm finding that the Cisco media gateway (AS54xx series) that can handle a T3 is ridiculously priced |
| 21:42.02 | Dr-Linux|home | i'm not sure if ever my "transfer the call to the agi" will be resolved or not :( |
| 21:42.20 | SimoAmi | I'm trying to play a background sound while retrieving information from a remote webserver! how would I do that? |
| 21:43.12 | terrapen | so i'm thinking demuxing the T3 into T1/PRI and putting 4 or 8 PRIs on each * server |
| 21:43.57 | terrapen | and having them convert to IAX2 w/ G.729, which would be sent out to the main * server that handles the call center queue |
| 21:43.59 | *** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com) |
| 21:44.20 | *** join/#asterisk adorah (n=admin@87.69.57.228.cable.012.net.il) |
| 21:45.30 | perd | is anyone familiar with 7902 and chan_skinny? |
| 21:46.44 | *** join/#asterisk SwK (n=Silik0nJ@rbn1-216-180-75-83.adsl.hiwaay.net) |
| 21:46.45 | *** join/#asterisk mikefoo (n=mikefoo@166.84.140.254) |
| 21:47.12 | tzanger | which is the format windows understands best, wav or wav49? I can never remember |
| 21:47.12 | mikefoo | Anyone know of a way to tell how many pages in a .tiff? |
| 21:48.05 | mikefoo | for windows.. its wav |
| 21:48.15 | mikefoo | wav49 is an asterisk thing, no? |
| 21:48.45 | zoa | not really |
| 21:50.15 | perd | mikefoo each page of a tiff is a layer |
| 21:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:50.35 | perd | i bet imagemagick could tell you how many laters an image has |
| 21:52.48 | Dr-Linux|home | perd: your config were nice :) |
| 21:53.09 | b11d | not if the image was flattened |
| 21:53.18 | b11d | if you had the project file, then yes.. |
| 21:54.34 | perd | ah nice dr, so it all works now? |
| 21:55.13 | Dr-Linux|home | perd: yeah, your configs were pretty easy to understand. |
| 21:55.23 | perd | if the image was flattened then it would only contain one page |
| 21:55.30 | Dr-Linux|home | perd: but i'd need your suggestion for one case |
| 21:55.32 | perd | unless tiffs have multiple ways of defining pages |
| 21:55.58 | perd | what case is that, dr |
| 21:56.17 | b11d | yeah im not so sure about tiff |
| 21:56.25 | b11d | all i know is tiff's are usually gigantic |
| 21:56.26 | *** join/#asterisk Bazy (n=bazy@86.125.51.251) |
| 21:56.29 | perd | haha yeah |
| 21:57.07 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 21:57.53 | Dr-Linux|home | perd: the hard part is, i want from the user 16 digits input, i mean card number |
| 21:58.24 | Dr-Linux|home | perd: but not sure, how timeout will work in this case |
| 21:58.39 | Dr-Linux|home | maybe your suggestion would helpfull |
| 21:58.44 | *** join/#asterisk oQPa (n=uawename@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 21:59.33 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 22:00.09 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189) |
| 22:00.57 | Dr-Linux|home | perd: timeout starts once caller ends speaking, or i will to put a specific timeout? |
| 22:01.25 | SimoAmi | I'm trying to use backbround(), but it blocks execution of the dial plan until someone dials a digit |
| 22:02.19 | SimoAmi | is there a way to start a background sound and stop it later on when done |
| 22:02.38 | b11d | you want to be able to control it via DTMF tones? |
| 22:03.13 | b11d | i guess i dont follow what you mean.. |
| 22:03.49 | Dr-Linux|home | SimoAmi: are you looking for Playback() ? |
| 22:06.48 | perd | dr it will stop when the file you're playing stops, i think |
| 22:06.55 | perd | i havent really messed with lumenvox past the basics |
| 22:07.08 | perd | for long entries play silence/20 or something |
| 22:07.19 | SimoAmi | playback won't multitask. I want to play a background sound while (at the same time, while the caller is waiting) retrieving his info from a remote server |
| 22:07.32 | b11d | MusicOnHold? |
| 22:08.05 | perd | just play tt-monkeys |
| 22:08.14 | *** join/#asterisk backblue (n=moo@87-196-15-195.net.novis.pt) |
| 22:08.14 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:08.35 | b11d | or 'beep' |
| 22:08.37 | b11d | thats classic |
| 22:08.44 | SimoAmi | and I'm planning to use phpagi |
| 22:08.51 | b11d | I recorded a few seconds of me breathing heavily.. i play that over top of conversations at random.. |
| 22:08.57 | b11d | in a loop |
| 22:09.39 | SimoAmi | how do you do that? |
| 22:09.49 | b11d | umm.. well.. first off, I was joking. |
| 22:09.52 | b11d | So.. theres that. |
| 22:09.52 | perd | hahha b11d |
| 22:09.52 | *** join/#asterisk lorinc (n=ang@caracas-5029.adsl.interware.hu) |
| 22:09.57 | perd | that's uhh, creepy and sexy |
| 22:10.01 | b11d | haha |
| 22:10.13 | SimoAmi | :) |
| 22:10.21 | b11d | you could probably do it with Local channels though.. somehow |
| 22:10.32 | perd | that would be funny as hell |
| 22:10.35 | b11d | that reminds me, i need to check out these ".call" files |
| 22:10.46 | perd | app_heavypetting |
| 22:11.26 | b11d | lol |
| 22:11.59 | *** join/#asterisk max_______ (i=max__@ts.bestserversllc.net) |
| 22:13.08 | SimoAmi | this is how I want it: " Please hold while I access your information...(background processing sound) (5 seconds later) (background stops) I got your information...." |
| 22:13.19 | b11d | sounds like Music On Hold to me |
| 22:14.08 | perd | those background processing sounds irk me |
| 22:14.42 | b11d | I wish all call menu's were started with "press 1 if you're a moron, press 2 if you know whats up" |
| 22:14.47 | perd | if you have a cingular prepay phone you know what i mean... every time you enter a piece of information they play this god damn loud ass sound of keys beign pressed annoyingly, like a damn typewriter |
| 22:14.50 | b11d | and then each would be a totally seperate menu system |
| 22:15.34 | Dr-Linux|home | anybody ever have problem with transfering the call to the agi() ? |
| 22:16.40 | *** join/#asterisk wiljacket (n=wilson@cpe-76-173-243-4.socal.res.rr.com) |
| 22:16.48 | b11d | im sure someone has.. |
| 22:17.16 | SimoAmi | I tried MusicOnHold() but I can't get past that line. It seems that it's waiting for an Answer event |
| 22:18.03 | Dr-Linux|home | b11d: i'm trying to solve this problem since 3 weeks but no luck so far, looks like it's an asterisk bug in agi app |
| 22:18.29 | Qwell[] | Dr-Linux: "I've been trying [...] for 3 weeks [...]" |
| 22:18.33 | b11d | cant get past what line? |
| 22:18.48 | b11d | you just put someone on hold, and they get MoH.. take them off hold, its returned |
| 22:18.52 | b11d | you dont do it in the dialplan |
| 22:18.55 | b11d | you do it in sip.conf |
| 22:19.01 | b11d | err.. you dont do it in extensions.conf |
| 22:19.25 | b11d | Dr-Linux.. submit a bug report? ask in #asterisk-dev ? |
| 22:19.51 | Dr-Linux|home | Qwell[]: thanks :) |
| 22:20.15 | perd | anyone have a clue as to why my cisco 7902 doesnt get dialtone until i dial it from a different extension, then it wont hangup unless you hang up the phone twice |
| 22:20.17 | perd | :( |
| 22:20.27 | b11d | its not registering ? |
| 22:20.32 | perd | it's registering |
| 22:20.36 | b11d | are you sure? |
| 22:20.45 | perd | debugging shows it register |
| 22:20.48 | b11d | could it be a bad phone? |
| 22:20.49 | perd | here i'll pastebin.. |
| 22:20.52 | b11d | have you tried a couple? |
| 22:20.58 | b11d | no I believe you |
| 22:20.58 | perd | no it works if i put it on the CCM system |
| 22:21.05 | b11d | hrm... weird eh |
| 22:21.15 | perd | yeah.. the 7912 does the same thing |
| 22:21.21 | Qwell[] | sip? |
| 22:21.22 | b11d | I've only played with the 7940 and 7914.. so.. I dunno.. |
| 22:21.23 | b11d | they worked fine |
| 22:21.23 | robl^ | chan_skinny is a weird beast. |
| 22:21.26 | perd | chan_skinny |
| 22:21.34 | perd | i use sip for 7912/7960 now |
| 22:21.35 | Qwell[] | perd: send me pcap dumps from CCM |
| 22:21.52 | perd | i have the pcap i used to build my config |
| 22:21.55 | perd | one sec |
| 22:22.03 | perd | i nevermodified it |
| 22:22.08 | perd | it was some pcap file that was on the system |
| 22:22.17 | *** part/#asterisk entelechy (i=user@72.54.40.206) |
| 22:22.20 | perd | oh it's ptag.dat that i have |
| 22:23.13 | SimoAmi | this is an automated call made to the client, so when setting the moh on, no one is there to answer, because it's just automated |
| 22:23.31 | perd | how do i create a pcap dump? |
| 22:23.42 | Qwell[] | with ethereal/wireshark |
| 22:23.46 | perd | oooh |
| 22:23.56 | perd | i thought you meant something else ok ... one sec |
| 22:24.02 | perd | i'm slow. |
| 22:24.47 | Dr-Linux|home | Qwell[]: what should i send you my 7936 ? :) |
| 22:24.57 | Dr-Linux|home | errr |
| 22:25.00 | Qwell[] | Dr-Linux|home: you should :P |
| 22:25.03 | Dr-Linux|home | 7935 |
| 22:25.15 | Qwell[] | shipping costs would be ridiculous though |
| 22:25.16 | perd | ar downloading ethereal source |
| 22:25.56 | Dr-Linux|home | Qwell[]: and you will not send me back? :P |
| 22:26.14 | Qwell[] | shipping costs would be ridiculous ;) |
| 22:26.23 | Qwell[] | Dr-Linux|home: pcap dumps from CCM would be best |
| 22:26.56 | *** join/#asterisk entelechy (i=user@72.54.40.206) |
| 22:27.13 | monsted | Qwell[]: you need ccm pcaps? |
| 22:27.26 | Qwell[] | yeah |
| 22:27.32 | Dr-Linux|home | Qwell[]: can you guide me with that, i really don't understand that how can i get those dumps |
| 22:28.08 | Dr-Linux|home | Qwell[]: how can i get that for you? |
| 22:28.14 | Qwell[] | with ethereal/wireshark |
| 22:28.18 | Qwell[] | and call manager |
| 22:28.23 | monsted | Qwell[]: anything specific you want to see? |
| 22:29.06 | Qwell[] | monsted: incoming call (ccm > phone), outgoing call (phone > ccm), hangup initiated by phone, hangup initiated by ccm |
| 22:29.06 | Dr-Linux|home | monsted: what phone you have? |
| 22:29.10 | Qwell[] | for...various phones |
| 22:29.33 | *** join/#asterisk unsucht (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 22:29.44 | unsucht | is it possible to create a virtual channel with asterisk |
| 22:29.59 | monsted | Qwell[]: just signalling, right? |
| 22:30.08 | Dr-Linux|home | undrdawg: dummy channel? |
| 22:30.11 | Qwell[] | monsted: yeah, rtp is pretty much covered |
| 22:30.26 | Dr-Linux|home | monsted: what phone you have? |
| 22:30.44 | monsted | Dr-Linux|home: which one of them? ;) |
| 22:30.50 | Qwell[] | monsted: this is with skinny, obviously |
| 22:30.57 | monsted | Qwell[]: 'course |
| 22:30.57 | Qwell[] | in case that wasn't clear :) |
| 22:30.59 | perd | almost done with little ethereal |
| 22:31.19 | unsucht | i need something that will dial a user and play a message when a situation is met |
| 22:31.26 | monsted | Qwell[]: i'll see if i can pick up something that doesn't expose too much of our IP :) |
| 22:31.27 | *** join/#asterisk `Sean (i=Un1x@CPE000c148d127c-CM00140458831c.cpe.net.cable.rogers.com) |
| 22:31.36 | Dr-Linux|home | monsted: mine is cisco 7935 |
| 22:31.48 | Qwell[] | monsted: you should be able to capture just skinny packets going to/from a specific device |
| 22:32.13 | monsted | Dr-Linux|home: i have about 10000 cisco phones of various shapes and sizes out there |
| 22:32.26 | Qwell[] | monsted: does that mean I get to request which caps I want? :P |
| 22:32.45 | entelechy | monsted: got any 7940G's you wanna get rid of cheap? |
| 22:32.48 | monsted | Qwell[]: not really, i've got to find a place that doesn't carry user data :) |
| 22:32.53 | Qwell[] | 793x, 797x, 791x, 7920, 7085 ;) |
| 22:32.58 | Dr-Linux|home | monsted: wow, ever you got 7935 running with asterisk? |
| 22:33.11 | Qwell[] | Those would be ideal captures, heh |
| 22:33.14 | monsted | i only play with our left-over test 7960s |
| 22:33.29 | entelechy | anyone got 10 7940G 's they wanna donate, give away or otherwise sell cheap :) |
| 22:33.39 | Qwell[] | entelechy: I bet ebay does |
| 22:33.40 | Dr-Linux|home | monsted: my 7960s are just fine with sip firmware |
| 22:34.00 | monsted | entelechy: i'm sure cisco wants to sell some ;) |
| 22:34.11 | Dr-Linux|home | entelechy: i've alot, but why i'd sell if they are working fine |
| 22:35.16 | *** join/#asterisk foxxtrot (n=craig@c-67-185-0-172.hsd1.wa.comcast.net) |
| 22:36.13 | monsted | Qwell[]: you don't want 7960 dumps? those are the most prolific phones by far and the easiest cap to get |
| 22:36.22 | perd | qwell will a tcpdump -w work? |
| 22:36.29 | Qwell[] | monsted: sure, they'd help, but the rest would be best |
| 22:36.42 | Qwell[] | perd: umm...lemme see |
| 22:37.05 | wiljacket | I'm having a problem with 7940s running sip firmware where the local user hears a bad echo when dialing out through pstn (listening on the other side they sound great), soft-phone to 7940 causes the same echo on their end, but sip to sip between 7940s is ok. Anybody got an idea what this could be? |
| 22:37.06 | monsted | Qwell[]: i don't have any on our test system, but i'll snoop around and see if i can find something useful |
| 22:37.10 | Qwell[] | yeah, 0w should be good |
| 22:37.11 | Dr-Linux|home | Qwell[]: please also tell me the way to get dumps. i'll send you |
| 22:37.12 | Qwell[] | -w |
| 22:37.32 | Qwell[] | Dr-Linux|home: the problem is, they need to be from cisco call manager |
| 22:37.43 | perd | ok cool im gonna send you the udp data from this 7902 if that's ok |
| 22:37.45 | perd | it's 1.7k |
| 22:37.52 | Qwell[] | perd: qwell@digium.com |
| 22:37.58 | perd | ok cool |
| 22:38.35 | Dr-Linux|home | Qwell[]: i've cisco login and mybe i can download/install CCM , would that help? |
| 22:38.43 | Qwell[] | Dr-Linux|home: ccm isn't free, heh |
| 22:39.02 | terrapen | so, can you run two 4-port PRI cards in a 2-CPU opteron server if you're just translating B channels to IAX/G.729? |
| 22:39.02 | Dr-Linux|home | Qwell[]: i didn't said, it's free login :P |
| 22:39.04 | Qwell[] | BUT, if you can somehow "get a working copy"...it'd work fine |
| 22:39.19 | terrapen | and sending the calls to another * server? |
| 22:39.19 | Qwell[] | just need ethereal/pcap/whatever dumps |
| 22:39.41 | Dr-Linux|home | Qwell[]: we are paying for login |
| 22:39.50 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:40.17 | Dr-Linux|home | Qwell[]: i think i already downloaded the CMM |
| 22:40.37 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:41.05 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:41.09 | monsted | you can't download a full CCM, i believe |
| 22:41.18 | terrapen | hmmm |
| 22:41.24 | Dr-Linux|home | monsted: why? |
| 22:41.41 | monsted | and if you do, it contains a hardware check to make sure it's running on supported hardware |
| 22:41.53 | terrapen | i'm debating buying Juniper media gateways to go from PRI to SIP or just building my own with Asterisk and 4-port digium cards |
| 22:41.58 | monsted | (a few specific HP and IBM servers) |
| 22:42.03 | Dr-Linux|home | i see |
| 22:42.14 | monsted | terrapen: hardware gateways are quite nice |
| 22:42.16 | perd | ok qwell i just emailed the dump to you |
| 22:42.23 | terrapen | the hard part is finding 1U servers that are fast enough to handle the 4-port cards but use compactflash disks |
| 22:42.28 | perd | it has the initial tftp transfer, the skinny registration and then an attempted call |
| 22:42.36 | terrapen | monsted: that's what i've heard |
| 22:42.40 | monsted | terrapen: a cisco 5300 does the job nicely and has excellent debugging abilities |
| 22:42.44 | Qwell[] | perd: call both ways? Both would be great |
| 22:42.49 | terrapen | I like the idea of using an appliance for PRI<->SIP |
| 22:42.49 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:42.49 | perd | yeah |
| 22:43.01 | Qwell[] | cool |
| 22:43.04 | terrapen | monsted, I need 300-400 channels, so the 53xx won't cut it |
| 22:43.06 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:43.20 | terrapen | and the 54xx series is absolutely ridiculously priced, even without the required DSPs |
| 22:43.21 | perd | i did this: tried to make a call right after the registration, it failed. then i used another phone and called it, it rang the 7902 fine, i answered and hung up, then i dialed out from the cisco 7902 and it worked |
| 22:43.22 | monsted | terrapen: and you can probably pick one up for a low price since most ISPs are dropping them as dial-in platforms |
| 22:43.26 | perd | but it wouldnt hang up correctly |
| 22:43.37 | terrapen | monsted, how many PRIs per box? |
| 22:43.43 | perd | let me know if you need anything else |
| 22:43.48 | Qwell[] | perd: this dump is from ccm, right? |
| 22:43.53 | perd | oh shit actually this one doesnt have a successfull call |
| 22:43.56 | perd | no this is from asterisk |
| 22:43.57 | Qwell[] | heh |
| 22:44.00 | perd | doh |
| 22:44.01 | Qwell[] | I need it from ccm :) |
| 22:44.07 | perd | haha uhh ok hold on |
| 22:44.14 | Dr-Linux|home | Qwell[]: http://chan-sccp.sourceforge.net/pics.html :P |
| 22:44.26 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:44.40 | monsted | terrapen: i know that we have upwards of 500 5300s sitting around somewhere (not that we can get rid of them due to internal policies and such) |
| 22:44.41 | Dr-Linux|home | Qwell[]: it's same site as Sergio one? |
| 22:44.49 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:45.03 | monsted | terrapen: i know that they do 4 PRIs, but i would imagine they did more |
| 22:45.16 | terrapen | 4 PRIs per unit...and you have 500? wow |
| 22:45.22 | perd | gotta dl ethereal for windows |
| 22:45.26 | terrapen | do yours have PRI cards in them? |
| 22:45.33 | terrapen | i might just buy some off of you |
| 22:45.40 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:45.49 | monsted | terrapen: we had upwards of 3 million dial up accounts... |
| 22:45.51 | terrapen | I'll probably need around 16 of them |
| 22:46.04 | monsted | can't sell 'em to you, sorry |
| 22:46.07 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:46.14 | monsted | but other ISPs probably will |
| 22:46.17 | perd | im sure he wont mind taking them for free then |
| 22:46.52 | monsted | ok, 120 channels max on a 5300 |
| 22:47.00 | *** join/#asterisk xnon (n=xnon@200.82.223.85) |
| 22:47.21 | terrapen | 6 PRIs |
| 22:47.26 | monsted | 4 |
| 22:47.27 | terrapen | well 5 |
| 22:47.29 | monsted | E1 |
| 22:47.33 | terrapen | ahhh |
| 22:47.59 | monsted | 96 on those ghetto PRIs you 'merkins use |
| 22:48.14 | terrapen | hahah |
| 22:48.32 | monsted | :) |
| 22:48.34 | Dr-Linux|home | Qwell[]: looks like that chan_sccp is for someone else than Sergio :S |
| 22:48.37 | terrapen | I'm thinking two Juniper J6350s |
| 22:48.40 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:48.42 | perd | qwell, out of curiosity why do you want to see the ccm traffic? |
| 22:48.46 | terrapen | but that would probably run almost US$60,000 |
| 22:48.51 | monsted | terrapen: ouch |
| 22:49.14 | terrapen | i wish someone would make a nice 1U opteron server that had a built-in CompactFlash slot |
| 22:49.18 | monsted | terrapen: you could possibly buy a catalyst 6500 and a 6608 gateway blade for that |
| 22:49.25 | terrapen | because I would run * and some 4-port digiums |
| 22:49.56 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:49.57 | monsted | terrapen: just use a CF-to-IDE adapter and stick it directly in the IDE connector |
| 22:50.22 | terrapen | I think we're gonna get fiber to our premises and get that broken out into a bunch of PRIs and also some ethernet |
| 22:50.26 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:50.29 | Qwell[] | perd: so I can fix chan_skinny |
| 22:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:50.46 | terrapen | monsted, the problem is, i've had trouble finding servers that have room for that aadapter |
| 22:50.56 | terrapen | monsted, we tried it with the Sun X2100s, didn't fit |
| 22:51.00 | monsted | i want an STM1 and speak SS7 instead of fiddling with these damn PRIs |
| 22:51.12 | terrapen | what is the deal with SS7 |
| 22:51.13 | Qwell[] | Dr-Linux|home: yeah, that's Sergios site. |
| 22:51.15 | monsted | terrapen: a cable and some duct tape then? :) |
| 22:51.28 | Qwell[] | or, wait, maybe not |
| 22:51.40 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:51.46 | terrapen | heh, this is why i'll probably buy the junipers |
| 22:51.58 | monsted | terrapen: SS7 is the backbone of the phone system - in this case i'd get a 155 Mbps fibre connection and just act like any other telephone switch |
| 22:52.00 | terrapen | our call center is crucial to our business |
| 22:52.00 | *** join/#asterisk TheAsp (n=asp@blk-7-162-225.eastlink.ca) |
| 22:52.19 | terrapen | monsted, i only need about 300 total voice channels |
| 22:52.27 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:52.50 | *** join/#asterisk sjobeck (n=sjobeck@70.89.186.65) |
| 22:52.51 | terrapen | i have XO coming out tomorrow to talk PRI/T3/etc |
| 22:52.51 | monsted | it would be awesome to just tell the network that i had a pile of DIDs and not have to care about how many phone lines people actually had |
| 22:53.09 | Dr-Linux|home | Qwell[]: i think sergio never got some random donations :P |
| 22:53.09 | terrapen | seriosly |
| 22:53.15 | Qwell[] | Dr-Linux|home: no comment |
| 22:53.19 | Qwell[] | ;) |
| 22:53.22 | *** join/#asterisk sjobeck (n=sjobeck@70.89.186.65) |
| 22:53.24 | terrapen | monsted, I would love to pay only for the channels that we use |
| 22:53.31 | Qwell[] | but, my feelings on that matter were made very clear in the past |
| 22:53.36 | terrapen | most of the time we will use about 100-150 channels |
| 22:53.50 | monsted | one customer had a third PRI added today and immediately went from saturating 60 channels to saturating 90... two more PRIs are on their way :) |
| 22:53.54 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:53.54 | terrapen | but let us scale up to 400 channels on the same piece of fiber |
| 22:53.58 | Dr-Linux|home | :P |
| 22:54.05 | terrapen | and not charge us for a bunch of PRIs that we won't use very often |
| 22:54.22 | monsted | terrapen: well, get something bigger than PRI or do it over IP then :) |
| 22:54.27 | terrapen | is there a better solution for us than a shiatload of PRIs? |
| 22:55.42 | terrapen | yeah, I suppose we could have XO provide SIP over that fiber to us |
| 22:55.42 | terrapen | i wonder how the quality would be |
| 22:55.42 | monsted | same |
| 22:55.42 | terrapen | let THEM buy the ciscoes and junipers |
| 22:55.42 | terrapen | i'm sure we'd pay "far out the ass for it" |
| 22:55.42 | monsted | all of our customers are using g711 - sounds just like an ISDN call |
| 22:55.56 | terrapen | we'd need at least 30Mbps for that |
| 22:56.26 | terrapen | i'm sure XO will offer multiple ways of doing this |
| 22:56.37 | terrapen | i hope that we can get them to justify a fiber build |
| 22:56.44 | terrapen | currently have no fiber to our callcenter/warehouse |
| 22:56.47 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:56.53 | Charles[NS] | hello quelqu'un aurrait un modele pour 2 context incoming sur asterisk |
| 22:57.14 | Charles[NS] | pour gerer 2 entreprises |
| 22:57.18 | monsted | i've got four GigE links to the MPLS backbone and have only ever peaked 40 Mbps - i'll live for some time yet :) |
| 22:57.43 | Charles[NS] | j'ai fais plusieurs essais de conf et j'aimerai savoir l'astuce |
| 22:57.44 | Qwell[] | Strom_C: lol@cake |
| 22:57.51 | Strom_C | :D |
| 22:57.54 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:58.05 | Strom_C | ill just eat twice as much cake when i'm in huntsville in marchish |
| 22:58.10 | Qwell[] | works |
| 22:58.27 | TheAsp | i'm looking at getting a sipura 3000, is there anything similar i should be looking at instead? |
| 22:58.57 | Charles[NS] | someone have ever make ipbx for 2 company on the same asterisk ? |
| 22:59.14 | *** join/#asterisk unsucht (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 22:59.16 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net) |
| 22:59.40 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 22:59.44 | *** join/#asterisk linlin (n=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 22:59.46 | TripleFFFF | how much calls can a p3 1000 hold ? |
| 22:59.55 | b11d | depends on the codec |
| 22:59.58 | unsucht | i can't seem to get channel_status working, i've tried all different types of ways to enter the channel but asterisk always says there is no such channel |
| 23:00.04 | Dr-Linux|home | TripleFFFF: RAM? |
| 23:00.29 | TripleFFFF | 512 ? |
| 23:01.40 | b11d | depends on the codec |
| 23:01.57 | b11d | tell us what one and then you can figure out how much ram you need and how many calls she'll handle |
| 23:02.52 | b11d | goodnight all |
| 23:04.04 | Qwell[] | TheAsp: 3102 or whatever it is |
| 23:04.21 | TripleFFFF | ulaw |
| 23:04.21 | TheAsp | Qwell: is that the one with the router? |
| 23:04.33 | Qwell[] | dunno, I just see people recommending that one |
| 23:05.12 | *** join/#asterisk CunningPike_ (n=CunningP@dhcp-10-153.district.north-van.bc.ca) |
| 23:05.32 | perd | fix chan skinny! wee |
| 23:06.20 | Qwell[] | perd: let me know when you send those new dumps, so I don't lose them |
| 23:06.44 | Strom_C | Qwell[]: eat cake for me at the party :) |
| 23:06.51 | Qwell[] | likely not going |
| 23:07.12 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 23:07.20 | Strom_C | ?! |
| 23:07.25 | Qwell[] | wow |
| 23:07.27 | Strom_C | how can you not go for FREE CAKE? |
| 23:07.28 | Qwell[] | "The eight-to-one Medimmune v. Genetech decision, written by Justice Scalia, held that by paying royalties to a patent holder, one does not necessarily waive the right to challenge the validity of the patent." |
| 23:07.42 | Qwell[] | that's pretty huge |
| 23:07.58 | Strom_C | yeah, that is |
| 23:08.24 | Druken | anyone have access to the ajax ratecenter in ontario ? |
| 23:09.16 | rudholm | I have a noob AGI question |
| 23:09.21 | rudholm | really really simple thing I want to do |
| 23:09.35 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:09.38 | Qwell[] | Druken: Colgate-Palmolive is based in NYC |
| 23:09.41 | rudholm | I want to send a call to a phone number that is defined by the output of a PERL script |
| 23:09.53 | *** join/#asterisk Jay97232 (n=jay@jayallen.dsl.pdx.spiretech.com) |
| 23:10.01 | EmleyMoor | Can I use the Voice Mails count in ekiga when it is connected to asterisk? If so, how? |
| 23:10.07 | EmleyMoor | (SIP, not OH323) |
| 23:10.09 | rudholm | I've tried just creating a little perl script that sends a Dial(blahblah) to STDOUT, but that doesn't seem to do it |
| 23:10.10 | De_Mon | what do you call a hardware device that translates UDP SIP <-> TCP SIP? |
| 23:10.22 | *** join/#asterisk karmatronic (n=karmatro@84.77.155.231) |
| 23:10.24 | Qwell[] | De_Mon: pointless |
| 23:10.28 | rudholm | Qwell[]: I'm looking at you :) |
| 23:10.28 | Jay97232 | ekiga works fine as a sip phone with asterisk |
| 23:10.34 | Druken | Qwell[]: i'll add that to my bank of useless information |
| 23:10.35 | Qwell[] | rudholm: from outside? |
| 23:10.39 | Qwell[] | because that would be creepy |
| 23:10.43 | De_Mon | I cant seem to find anything |
| 23:10.50 | EmleyMoor | Jay97232: How do I get the voicemail count to work? |
| 23:10.54 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:10.56 | Strom_C | Qwell[]: i set up a secret webcam in your phone when i was repairing it |
| 23:11.04 | rudholm | Qwell[]: I'm not sure what you mean by "from outside" |
| 23:11.06 | Qwell[] | Strom_C: it's at home, unplugged :P |
| 23:11.09 | Jay97232 | unknown, I've not tried that |
| 23:11.14 | Strom_C | there goes my joke |
| 23:11.25 | Jay97232 | any ABE users here? |
| 23:11.45 | Qwell[] | Jay97232: if it's a support question, you'll need to call Digium support |
| 23:11.45 | perd | ok qwell i sent you the dump from CCM |
| 23:11.53 | Jay97232 | I'm wondering if I re-register my ABE license, will it kill the old server... |
| 23:11.57 | perd | i dialed to and from it, letting the phone ring once each time |
| 23:11.59 | EmleyMoor | I have the stuttered dialtone working on my Zap phones now :-) |
| 23:12.03 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net) |
| 23:12.11 | Qwell[] | perd: answer is important too. Sorry, I should've been clear |
| 23:12.16 | perd | darn |
| 23:12.21 | perd | ok let me make another one |
| 23:12.28 | perd | atl east now i know how the software works |
| 23:12.41 | rudholm | Qwell[]: basically, I have an inbound call (from outside) that I want to redirect to one of several destination numbers (off-asterisk) based on, say, time of day or whatever. so I need the PERL script to be able to define the Dial() command parameters. |
| 23:12.42 | Qwell[] | and if possible, I'd like to see each side hang up one of the calls |
| 23:12.55 | Qwell[] | doesn't matter which, just on one of them |
| 23:12.57 | perd | ok i'll make two calls from both sides |
| 23:13.05 | Qwell[] | only need one from each side |
| 23:13.11 | perd | do you want the data from the phone i call as well? |
| 23:13.16 | Qwell[] | yeah |
| 23:13.18 | perd | ok |
| 23:14.47 | Qwell[] | perd: this is perfect otherwise |
| 23:17.54 | *** join/#asterisk DrCron (n=rszasz@c-24-7-33-87.hsd1.ca.comcast.net) |
| 23:20.35 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:20.58 | Aurs | cdr_odbc question: if i have a column named foo in my cdr-table. will CDR(foo) be saved to this column? |
| 23:21.15 | Aurs | or does that require changes to cdr_odbc.c? |
| 23:21.43 | perd | ok qwell! i just sent you the dump with dialing |
| 23:21.59 | perd | i made two calls, the first call i hung up on the remote phone after saying 'test' |
| 23:22.06 | perd | the second call i hung up on the 7902 after asying 'test' |
| 23:22.37 | perd | anything else you need just let me know |
| 23:22.46 | perd | i'm more than happy to make dumps if it's goign to help |
| 23:22.51 | Jay97232 | Qwell, so there is no ABE support on IRC? |
| 23:23.11 | Qwell[] | Jay97232: well, you're paying for support |
| 23:23.21 | Qwell[] | and there are some things we simply can't help you with |
| 23:24.20 | perd | damn this 7902, i am going to have to set up a second 1.2 asterisk system for chan_sccpif i cant get this thing working heh |
| 23:24.45 | perd | it's a shame the 7902 doesnt use the same firmware as the 7912 |
| 23:26.20 | perd | i just like phones that i dont have to torture myself to get working |
| 23:26.41 | perd | the 7912 was cake, the 7960 a pain in the ass and the 7902 just plain old doesnt work :) |
| 23:28.18 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-24-234.red.bezeqint.net) |
| 23:29.44 | monsted | it's nice to just pick up a handset and dial instead of fiddling with software... dunno why, but it just feels right |
| 23:30.07 | perd | holy shit /me overload |
| 23:30.46 | perd | monsted no lie |
| 23:30.47 | monsted | EmleyMoor: take a hint from someone with experience: pull twice as many cables to twice as many places as you think you'll need - you WILL run out |
| 23:30.52 | perd | the sip stuff is very nice though |
| 23:31.16 | monsted | EmleyMoor: that works too :) |
| 23:31.39 | monsted | zap? |
| 23:31.40 | EmleyMoor | I like my 746 |
| 23:32.02 | EmleyMoor | monsted: A connection to an FXS port |
| 23:32.53 | monsted | why not just pull Cat5e and plug phones into the RJ45 ports, FXS or ethernet as required? |
| 23:33.27 | EmleyMoor | That is somewhat my plan - but I have to mount my NTEs somewhere |
| 23:33.29 | monsted | or am i missing something? :) |
| 23:33.44 | codefreeze | JUST WONDERING: RANDOM POLL: of the 298 people here, has ANYONE ever used ForkCDR to your satisfaction? Has ANY of you EVER even tried to use it? |
| 23:33.59 | Qwell[] | </crickets> |
| 23:34.00 | monsted | mount all the ugly stuff in the wiring closet, ofcourse |
| 23:34.26 | EmleyMoor | The phone plugs here are different too |
| 23:34.41 | ManxPower | Since we don't bill for calls, there is no need to do ANY CDR stuff. |
| 23:35.23 | monsted | EmleyMoor: so are ours, but you can get rj45-to-phone adapter cables for next to nothing |
| 23:35.32 | EmleyMoor | monsted: Where are you? |
| 23:35.32 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 23:35.32 | monsted | .dk |
| 23:35.59 | EmleyMoor | My wiring here is "right but slack" |
| 23:36.00 | Marty-OTT | I MA BACK! |
| 23:36.05 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:36.08 | EmleyMoor | I will tighten it at some suitable point |
| 23:36.16 | Marty-OTT | My call blocking problem is LICKED! |
| 23:36.18 | EmleyMoor | Marty-OTT: Are uoy really? |
| 23:36.25 | Marty-OTT | Time to do Call Forwarding and CAll Return |
| 23:36.33 | Marty-OTT | EmleyMoor: Not really |
| 23:36.49 | Marty-OTT | Pizza's on it's way .. in my EST |
| 23:37.03 | codefreeze | ManxPower: understood... that's logical... I hope... |
| 23:39.24 | [TK]D-Fender | Marty-OTT : Call return is a VERY difficult proposition... |
| 23:40.18 | Strom_C | pfft, it's simple |
| 23:40.28 | Marty-OTT | TK: huh? |
| 23:40.37 | Strom_C | put the number in the DB on the last inbound call, look it up when dialing *69 |
| 23:40.48 | Marty-OTT | that's what I was thinking... |
| 23:40.56 | Strom_C | 8-20 lines of dialplan depending on how fancy you want to get |
| 23:41.14 | Marty-OTT | well, I'm doing Call Forward first right now |
| 23:41.18 | Marty-OTT | anyways |
| 23:41.53 | [TK]D-Fender | Strom_C : Maybe I'm missing the right name for the one where if they number is busy it'll keep calling till its not... |
| 23:42.00 | Strom_C | no, that's repeat dial |
| 23:42.01 | Strom_C | *66 |
| 23:42.04 | Strom_C | call return is *69 |
| 23:42.08 | [TK]D-Fender | *666! |
| 23:42.21 | Marty-OTT | oh! ok, I'll do that WAYYY last |
| 23:42.22 | Marty-OTT | lol |
| 23:42.25 | [TK]D-Fender | Marty-OTT : yeah, scratch that! *69 = Easy, *66 = bitch |
| 23:42.25 | Strom_C | and *66 would be easy too; call files and some AGI magic |
| 23:42.55 | Marty-OTT | Actually, once I have half a dozen basic features done, I want to make one where people can dial for the weather.. it would be REALLY neat |
| 23:43.02 | [TK]D-Fender | Strom_C : Cahnnel type dependant :) If you have an analog card, lack of progress will screw you :) |
| 23:43.05 | Marty-OTT | *55 "Welcome to the Weather NEtwork" |
| 23:43.15 | [TK]D-Fender | Strom_C : in many cases even if you DO, heh |
| 23:43.20 | Strom_C | [TK]D-Fender: well of course, but what kind of nudnik would do this on an analog card? :) |
| 23:43.20 | Marty-OTT | "Dial 100 for Ottawa, 101 for Toronto, 102 for Miami" |
| 23:43.22 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 23:43.38 | [TK]D-Fender | Strom_C : Careful.. you'll offend %90 of this channel! |
| 23:43.41 | Marty-OTT | or... "Enter the city name using your keybad" |
| 23:43.43 | Grnd-Wire | Marty-OTT: Better yet.. Prompt them for the zip code of the area.. |
| 23:43.54 | [TK]D-Fender | mmm KEY BAD! |
| 23:44.01 | Strom_C | Marty-OTT: that already exists. it's called 800-555-TELL |
| 23:44.11 | Grnd-Wire | Marty-OTT: Kinda like the *61 Weather functionality in TrixBox does.. :P |
| 23:44.24 | Strom_C | *61? |
| 23:44.29 | Marty-OTT | Oh shit.. YEAH!! That's good or maybe just the airprot.. yes.. but people don't know what.. but yo ujust gave me a GREAT idea.. I'll code a *55 to forward to that number!!!! LOL!!! |
| 23:44.46 | Marty-OTT | What's *61? |
| 23:45.02 | Strom_C | *61 is already reserved for Distinctive Ringing/Call Waiting Activation |
| 23:45.10 | Marty-OTT | ok |
| 23:45.26 | Grnd-Wire | Strom_C: ok, well it doesn't prompt.. but my default TrixBox has that in it! |
| 23:45.36 | Strom_C | Marty-OTT: *55 is already reserved for Single Line Variety Package (SVP) - Distinctive Ring D |
| 23:45.37 | Marty-OTT | *69: "Hello There, how may I help you tonight big boy?" |
| 23:45.48 | Strom_C | trixbox can go play with a nut |
| 23:45.50 | [TK]D-Fender | Strom_C : Screw Ma's plan! |
| 23:46.04 | Marty-OTT | great... well,... if I ever decide to pick extensions... I'll look up the reseved first |
| 23:46.05 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
| 23:46.15 | Strom_C | Marty-OTT: http://nanpa.com/number_resource_info/vsc_assignments.html |
| 23:46.27 | Qwell[] | ~vsc |
| 23:46.32 | Qwell[] | stupid bot |
| 23:46.42 | Qwell[] | jbot: vsc is Vertical Service Codes |
| 23:46.44 | jbot | okay, Qwell[] |
| 23:46.44 | Strom_C | qwell: i was just checking to see if jbot knows about vsc |
| 23:47.09 | Marty-OTT | You know what would be great (bookmarked - thanks)... design a perfect Astrisk box, have a programmer write it for VxWorks with full Web Manageability and call it: ASTROID (Asterisk on Steroids) |
| 23:48.08 | [TK]D-Fender | jbot: no, vsc is Vertical Service Codes - http://nanpa.com/number_resource_info/vsc_assignments.html |
| 23:48.14 | jbot | [TK]D-Fender: okay |
| 23:48.15 | *** join/#asterisk Cunk (n=chatzill@pool-70-109-139-114.cncdnh.east.verizon.net) |
| 23:48.15 | Strom_C | jbot: no, vsc is Vertical Service Codes - these codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html |
| 23:48.18 | jbot | okay, Strom_C |
| 23:48.18 | dlynes_laptop | Marty-OTT, pipe dream :) |
| 23:48.42 | [TK]D-Fender | Strom_C : Horse by commitee! Lets start on the "buggy" now ;) |
| 23:48.50 | Strom_C | hahah |
| 23:49.15 | mercestes | can I play with the bot? |
| 23:49.25 | [TK]D-Fender | mercestes : back to your room! |
| 23:49.26 | Marty-OTT | dlynes: Every journey begins with a FIRST step... maybe Mr. and Mrs Bosack were being told the same thing. |
| 23:49.40 | dlynes_laptop | whoever mr. and mrs. bosack are :) |
| 23:49.46 | Marty-OTT | .. :P |
| 23:49.50 | Qwell[] | Strom_C: there are a ton of those things... |
| 23:49.52 | *** part/#asterisk oQPa (n=uawename@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 23:49.55 | [TK]D-Fender | Marty-OTT : In that case we'll make a shorter plank for you ;) |
| 23:49.55 | Marty-OTT | The creators of CISCO |
| 23:49.58 | Strom_C | Qwell[]: VSCs? |
| 23:50.01 | Marty-OTT | lol!! |
| 23:50.01 | Qwell[] | yeah |
| 23:50.05 | dlynes_laptop | ah |
| 23:50.13 | Strom_C | Qwell[]: note the six at the end that are "reserved for local assignment" |
| 23:50.15 | Qwell[] | are there are reservered? heh |
| 23:50.18 | Qwell[] | ahh |
| 23:50.19 | dlynes_laptop | I think I agree with [TK]D-Fender on that one |
| 23:50.25 | codefreeze | jbot: send me a muffin! |
| 23:50.27 | jbot | Go (.*?), codefreeze |
| 23:50.32 | Qwell[] | figured "local assignment" meant the local telco, but I guess that could work |
| 23:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:50.42 | Marty-OTT | Believe it or not, Mrs Bosack (divorced) actually retired with her billions on a farm (with tons of animals) in a small shack. |
| 23:50.46 | codefreeze | jbot: you are NOT nice! |
| 23:50.47 | jbot | codefreeze: what are you talking about? |
| 23:50.52 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 23:51.03 | [TK]D-Fender | codefreeze : how is that supposed to be read? |
| 23:51.04 | dlynes_laptop | Marty-OTT, she should take cisco and throw in that small shack, too |
| 23:51.12 | Marty-OTT | lol!! |
| 23:51.16 | dlynes_laptop | Marty-OTT, that crap's way too expensive |
| 23:51.20 | Marty-OTT | ebay |
| 23:51.38 | Marty-OTT | but brand new ... I agree. I'd rather buy HP Switches - less expensive |
| 23:51.43 | dlynes_laptop | Marty-OTT, and judging by all the complaints I've heard in here about the sip firmware, the price is totally unwarranted, too |
| 23:52.00 | codefreeze | [TK]D-Fender: with a grain of salt! |
| 23:52.01 | Marty-OTT | Oh, I didn't talke about doing voice on a Cisco router... heh heh |
| 23:52.14 | Marty-OTT | I mean... doing SIP to clarify |
| 23:52.29 | [TK]D-Fender | dlynes_laptop : Cisco is like Apple. The hardware is great, just that it you don't like it "Their Way", you know what you can do with it... |
| 23:52.37 | dlynes_laptop | hehehe |
| 23:52.38 | [TK]D-Fender | dlynes_laptop : AFTER forking over way too much :) |
| 23:52.45 | codefreeze | jbot: how smart are you? |
| 23:52.47 | jbot | I think you lost me on that one, codefreeze |
| 23:52.47 | Qwell[] | Strom_C: how are these implemented? like...do all telcos implement all, some subset, or? |
| 23:52.57 | [TK]D-Fender | codefreeze : More like a PILLAR |
| 23:52.57 | Qwell[] | is it completely telco dependent? |
| 23:53.09 | Strom_C | Qwell[]: they're generally implemented based on the services offered by the telco |
| 23:53.10 | [TK]D-Fender | ~jbot |
| 23:53.11 | jbot | [jbot] only marginally useful at best, He got a C- on his Turing Test |
| 23:53.16 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:53.16 | [TK]D-Fender | see!? |
| 23:53.50 | codefreeze | well, at least jbot's honest! |
| 23:54.57 | robl^ | Strom_C: "Hideaway"? |
| 23:55.12 | rudholm | Strom_C: ship of fools? |
| 23:55.13 | Strom_C | no, i dont have that one |
| 23:55.18 | Strom_C | "chains of love" |
| 23:55.22 | *** join/#asterisk newbie22 (n=newbiew@jffwpr02.jf.intel.com) |
| 23:57.36 | newbie22 | hello |
| 23:57.47 | codefreeze | ~vsc |
| 23:57.55 | jbot | extra, extra, read all about it, vsc is Vertical Service Codes - these codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html |
| 23:59.01 | codefreeze | jbot: codefreeze is the most wonderful guy you'd ever like to meet! |
| 23:59.03 | jbot | codefreeze: okay |
| 23:59.14 | *** join/#asterisk infoeng2006 (n=chatzill@adsl-208-191-147-176.dsl.hstntx.swbell.net) |
| 23:59.29 | codefreeze | jbot: codefreeze is an alias for Steve Murphy |
| 23:59.30 | jbot | ...but codefreeze is already something else... |
| 23:59.54 | codefreeze | jbot: codefreeze is also an alias for Steve Murphy |
| 23:59.56 | jbot | codefreeze: okay |