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| 03:35.25 | Micc_ | so far so good with asterisk 1.8.5.0 after fixing a couple things like multi-tenant parking. Tomarrow is the real test though. |
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| 03:42.35 | shmaltz | hi everyone |
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| 04:59.18 | _R0N_ | does anyone see what's wrong with this? ExecIf($[${CHARGE} = YES]?GET_upddebit(${CALL},${CHANNEL},${UNIQUEID},${CDR(billsec)})) |
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| 05:16.47 | kaldemar | _RON_: missing "" around ${CHARGE} and YES or GET_upddebit |
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| 05:54.01 | schmidts | good morning |
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| 06:05.52 | kleszcz | morning |
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| 06:12.56 | wdoekes2 | morning |
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| 06:20.49 | ChannelZ | blah! |
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| 06:24.57 | nny1 | if anyone is up or around, trying to figure out why queue_log stopped logging all of a sudden last week. Last entry was on the 13th and then... no more. Any advice for troubleshooting appreciated |
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| 06:27.15 | Babak__ | can someone help with cisco xmo object, CiscoIpPhoneStatus? |
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| 06:46.23 | singler | nny1: did you try to restart asterisk? |
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| 08:55.04 | bipul | p3nguin: ping |
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| 10:12.50 | wasanzy | hi p3nguin |
| 10:13.02 | wasanzy | hope you are doing well |
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| 10:24.51 | Bipul | p3nguin, ping |
| 10:29.11 | Bipul | any one around |
| 10:29.16 | benno2 | anyone using a junghanns duobri or quadbri card here ? I followed the install pdf on their site (using debian lenny) install.sh works correctly. bristuff installed. but then the doc says to copy zapata.conf.duoBRI to /etc/asterisk but I don't have the file in the bristuff tree I downloaded (0.4.0-RC16). any idea ? |
| 10:30.16 | benno2 | are the junghanns people hanging around here on irc or better to call them on the phone? |
| 10:39.40 | Bipul | i have fix present IP of asterisk system in my z0iper softphone but when i see on CLI mode the IP is diffent |
| 10:40.29 | Bipul | i have fix present IP of asterisk system in my z0iper softphone but when i see on CLI mode <system runing Asterisk>the IP is diffent but ip belongs to my isp any one any idea |
| 10:40.57 | *** join/#asterisk hrolf (~hrolf@202.61.49.9) |
| 10:42.11 | kaldemar | Bipul: which IP? |
| 10:42.30 | kaldemar | Bipul: pastebin a trace and point out what bothers you. |
| 10:42.42 | Bipul | am the IP i have fix in my zoiper is totally diffrent when i see in my asterisk PBX |
| 10:42.48 | Bipul | ok |
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| 10:45.08 | Bipul | http://pastebin.com/BNxv9Vep this IP is 113.19.130.55 not mine |
| 10:46.22 | Bipul | i have fixed this IP at my zoiper 113.19.128.194 |
| 10:46.41 | Bipul | which is the IP of a system runing asterisk |
| 10:48.05 | kaldemar | pastebin a sip debug |
| 10:48.20 | kaldemar | of registration messages for example. |
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| 10:49.10 | Bipul | For example? |
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| 10:51.05 | techknowlust | it is possible to have asterisk run a macro when a queue'd call has ended ? |
| 10:51.21 | Bipul | kaldemar, ? |
| 10:51.30 | techknowlust | basically I have a macro I want to run at the start and end of every call |
| 10:51.47 | *** join/#asterisk nightrid3r (~nightrid3@91.176.182.178) |
| 10:52.10 | Bipul | nightrid3r, there is an issue |
| 10:52.18 | kaldemar | Bipul: pastebin a sip rebug of a registration. |
| 10:53.09 | *** join/#asterisk Sunil (~sunil@115.113.211.130) |
| 10:53.39 | kaldemar | techknowlust: yes. hangup extension is one way. |
| 10:55.36 | nightrid3r | hi |
| 10:57.10 | techknowlust | kaldemar: the problem here https://img.skitch.com/20110718-jbiq8p8ch61ij81krudag9f15e.png is that the Set and subsequent CURL aren't being run |
| 11:00.14 | Bipul | kaldemar, http://pastebin.com/QnjNmPGx |
| 11:01.21 | Bipul | i have hide the password and ID |
| 11:02.21 | kaldemar | Bipul: and the sip debug? that is sip.conf. i asked for sip debug that is enabled with "sip set debug on" in CLI. then make zoiper register again. |
| 11:03.08 | *** part/#asterisk Sunil (~sunil@115.113.211.130) |
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| 11:04.10 | Bipul | i kaldemar idont know how to know sip debug |
| 11:06.01 | kaldemar | Bipul: shut down zoiper, go to CLI, give command "sip set debug on", start zoiper, pastebin what you see in CLI. |
| 11:06.10 | Bipul | ok |
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| 11:07.44 | techknowlust | what variables are there for the agent that answers a queue call ? |
| 11:08.33 | techknowlust | does EXTEN change when a call is connected to an agent ? |
| 11:12.19 | kwk | Can anyone recommend a good AMI proxy? |
| 11:12.36 | kwk | for asterisk 1.8 |
| 11:13.12 | Bipul | http://pastebin.com/cyUJ8fX3 |
| 11:13.19 | Bipul | kaldemar, http://pastebin.com/cyUJ8fX3 |
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| 11:17.00 | kaldemar | Bipul: the packets are coming from 113.19.130.55 indeed. and packets sent to 113.19.130.55 port 5060 go to your zoiper since it answers too. |
| 11:18.39 | Bipul | in zopiper i have fixed this IP 113.19.128.194 |
| 11:21.00 | kaldemar | Bipul: and it is inside the messages. why have you configured that address in zoiper? |
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| 11:24.21 | techknowlust | is the only way to execute a macro on Hangup to use the AGI ? |
| 11:25.07 | kaldemar | techknowlust: no, use hangup extension. "exten => h,1,..." |
| 11:25.36 | techknowlust | awesome. just saw that |
| 11:25.55 | techknowlust | do you know also if the EXTEN variable changes when you're connected in a queue ? |
| 11:26.53 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 11:27.36 | kaldemar | it only changes when the current extension in the dialplan changes. |
| 11:28.04 | techknowlust | is there any variable that will give me the extension of the current connected agent L |
| 11:28.14 | techknowlust | I'm trying to use it in the Queue macro callback |
| 11:28.25 | kaldemar | extension as in the dialplan extension of the agent? |
| 11:28.30 | Bipul | kaldemar, what about my problem |
| 11:28.42 | techknowlust | yep |
| 11:28.54 | kaldemar | Bipul: what problem? what about my question? |
| 11:29.16 | Bipul | sory i got disconnected what is you Question ? |
| 11:29.45 | kaldemar | techknowlust: you mean the agent that the channel was connected to, after the call has exited the queue app? |
| 11:30.00 | kaldemar | Bipul: "Bipul: and it is inside the messages. why have you configured that address in zoiper?" |
| 11:30.16 | *** join/#asterisk esperanto (~rusty@ip-178-202-18-0.unitymediagroup.de) |
| 11:30.31 | Bipul | beaucuse my IP is dynamic i have to configure |
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| 11:31.48 | kaldemar | Bipul: thats precisely when you should NOT configure an address like that. |
| 11:32.19 | Bipul | ok just tell me the way to configure |
| 11:32.42 | kaldemar | i just did. do not configure an ip address in zoiper. |
| 11:33.00 | Bipul | then what to configure? |
| 11:33.09 | kaldemar | what? |
| 11:33.28 | Bipul | nuthing you said not to configure IP adress |
| 11:33.29 | kaldemar | it is configured already because it registers to asterisk. |
| 11:33.32 | Bipul | in zoiper |
| 11:33.37 | Bipul | ok |
| 11:34.11 | kaldemar | remove the address that you configured for the zoiper client. leave everything else as they are now. |
| 11:34.37 | Bipul | yes i have fixed asterik insted of IP |
| 11:35.16 | Bipul | pul-desktop*CLI> sip show peers |
| 11:35.17 | Bipul | Name/username Host Dyn Forcerport ACL Port Status |
| 11:35.17 | Bipul | bipul/bipul (Unspecified) D 0 Unmonitored |
| 11:35.17 | Bipul | voipon/105696_92 78.129.153.20 A 5060 Unmonitored |
| 11:35.17 | Bipul | 2 sip peers [Monitored: 0 online, 0 offline Unmon |
| 11:35.31 | kaldemar | i don't understand you. |
| 11:37.04 | Bipul | kaldemar, i have open zoiper account |
| 11:37.14 | Bipul | and i have removed the IP |
| 11:37.22 | Bipul | from account. |
| 11:42.07 | Bipul | i dont understand what's wrong with my asterisk :( |
| 11:42.16 | Bipul | tomorrow it was working fine |
| 11:42.48 | SunTsu | Bipul: so it resolves itself? ;) |
| 11:42.57 | kaldemar | Bipul: you have not shown anything to be wrong with it. |
| 11:43.08 | Bipul | so why i cant make a call |
| 11:43.44 | pabelanger | ~collectdebug |
| 11:43.44 | infobot | i heard collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 11:43.49 | pabelanger | Bipul: ^ |
| 11:43.59 | Bipul | pabelanger, yes |
| 11:44.14 | pabelanger | pb the debug log |
| 11:44.46 | techknowlust | does anyone know if I can pass variables in the member macro option for queues ? |
| 11:46.52 | kaldemar | Bipul: configure the IP address of asterisk in zoiper. |
| 11:48.12 | Bipul | http://pastebin.com/d7pvNfbf logger.conf |
| 11:50.06 | wasanzy | kaldemar: how are you? |
| 11:50.19 | kaldemar | wasanzy: fine thank you. |
| 11:50.37 | wasanzy | I did every thing I could but my problem still exist |
| 11:51.52 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:51.56 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:51.58 | wasanzy | now, when two people call each other using twinkle, only one of them could hear his partner talk but the other partner doesn't hear |
| 11:52.50 | Bipul | yes but not when i am calling it says bearerecapability notauth |
| 11:53.18 | wasanzy | I followed this: http://www.freepbx.org/support/documentation/howtos/howto-resolving-audio-problems but still, it is not solve |
| 11:53.39 | kaldemar | Bipul: what says? when? |
| 11:53.41 | Bipul | quit |
| 11:53.56 | Bipul | when i call any number on zoiper |
| 11:54.19 | kaldemar | what says? |
| 11:54.46 | Bipul | when i call any number on zoiper it says "bearerecapability notauth" |
| 11:54.50 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 11:54.51 | techknowlust | is it possible to access channel variables in macros ? |
| 11:54.52 | Bipul | and then call ended. |
| 11:54.53 | wasanzy | can you help me kaldemar? |
| 11:55.02 | *** part/#asterisk billmania (~bill@38.98.130.98) |
| 11:55.04 | kaldemar | wasanzy: pastebin is the name of the game... |
| 11:55.33 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:55.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:55.42 | wasanzy | what game please? |
| 11:56.06 | kaldemar | Bipul: besides, you already had zoiper registered to asterisk. what did you do to break that? |
| 11:56.36 | Bipul | kaldemar, i have to shut down my system |
| 11:56.49 | Bipul | i am student i have to go my college also |
| 11:57.24 | kaldemar | wasanzy: pastebin a sip debug of a call. |
| 11:57.34 | wasanzy | oh ok |
| 11:57.40 | wasanzy | just a minute |
| 11:59.30 | techknowlust | anyone know about channel variables in macros ? |
| 12:01.12 | Bipul | ok |
| 12:01.22 | Bipul | i dont know how it going to short out |
| 12:02.11 | kaldemar | techknowlust: channel variables can be accessed in macros just like in any other part of the dialplan. EXTEN will change to "s" when a macro is run, but the original is held in MACRO_EXTEN. https://wiki.asterisk.org/wiki/display/AST/Dialplan+Macros+Channel+Variables |
| 12:03.37 | techknowlust | hmm weird |
| 12:03.44 | techknowlust | I'm trying to use membermacro with the queue |
| 12:03.58 | techknowlust | and it's not able to see a channel variable I've set before the call enters the queue |
| 12:05.14 | Bipul | kaldemar, http://itchecker.blogspot.com/2010/03/zoiper-registration-error.html |
| 12:05.16 | Bipul | check this |
| 12:07.04 | wasanzy | Kaldemar: here is my sip debug: http://pastebin.com/RUTCp2u0 |
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| 12:11.34 | wasanzy | kaldemar: are you there? |
| 12:15.08 | kaldemar | wasanzy: deja vu. both your twinkles are trying to use the same UDP port for RTP. again. |
| 12:16.02 | kaldemar | and they are running on the same machine with asterisk. what is the goal in all this? |
| 12:16.12 | wasanzy | oh sorry let me try using a different machine and see |
| 12:17.03 | wasanzy | My goal is to enable other computers connect to astrisk and be able to talk |
| 12:17.20 | wasanzy | to each other |
| 12:19.01 | kaldemar | test is with different computers then instead of giving yourself headaches with that setup. |
| 12:20.02 | *** join/#asterisk fyrfaktry (~dread@adsl-070-145-120-039.sip.pns.bellsouth.net) |
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| 12:22.21 | wasanzy | yes, am waiting for the other machine to finish installing twinkle then I will try |
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| 12:32.59 | kwk | @kaldemar, @pabelanger: I'm looking for an AMI proxy for asterisk 1.8 and was wondering if you guys could recommend using astmanproxy. |
| 12:34.24 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 12:37.44 | techknowlust | can someone help me with debugging variable scope in a dialplan macro ? |
| 12:39.49 | kaldemar | techknowlust: normally channel variables are accessible in a macro. it must be the queue app that interferes in some way. sounds like the macro is executed in a child channel. try prepending the variable with _ when you set it to see if it gets inherited. |
| 12:40.11 | techknowlust | hmm ok. |
| 12:41.30 | *** join/#asterisk ssureshot (~digitolx@rrcs-24-142-174-146.central.biz.rr.com) |
| 12:41.50 | techknowlust | nope, unfortunately not |
| 12:44.15 | techknowlust | if variables aren't being inherited is there any other way to access them on a per-call basis ? |
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| 12:53.07 | techknowlust | kaldemar: do you know if it's possible to pass parameters to the macro callback in the Queue command ? |
| 12:53.40 | techknowlust | I tried macro(arg1, arg2) syntax but that doesn't work. It instead looks to execute the macro macro(arg1 |
| 12:55.35 | bchia | the correct syntax is macro(macroName,arg1,arg2,arg3,etcâ¦) |
| 12:56.20 | *** join/#asterisk darkskiez (~darkskiez@2001:470:9278:5:2e0:4cff:fe68:1e29) |
| 12:56.55 | techknowlust | so the Queue command should look like Queue(sales,,,,,,macro(Macro,arg1)) ? |
| 12:57.39 | wasanzy | Kaldemar: my new sip debug when, call was make from two different machines : http://pastebin.com/DwduL6WE |
| 12:58.44 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 12:59.14 | techknowlust | No such context 'macro-macro(queue-answered' for macro 'macro(queue-answered' |
| 12:59.25 | Bipul | p3nguin, ping |
| 13:00.22 | bchia | ok, I see your calling from within queue app, don't know that you can pass variables that way (I'm looking at core show application queue) |
| 13:02.54 | wasanzy | kaldemar: have you seen it? |
| 13:03.05 | techknowlust | bah. I can't pass variables and channel variables aren't inherited. |
| 13:05.57 | bchia | You could potentially set as a Global or a Shared variable (If you need EXTEN, CONTEXT or PRIORITY you could try MACRO_EXTEN, MACRO_CONTEXT, and MACRO_PRIORITY) |
| 13:06.30 | techknowlust | I am setting a unique ID to the call and assigning it as a channel variable |
| 13:06.47 | techknowlust | and the second thing I need is the extension of the agent that has answered the call |
| 13:06.49 | Kobaz | UNIQUEID is already unique for a single server |
| 13:07.21 | techknowlust | unique per call ? |
| 13:07.23 | Kobaz | techknowlust: the option is M not macro... you should use U instead though |
| 13:07.34 | Kobaz | yes per call |
| 13:08.06 | techknowlust | hmm ok. and what option are you talking about ? |
| 13:08.24 | Kobaz | gosub on pickup for Dial |
| 13:08.35 | Kobaz | oh you need one for Queue |
| 13:08.41 | techknowlust | yep. |
| 13:09.04 | techknowlust | are those UNIQUEID's unique over the lifetime of the box or just unique to the calls at the moment ? |
| 13:10.25 | Kobaz | okay, i was talking about dial... yeah the macro option is what you need |
| 13:10.43 | Kobaz | or gosub, but you should use gosub it's the modern way |
| 13:10.54 | kaldemar | wasanzy: pastebin a whole call. there was no answer in that one. |
| 13:11.02 | Kobaz | lifetime of the box (assuming the time and date doesn't change) |
| 13:11.10 | Kobaz | like, you set the date in the past or something |
| 13:11.15 | techknowlust | hmm ok. |
| 13:11.22 | techknowlust | how would using a gosub solve this problem ? |
| 13:11.22 | wasanzy | ooops, then some mistake some where, I will try again |
| 13:11.41 | Kobaz | when the agent picks up, the context will run |
| 13:11.46 | Kobaz | and you can do ${CHANNEL} and see who picked up |
| 13:12.01 | techknowlust | will it have access to the previous channel variables though ? |
| 13:12.15 | techknowlust | I'm not entirely sure of the difference between a GoSub and Macro |
| 13:12.20 | Kobaz | you need to set something like |
| 13:12.28 | Kobaz | __original_uniqueid |
| 13:12.44 | Kobaz | in the initial channel... the __ will make it pass through to the dest channel |
| 13:12.52 | techknowlust | ok. |
| 13:12.58 | Kobaz | and then you get ${original_uniqueid} |
| 13:13.05 | Kobaz | in the agent channel on pickup |
| 13:13.23 | techknowlust | hmm ok I'll give this a shot |
| 13:13.37 | techknowlust | thanks for the help btw, appreciate |
| 13:13.37 | *** join/#asterisk Nasga (~Nasga@85.212.10.93.rev.sfr.net) |
| 13:13.52 | Kobaz | yeah, i do that sort of stuff all the time |
| 13:14.00 | Kobaz | pickup macros/gosub is great |
| 13:14.56 | techknowlust | awesome it worked! |
| 13:15.03 | techknowlust | fucking love you right now |
| 13:15.28 | Kobaz | that'll be $199.99 |
| 13:15.31 | Kobaz | plus tax |
| 13:15.41 | Kobaz | and some homemade cookies |
| 13:16.22 | *** join/#asterisk dfamorato (~dfamorato@2001:470:5:630:cabc:c8ff:fe9b:26b7) |
| 13:16.33 | Kobaz | maybe some coffee too |
| 13:16.36 | *** part/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 13:16.39 | techknowlust | god damn you're a shrude businessman |
| 13:16.44 | techknowlust | shrewd* |
| 13:16.53 | techknowlust | bleh I haven't slept in coming up on 20 hours |
| 13:16.57 | Kobaz | gotta pay the rent |
| 13:17.13 | Kobaz | sleep is good |
| 13:17.29 | Sylnai | nothing wrong with lack of sleep, some of the best code I've written was after being up for 30 hours |
| 13:18.36 | techknowlust | i don't feel tired. when you're working on something you like putting in the effort is easy |
| 13:18.43 | techknowlust | also setting yourself deadlines |
| 13:18.53 | Kobaz | deadlines are good |
| 13:19.01 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 13:19.48 | WIMPy | Save the lines! |
| 13:19.56 | WIMPy | They want to live! |
| 13:20.11 | SunTsu | but beware the zombielines |
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| 13:21.35 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 13:22.13 | Kobaz | are you using res_timer_timerfd |
| 13:22.39 | Kobaz | blarg |
| 13:22.46 | Kobaz | it's too early |
| 13:24.25 | eject_ck | Hi all, I'm trying to make fail2ban working with * logs :). I have question - It was working before but somewhere after upgrade to 1.8 it don't ban brute-force(rs). Just looked into logs and found that current log event looks like this: 464]: NOTICE[8525]: chan_sip.c:23511 in handle_request_register: Registration from 'sip:111@myhost.name' failed for '91.151.216.11:51211' - Wrong password |
| 13:25.02 | eject_ck | where fail2ban regex is NOTICE.* .*: Registration from '.*' failed for '<HOST>' - Wrong password |
| 13:25.11 | techknowlust | I presume exiting non-zero means that it didn't execute a Hangup ? |
| 13:26.59 | eject_ck | I'm concerned about <HOST> where only IP address is expected as I see, but asterisk's log contain port number as well |
| 13:27.59 | eject_ck | it was changed recently or not ? I mean does anybody know if in early versions there was no :port in that event line ? |
| 13:30.04 | Bipul | p3nguin, ? hellow there |
| 13:31.56 | kaldemar | Bipul: you need to look at the asterisk CLI when you try to call. all you can see from zoiper is that the call fails. asterisk CLI will say why. |
| 13:32.10 | Bipul | yes it working dude |
| 13:32.43 | Bipul | kaldemar, it's working |
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| 14:13.50 | ssureshot | WIMPy: I just wanted to update you,, the reason I couldn't dial tollfree was because I wasn't sending the proper phone number that was associated with that trunk |
| 14:14.29 | ssureshot | proper caller ID,, I was sending the number from the primary line |
| 14:15.02 | WIMPy | Interesting feature. |
| 14:15.11 | WIMPy | What cause did you get? |
| 14:15.24 | ssureshot | _Corey_: read above you helped also lol.. |
| 14:15.28 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:15.36 | _Corey_ | hm? |
| 14:15.48 | ssureshot | WIMPy: I was getting a cause 58 |
| 14:16.03 | _Corey_ | oh, cisco 7.4 sip firmware... i remember :) |
| 14:16.40 | _Corey_ | no, that doesn't make sense |
| 14:16.54 | WIMPy | ssureshot: By spending time in here, I more and more understand, why americans think that ISDN doesn't work. It seems to be true there. |
| 14:17.22 | ssureshot | _Corey_: lol,, yeah that was a while back :) friday we chatted about long distance calling |
| 14:17.54 | ssureshot | WIMPy: yeah we Americans lock everything down to where it's impossible to make it work without hoops I guess heh |
| 14:18.05 | _Corey_ | I remember now... :) You were nice enough to change the subject after that guy was lecturing everyone about how telnet can cure cancer |
| 14:18.47 | ssureshot | ha,, |
| 14:25.25 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 14:26.18 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 14:26.30 | Katty | I"M IN THE MOOD FOR LOVE SIMPLY BECAUSE YOUR"E NEAR ME! |
| 14:27.27 | jacc0 | ty katty |
| 14:27.33 | Katty | :> |
| 14:27.52 | jacc0 | :) |
| 14:28.15 | Katty | brain is so foggy |
| 14:29.05 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:31.07 | ssureshot | DARLING BUT WHEN YOUR NEAR ME, I'M IN THE MOOD DON'T MEAN TO BE RUDE |
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| 14:39.50 | Chainsaw | WIMPy: Not to mention the robbed bit signalling made it look inferior compared to V90 modems (56kbit vs "56k modem") |
| 14:41.39 | Chainsaw | WIMPy: Whereas overseas, "56k" to proper 64k and multiple inbound phone number made it a step up. |
| 14:42.51 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 14:43.47 | WIMPy | Or the different standards? |
| 14:45.02 | Chainsaw | WIMPy: It's the US. It has a different standard to the rest of the world. That much is tradition I'm afraid. |
| 14:45.06 | WIMPy | I do remember when I first looked at the book and read the comparison between ISDN and SIP I was sure I shouldn't expect much from it. |
| 14:45.28 | WIMPy | But I think it might be true for NA. But for the rest of the world, the exact opposite ist true. |
| 14:45.40 | WIMPy | And to itself? |
| 14:45.53 | Chainsaw | WIMPy: ISDN gets you so much more call signalling. I don't understand why anyone would want to put up with analog lines. |
| 14:47.23 | WIMPy | No idea. But the point is that the VOIP stuff is still a long way behind since H.323 has gone out of fashion. |
| 14:48.41 | Linuturk | price |
| 14:48.57 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 14:49.12 | Linuturk | a pots line is cheaper than a pri |
| 14:49.40 | WIMPy | There's BRI in between. |
| 14:49.53 | Chainsaw | has BRIs, indeed |
| 14:50.44 | Chainsaw | WIMPy: My gateways seem to map ISDN cause codes to SIP errors quite well though. |
| 14:52.43 | WIMPy | There's an almost endless list of other stuff that suffers from translational losses. |
| 14:54.30 | Chainsaw | WIMPy: If it isn't cause codes, I'm not sure what you think you're losing out on. |
| 14:54.58 | Chainsaw | WIMPy: Audio on 64kbit ulaw or alaw is as good as it gets. |
| 14:55.33 | WIMPy | 64k is 64k, no, I'm just about signalling. |
| 14:55.59 | Chainsaw | signals for WIMPy to be more specific |
| 14:56.35 | WIMPy | One extremely basic issue is that you don't have a type of number in SIP. And even worse it suggests you should always send complete, unformatted numbers. |
| 14:57.21 | WIMPy | Now we all know that isn;t always the case, but without any clear format you can only guess the meaning. |
| 14:58.19 | WIMPy | Using a + in front of a number ist the only unambigous format, but not really within secification. |
| 14:59.02 | WIMPy | That's a major issue with the Asterisk dialplan as well, BTW. |
| 14:59.41 | *** join/#asterisk irroot (~irroot@197.168.45.231) |
| 14:59.52 | WIMPy | And the sheer number of additions to SIP makes it completely undeterminable. |
| 14:59.53 | *** join/#asterisk minaguib (~mina@modemcable098.129-202-24.mc.videotron.ca) |
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| 15:39.59 | doctorray | If I have my voicemail defined in users.conf, is there any way to get app Directory to work? using 1.6.2.6 |
| 15:40.16 | doctorray | or do I need to transition everyone to voicemail.conf |
| 15:40.39 | Micc_ | Is there a max number of blf subscriptions I should put on a server? Any rule of thumb here blf to peer ratio or anything? |
| 15:42.01 | Micc_ | I've got almost 600 peers and 1500 active subscriptions on one server, I feel like its getting to be on the high side and I might be hitting a lock more often because of it. |
| 15:43.01 | Micc_ | and cps ranges from about 1-5 |
| 15:44.17 | irroot | Micc try not have all phones watching all the same phone massive performance hit |
| 15:45.05 | irroot | there some locks in the statechange code but been worked on and should be thing of the past soon |
| 15:45.12 | Micc_ | irroot, yeah I've tried to keep them down to 10-15 subscriptions per phone. Some have less. |
| 15:45.54 | Micc_ | irrot, so it could still be in 1.8.5.0 or you think its fixed in that version? |
| 15:45.57 | irroot | if you have 100 phones all monitoring a busy phone it will get messy |
| 15:45.58 | *** join/#asterisk darkskiez (~darkskiez@62-50-207-4.client.stsn.net) |
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| 15:46.32 | irroot | 1.8.6 should have the latest in it was in 1.8 after 1.8.5 was tagged |
| 15:47.13 | Micc_ | I hope I can get through till that release without a lock then. |
| 15:47.44 | Micc_ | maybe i'll turn off blf for a couple of the big ones till then. |
| 15:48.27 | Jcook_5xData | anyone know if there a way to restart dahdi safely ? |
| 15:48.57 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:51.47 | Bipul | p3nguin, p ing |
| 15:52.10 | p3nguin | yes? |
| 15:52.17 | Bipul | can i pm you |
| 15:52.46 | p3nguin | Yes. |
| 15:53.40 | Chainsaw | irroot: We have a 1.8.6 already? |
| 15:53.43 | Chainsaw | irroot: Did I miss it? |
| 15:55.41 | irroot | no no its .next |
| 15:55.58 | irroot | chainsaw the dev cycle is 1.8.6 current |
| 15:56.25 | Chainsaw | irroot: Ah, good. You had me worried there for a minute. |
| 15:59.48 | ChannelZ | Jcook_5xData: safely? LIke not setting your house on fire in the process or..? |
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| 16:06.03 | Micc_ | irroot, I just hit it again, is this the one that will be fixed in 1.8.6? http://pastebin.com/kVRXMRQr |
| 16:06.43 | Micc_ | I've limited my subscriptions down to about 500 from 1500, so I'm hoping that will help me get through the week. When is 1.8.6 scheduled to release the rc? |
| 16:08.49 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 16:09.08 | Micc_ | If thats thats not the same one, I guess I need to file a bug. |
| 16:09.49 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 16:12.17 | irroot | Micc please check bugs and report there is a locking order violation here |
| 16:12.45 | irroot | Micc_ its related to pickup |
| 16:13.07 | Micc_ | irroot, directected call pickup? I don't use that. |
| 16:13.54 | irroot | or features |
| 16:13.58 | irroot | *8 |
| 16:14.04 | irroot | or some other masq |
| 16:15.50 | Micc_ | we don't use *8, but someone could hit it accidentally I suppose. |
| 16:15.55 | Micc_ | we use # transfers |
| 16:16.13 | Micc_ | can I turn off some features to prevent this? |
| 16:18.17 | Micc_ | if I comment out the lines in features.conf will that turn off that feature or do I need to set it = to nothing |
| 16:18.34 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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| 16:22.28 | benno2 | I have a problem with asterisk writing NO ANSWER in the calllog while call is is still in progress (after the 2nd Dial() is executed which calls an ext nr). any idea what's wrong ? see here for an explanation of my problem: http://pastebin.com/QxBnFWh4 thanks alot for the hints how I could solve the problem! |
| 16:24.21 | ChannelZ | Without testing it, my initial guess is that the 'NO ANSWER' is coming from the first Dial that no one answers |
| 16:25.15 | ChannelZ | oh except I didn't notice your cdr paste at the bottom |
| 16:26.21 | ChannelZ | hmm |
| 16:27.20 | adyn | is there anyway to flush the Def. Username from the sip peer? I've tried reloading the sip config, deleteing the sip entry and reloading then restoring it and reloading. Next to just completely restarting the who process is there any ways to get flush it out? |
| 16:27.43 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 16:27.56 | benno2 | ChannelZ, do I need to add a hangup statement or something ? basically the SIP phones ring and the external phone too, but then someone answers on the SIP phones and the outbound DAHDI channel gets a hangup and at this moment the NO ANSWER line is written into the log. then the call contnues but after the SIP phone is hung up no line is written. strange |
| 16:28.10 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 16:29.16 | Micc_ | irroot, I've turned off all features for now. I don't think many people use them anyways. How do I go about finding if theres already a bug filed for this one? I'm guessing searching for some of the stuff in the locks |
| 16:38.13 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 16:38.26 | *** join/#asterisk abel408 (429863d2@gateway/web/freenode/ip.66.152.99.210) |
| 16:39.00 | FLeiXiuS | Anyone know of any free web-based SIP clients |
| 16:39.59 | *** join/#asterisk g-ram (~gsaathoff@12.200.95.45) |
| 16:43.27 | *** join/#asterisk abel408_ (429863d2@gateway/web/freenode/ip.66.152.99.210) |
| 16:44.10 | Jcook_5xData | ChannelZ, lol mean like restart asterisk safely, so it will restart when no call are active |
| 16:44.34 | ChannelZ | core stop when convenient |
| 16:44.36 | abel408_ | Hey everyone, I'm seeing a very strange problem with my iax trunk. I have to asterisk boxes and the iax trunk was working great for a while. Now, for some reason 'iax2 show peers' on server1 shows server2 as unreachable and calls do not connect. But I can ping server2 from server1 and calls can come through to server 1 from server2 |
| 16:45.00 | abel408_ | two* asteisk boxes |
| 16:45.03 | ChannelZ | although I don't know which version that appeared |
| 16:48.40 | *** join/#asterisk voxio (mark@unaffiliated/voxio) |
| 16:49.07 | voxio | hi guys.. I'm using SIP for user extensions, do you think 60ms is too high from the asterisk server to the client? |
| 16:49.09 | ChannelZ | benno2: sorry I'm not sure what the expected behavior is, if it's a bug or what. What is the DAHDI channel? (analog, t1, etc) |
| 16:50.16 | benno2 | ChannelZ, isdn bri card from junghanns |
| 16:50.39 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 16:55.46 | Jcook_5xData | FLeiXiuS, this may point you some where http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software |
| 16:58.57 | Jcook_5xData | FLeiXiuS, also http://sourceforge.net/projects/shone/ |
| 16:59.53 | Micc_ | Hit the lock again. I need to hire an asterisk dev to take a look at this and make a patch today. |
| 17:06.09 | Micc_ | Who is available for some consulting to make a patch to fix a locking issue? |
| 17:06.36 | Micc_ | http://pastebin.com/kVRXMRQr and http://pastebin.com/TZvPfJxx, not sure if they are related but I suspect they are. |
| 17:06.48 | Micc_ | This is in 1.8.5.0 |
| 17:07.11 | WIMPy | Have you tried the SVN version? |
| 17:08.00 | Micc_ | thats a little scary with a production system with 100 businesses on it. |
| 17:08.40 | Micc_ | But I guess it won't matter if we lock up again today we could be out of business. |
| 17:09.20 | anonymouz666 | that's why I am waiting version 1.8 to make more birthdays |
| 17:09.46 | Micc_ | anonymouz666, I was waiting too, but we hit the same lock in 1.6.2.19 two times last week. |
| 17:10.06 | anonymouz666 | still running the 1.4 |
| 17:10.07 | Micc_ | and since 1.6.2.19 isn't supported, we had to make to switch in hopes we could get some resolution to this problem. |
| 17:10.09 | anonymouz666 | :P |
| 17:10.35 | Micc_ | we run a multi tenant system with parking, so we can't run 1.4. |
| 17:11.13 | anonymouz666 | I am not saying that everybody should be using 1.4. If you need the new features that is a good reason to try newer versions |
| 17:11.40 | Micc_ | I'm gonna have to try the svn version soon. |
| 17:12.33 | anonymouz666 | Did you see the commit logs? Maybe there's a fix for that |
| 17:15.29 | Micc_ | where can I look at the commit logs? |
| 17:16.14 | chazzam | http://svnview.digium.com/svn/asterisk/branches/1.8/ ? |
| 17:22.16 | FLeiXiuS | Jcook_5xData, thanks - I'll look into both of those. |
| 17:24.01 | Gugge | Micc_: i downgraded to 1.6.2.11, and havent had a lockup since |
| 17:24.45 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 17:25.18 | Gugge | Micc_: i had problems like yours in 1.6.2.18 too |
| 17:25.49 | Micc_ | Gugge, good to know. I remember 1.6.2.11 was pretty good. |
| 17:26.08 | Micc_ | I wonder if this is the same thing https://issues.asterisk.org/view.php?id=18823 |
| 17:26.59 | *** join/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 17:27.34 | *** join/#asterisk McBoingbo (~McBoingBo@mail.hrsg.ca) |
| 17:28.43 | McBoingbo | hey guys, how goes the battle? |
| 17:28.53 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.109) |
| 17:29.54 | SuPrSluG | what causes this message on a subscribe? -> SIP/2.0 401 Unauthorized |
| 17:31.06 | SuPrSluG | the it has the subscription, but always gives a 401 message |
| 17:34.44 | jmls | ok. who wants to make an old man happy ? |
| 17:34.54 | Qwell | umm |
| 17:35.05 | jmls | I should know better, but for the life of me I can't get this working :) |
| 17:35.10 | jmls | Qwell: steady .... |
| 17:35.13 | jmls | http://pastebin.com/PA4TCDcy |
| 17:35.41 | jmls | wtf do I need to do to make dialling 800@example call the music on hold ? |
| 17:36.06 | Qwell | I think the include is reducing the specificity. |
| 17:36.12 | Qwell | or, rather, reducing the value of |
| 17:36.29 | *** join/#asterisk ExpertOrBust (~ExpertOrB@wsip-24-234-159-70.lv.lv.cox.net) |
| 17:36.32 | brainiac | I have a TE1110P card that I can't get to come up. When I run dahdi_cfg -vvvvv, it says DAHDI_SPANCONFIG failed on span 1: Invalid argument (22). Where can I look to fix the problem? |
| 17:36.40 | Qwell | you don't want your patterns in the including context |
| 17:37.05 | kaldemar | jmls: matching order is extensions first, then includes in the order of appearance. |
| 17:37.22 | atheos | brainiac - what does lsmod show you? is the driver loaded? what does dmesg tell you, does it detect the hardware? |
| 17:37.58 | chazzam | brainiac: sounds like something is configured wrong in /etc/dahdi/system.conf |
| 17:38.36 | chazzam | could be due to any of the things that atheos just mentioned, or just a typo or other error in the file |
| 17:39.29 | brainiac | driver is loaded and I am looking in /etc/dahdi/system.conf |
| 17:39.42 | jmls | ffs. i've got this arse-about-face. |
| 17:40.04 | jmls | the _X should be included , not the 800 |
| 17:40.28 | jmls | heh. Only been using * since mid 2005 ... |
| 17:40.42 | jmls | thanks |
| 17:40.56 | brainiac | everything looks ok. dmesg shows the hardware was detected |
| 17:41.13 | Qwell | jmls: there've been some tweaks to the way that stuff works. are you using like 1.4 or something? |
| 17:42.49 | *** join/#asterisk cerberus_za (~coert@196-215-13-168.dynamic.isadsl.co.za) |
| 17:42.55 | jmls | 1.10 ;) |
| 17:43.02 | McBoingbo | xchat is the shiznit, wasnt this nice when I looked at it like a few years back |
| 17:43.04 | Qwell | well then |
| 17:43.06 | jmls | wait. |
| 17:43.12 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 17:43.14 | jmls | 1.8 trunk. Should be 1.10 |
| 17:43.20 | jmls | don't know what went wrong there. |
| 17:43.30 | Qwell | no such thing as 1.8 trunk O.o |
| 17:43.31 | MrTelephone | Where can I find definitions for setcallerpres indications? |
| 17:43.43 | jmls | heh. 1.8 svn |
| 17:43.51 | Qwell | McBoingbo: xchat really hasn't changed much in years |
| 17:43.59 | MrTelephone | i have no idea what prohib_not_screened means :( |
| 17:44.02 | McBoingbo | well maybe it was longer than that :P |
| 17:44.10 | McBoingbo | the gui wasnt so...polished |
| 17:44.15 | jmls | Qwell: and the tweaks are ? |
| 17:44.38 | Qwell | jmls: dunno, but there've been some subtle changes since 1.4, that make that type of thing happier. |
| 17:44.42 | WIMPy | MrTelephone: Number withheld. |
| 17:45.13 | MrTelephone | Wimpy, then there are passed and withheld options but they do not indicate name or number |
| 17:45.22 | MrTelephone | s/there/they |
| 17:45.27 | p3nguin | sed: -e expression #1, Unterminated `s' command |
| 17:45.34 | MrTelephone | actually that was wrong again :( |
| 17:46.20 | McBoingbo | Im excited to get Asterisk 1.8.5 into our system, created the VM, just got to figure out how to couple..."peer" my production Asterisk 1.2.12 to it, if you guys have any suggestions on what articles I should be reading up on, shoot! |
| 17:46.26 | MrTelephone | I received my atlas 550 with t1 modules and that thing rocks. It allows you to share a t1 and have backup t1 switching. I'm surprised to not see that machine documented under the high availibilty section of the wiki |
| 17:46.26 | p3nguin | "core show application SetCallerPres" shows what each type does. |
| 17:46.34 | MrTelephone | thanks penguin |
| 17:47.22 | p3nguin | It's not detailed, but it's better than nothing. |
| 17:47.32 | MrTelephone | I'll check the source code and see what the deal is. core show application didn't show much |
| 17:48.29 | p3nguin | http://pastebin.com/ax9rYckQ |
| 17:48.54 | WIMPy | MrTelephone: There are several -pres entries in CALLERID. |
| 17:51.35 | MrTelephone | SetCallerPres is deprecated. Please use Set(CALLERPRES()=%s) instead.\n" |
| 17:51.39 | MrTelephone | I guess that would help too |
| 17:51.45 | MrTelephone | using the proper syntax |
| 17:54.13 | ChannelZ | Syntax is for squares |
| 17:55.16 | *** join/#asterisk samandiriel (~logos@S0106c03f0eb613f5.ed.shawcable.net) |
| 17:58.58 | *** join/#asterisk lcat (~lcat@187.45.253.242) |
| 18:06.29 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 18:07.36 | kaldemar | MrTelephone: CALLERPRES is also deprecated in favor of CALLERID. |
| 18:07.55 | p3nguin | CALLERID(pres)? |
| 18:08.38 | kaldemar | name-pres and num-pres are the fields for CALLERID. |
| 18:09.34 | *** part/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com) |
| 18:10.46 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 18:19.56 | MrTelephone | ok |
| 18:20.11 | Qwell | tzafrir: ping? |
| 18:20.27 | Qwell | tzafrir: neat trick for you. sed -e 's/;.*//;/^ *$/d' |
| 18:20.52 | Qwell | basically s/' -e '/;/ |
| 18:21.21 | Qwell | in case you ever need to play sed golf... |
| 18:22.27 | *** join/#asterisk capt_cassimir (~arch@178.73.217.49) |
| 18:22.43 | capt_cassimir | Quick question: What is "call appearance"? |
| 18:23.04 | Qwell | Do you mean line appearance, maybe? We need context. |
| 18:23.32 | capt_cassimir | "A client station is a Linksys IP phone with one or more extensions and call appearances provisioned" |
| 18:23.45 | capt_cassimir | from the manual for my SPA921 |
| 18:23.53 | Qwell | yes, line appearances |
| 18:24.11 | capt_cassimir | Where can i read about this concept? My google searches aren't ammounting to much |
| 18:24.32 | Qwell | It just means the number of buttons configured to different "lines"(accounts). |
| 18:24.51 | capt_cassimir | Ok, that makes sense on some level |
| 18:25.04 | capt_cassimir | but the phone is listed as having support for "1 line" and "2 call appearances" |
| 18:25.24 | jmls | so, if you have a device like a Cisco7960, does it have 6 lines, or 6 accounts ? (just wanting to know the correct terms) |
| 18:25.30 | wasanzy | Qwell: I still have the problem |
| 18:25.54 | wasanzy | with my sound |
| 18:26.55 | jmls | and would you construct a Model like Device->Account[1..n] or just Account1..n for each "line" ? |
| 18:29.06 | wasanzy | the sip debug of a call is here: http://pastebin.com/Jd1eRtmt |
| 18:29.17 | *** join/#asterisk abel408 (~abel408@66-152-99-210.tvc-ip.com) |
| 18:29.37 | wasanzy | The issue is that, One person can hear the other but the other can't hear him |
| 18:29.46 | wasanzy | please any body help me |
| 18:31.51 | *** join/#asterisk radic (~radic@tmo-096-83.customers.d1-online.com) |
| 18:32.43 | capt_cassimir | jmls: I'm working through the verbage now, and based on the context in my phone manual, it looks like the "call appearance" in cisco parlance refers to the entries in the dialplan, while "line appearance" is a cisco specific term that refers to extension availability |
| 18:33.02 | Qwell | jmls: It has 6 buttons. |
| 18:33.27 | wasanzy | any one to help me? |
| 18:33.28 | Qwell | jmls: the buttons can be speeddials, or various other things |
| 18:33.33 | abel408 | Hey everyone, I'm seeing a very strange problem with my iax trunk. I have to asterisk boxes and the iax trunk was working great for a while. Now, for some reason 'iax2 show peers' on server1 shows server2 as unreachable and calls do not connect. But I can ping server2 from server1 and calls can come through to server1 from server2 |
| 18:34.00 | jmls | Qwell: ya, but each "button" can be assigned to a different sip account |
| 18:34.41 | Qwell | using the word "line" on voip is silly |
| 18:35.12 | Qwell | The reason you have "2 line" analog phones, is because it required 2 actual physical lines. |
| 18:35.18 | jmls | yeah, that's why I was trying to see if Account was a better term |
| 18:35.21 | wdoekes2 | wasanzy: it looks like it gets set up ok |
| 18:35.26 | wdoekes2 | rtp set debug on |
| 18:35.37 | wdoekes2 | and check if there is rtp flowing in all directions |
| 18:37.10 | wasanzy | ok |
| 18:37.22 | wasanzy | will get back to you. |
| 18:38.24 | abel408 | If I'm not mistaken, the only thing the qualify command does is send a ping request every few seconds to monitor the connection correct? |
| 18:39.10 | abel408 | So if a trunk was listed as unreachable that means that ping request failed right? |
| 18:39.25 | p3nguin | jmls: The term is buttons. You can assign those buttons to do a whole bunch of stuff, not just use accounts. |
| 18:39.53 | p3nguin | But if you want to assign a different account to each button, you can do that. |
| 18:40.05 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 18:40.51 | jmls | right. but each phone (7960 / 7940) can only have 6 / 2 accounts assigned at max, right ? |
| 18:41.54 | jmls | and a 3cx softphone seems to have any number of accounts ... |
| 18:42.00 | p3nguin | It can't have two accounts per button, as far as I know, so yes. |
| 18:42.12 | jmls | but has 5 "lines" |
| 18:43.42 | capt_cassimir | I'm kind of curious why they call it a "line" - is this a legacy / roundabout way to refer to SIP accounts supported? |
| 18:44.01 | Qwell | capt_cassimir: see above. |
| 18:44.03 | p3nguin | It's certainly a legacy term. |
| 18:44.25 | abel408 | If a trunk is listed as unreachable that means a ping request failed right? |
| 18:44.57 | capt_cassimir | So a "1 line" voip phone is really a "1 SIP account" voip phone |
| 18:45.07 | p3nguin | It's a one-button phone. |
| 18:45.14 | capt_cassimir | oh oh oh |
| 18:45.28 | capt_cassimir | ok, back to the point earlier "it has 6 buttons" |
| 18:45.28 | _Corey_ | lines usually mean concurrent calls with these things... |
| 18:45.45 | p3nguin | And that one button can have one account assigned on it, or probably something other than an account (which would make the phone somewhat difficult to use). |
| 18:46.02 | capt_cassimir | heh |
| 18:46.23 | jmls | can 2 different phones have the same sip account assigned ? |
| 18:46.25 | jmls | thinks not |
| 18:46.30 | capt_cassimir | _Corey_: That was my interpretation, but once i started getting into vendor docs, they didn't want to spell it out so clearly |
| 18:46.30 | Qwell | yes |
| 18:46.33 | Qwell | but it won't do what you want |
| 18:46.35 | jmls | what? |
| 18:46.40 | capt_cassimir | jmls: My manual calls that "shared line appearance" |
| 18:46.59 | Qwell | jmls: they'd fight for the registration |
| 18:46.59 | capt_cassimir | oh wait, no it doesnt. Disregard. |
| 18:47.03 | jmls | heh |
| 18:47.09 | Qwell | so you'd get the call on one or the other |
| 18:47.11 | p3nguin | Other channel drivers can support it. |
| 18:47.18 | jmls | so, not a good idea. |
| 18:47.22 | Qwell | right |
| 18:47.23 | p3nguin | not in SIP. |
| 18:51.09 | capt_cassimir | cisco kills me with their love of acronyms |
| 18:51.46 | capt_cassimir | VOIP Interface = VI. VoiceMail Waiting Indicator = VMWI. |
| 18:52.03 | Qwell | [v]mwi is common |
| 18:52.22 | capt_cassimir | that's almost encouraging |
| 18:52.31 | capt_cassimir | what about "full cone NAT" |
| 18:52.38 | p3nguin | That's real. |
| 18:52.40 | Qwell | common networking term |
| 18:52.53 | _Corey_ | VI is the only fishy one |
| 18:52.54 | capt_cassimir | Common networking term or common telecom term? |
| 18:52.59 | Qwell | networking |
| 18:53.13 | capt_cassimir | I work with a bunch of networking nerds and have never heard that |
| 18:53.13 | p3nguin | It's not something a lot of people say when talking about networking, but it's a real term. |
| 18:53.19 | Qwell | http://tools.ietf.org/html/rfc3489 |
| 18:53.23 | p3nguin | It's a type of NAT. |
| 18:53.47 | capt_cassimir | oh i saw, but it's the type that most of my crowd calls "1 to 1" |
| 18:54.56 | p3nguin | That's the more common term. |
| 18:56.33 | capt_cassimir | Well, that's kind of why i'm reading this stuff anyway. I find that when i want to talk about this stuff, i don't know the industry terms |
| 18:56.54 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.109) |
| 18:57.04 | p3nguin | A lot of people make up their own terms and never use the industry terms. |
| 18:57.30 | p3nguin | And then we have no idea what the hell they are talking about when they come here asking questions. |
| 18:57.41 | capt_cassimir | LOL i've noticed |
| 18:58.13 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-twqyjdmmrdruwsyl) |
| 18:58.25 | p3nguin | Sometimes it can be figured out, but I don't know why we should have to TRY to figure it out. |
| 18:58.28 | abel408 | It baffles me how asterisk can report a trunk as unreachable, yet if I just ping the trunks dns name I can reach it fine... |
| 18:58.45 | Qwell | abel408: icmp ping != sip ping |
| 18:58.50 | p3nguin | Like this guy and his incorrect usage of the term "trunk." |
| 18:59.09 | abel408 | Qwell: It's an iax2 trunk |
| 18:59.17 | Qwell | abel408: icmp ping != iax2 ping |
| 18:59.42 | capt_cassimir | :D |
| 19:00.16 | *** join/#asterisk sourcode (~code@ppp-58-11-114-252.revip2.asianet.co.th) |
| 19:00.33 | p3nguin | Where's that one guy to tell abel408 to just telnet it to see if it works? :/ |
| 19:00.34 | Nugget | telnet is eeeeeeevil! |
| 19:00.46 | _Corey_ | let's not go there again |
| 19:00.50 | p3nguin | hahahahaha |
| 19:00.54 | capt_cassimir | telnet makes a handy TCP port availability checker. :) |
| 19:01.05 | abel408 | Qwell: Sorry, but I'm not sure how to use that command. Never seen it before :/ |
| 19:01.08 | p3nguin | Too bad we use UDP for our VoIP stuff, huh? |
| 19:01.29 | capt_cassimir | And DNS. |
| 19:01.45 | p3nguin | To be fair, DNS uses both. |
| 19:01.54 | capt_cassimir | when you're doing an XFR. :) |
| 19:02.07 | p3nguin | or a lookup on a huge record. |
| 19:02.33 | p3nguin | It's the size that makes the difference, not the action. |
| 19:02.55 | capt_cassimir | oh, no shit, if it's over 512b it goes to TCP |
| 19:03.05 | capt_cassimir | TIL |
| 19:04.51 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 19:05.22 | abel408 | Is anyone willing to help me troubleshoot this? I don't mean to be annoying, but I am completely lost. If I can ping and connect to a trunk then why is asterisk showing it as unreachable? |
| 19:06.57 | p3nguin | What do you mean by "connect to a trunk" and how are you doing it? |
| 19:07.16 | abel408 | I also should mention that I have 2 other asterisk servers which shows this iax trunk status as OK. |
| 19:16.45 | *** join/#asterisk gxdssoft (~gxdssoft@190.81.6.83) |
| 19:17.06 | abel408 | p3nguin: What I meant by connect is that netowkring on both sides are working great. Not a single problem. |
| 19:17.23 | capt_cassimir | sounds kind of like a network problem though |
| 19:18.25 | p3nguin | How did you determine the unavailability of the IAX2 "trunk?" |
| 19:19.36 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 19:20.55 | abel408 | iax2 show peers |
| 19:23.29 | abel408 | shows that this particular trunk is "unreachable" while the other two are OK |
| 19:24.11 | *** join/#asterisk Azrael808 (~peter@5ada38a3.bb.sky.com) |
| 19:24.12 | abel408 | the qualify statement is what tells me if a trunk is OK or unreachable |
| 19:24.40 | ChannelZ | abel408: Anyone behind nat? |
| 19:24.41 | abel408 | and if I'm not mistaken, all the qualify statement does is send a ping request |
| 19:25.14 | abel408 | no, 2 different subnets but nothing is natted between them |
| 19:26.00 | capt_cassimir | Got your masks set right? |
| 19:26.32 | abel408 | Yup... Like I said. This was working great for about 2 months. Then all the sudden it stopped working |
| 19:27.08 | capt_cassimir | i bet your firewall admin threw a change in |
| 19:27.13 | abel408 | Could have been after a power outage, but I'm not certain |
| 19:27.25 | abel408 | I am the firewall admin |
| 19:27.35 | ChannelZ | I'd verify you have good communication between the two systems via UDP |
| 19:27.36 | capt_cassimir | Well, what have you been up to lately? |
| 19:28.08 | ChannelZ | Stop asterisk, use something like netcat to see if packets are making it where they should |
| 19:30.40 | abel408 | Haha, nothing. These 2 sites are on a p2p so I don't have a firewall setup between them. |
| 19:30.55 | capt_cassimir | Yeah, netcat is one of the only things i can think of to test UDP |
| 19:32.08 | abel408 | Well what baffels me is that the other 2 asterisk servers I have can call and connect to this iax trunk fine |
| 19:33.05 | ChannelZ | It could be a dozen things.. confused routers, DNS change sending packets off into nowhere... |
| 19:33.28 | capt_cassimir | Try netcat - run it as a listener on the server that hosts the "trunk", then send it data from the problem child and see if it gets there |
| 19:39.04 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:39.55 | *** join/#asterisk okei (bc81c390@gateway/web/freenode/ip.188.129.195.144) |
| 19:39.59 | okei | hellog guys, i have some questionn, if i dont need to enter password when i calling to mailbox to check mailboxs : d , what i need? only remove password section from voicemail.conf? |
| 19:40.35 | *** join/#asterisk hfb (~hfb@pool-96-247-108-172.lsanca.dsl-w.verizon.net) |
| 19:40.39 | Kobaz | what's the current way to run a terminal server with asterisk |
| 19:40.44 | Kobaz | is it still iaxmodem? |
| 19:41.12 | okei | p3nguin: i hope your help :P |
| 19:46.16 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 19:49.07 | *** join/#asterisk binbash_ (~peter@insley.demon.nl) |
| 19:50.48 | ChannelZ | okei: yes just remove the password from voicemail.conf but leave the , there followed by the remaining arguments |
| 19:51.51 | p3nguin | Or use the s option when calling VoiceMailMain(). |
| 19:54.05 | okei | p3nguin: i have one problem, when i trying to enter password, output is this 123456' for user '1001' (context = default |
| 19:54.23 | p3nguin | How is that a problem? |
| 19:54.23 | okei | i have not default context |
| 19:54.34 | p3nguin | Then you made a mistake in your dial plan. |
| 19:54.45 | okei | p3nguin: Incorrect password '123456' for user '1001' (context = default) |
| 19:55.10 | okei | i have't default context, and another configs |
| 19:55.11 | okei | wtf? |
| 19:55.13 | p3nguin | When calling VoiceMail() or VoiceMailMain() and you are specifying a mail box, you are SUPPOSED TO include the voice mail context. |
| 19:55.21 | p3nguin | Your mistake. Fix it. |
| 19:55.32 | okei | p3nguin: i have voicemail context |
| 19:55.42 | p3nguin | Good, then the problem is solved. NEXT! |
| 19:55.56 | okei | p3nguin: what's next? |
| 19:56.02 | p3nguin | You tell me. |
| 19:56.25 | okei | p3nguin: i have context in voicemail.conf and sip.conf mailbox=1001@test |
| 19:56.31 | okei | and nothing :/ |
| 19:56.36 | okei | when i trying to check |
| 19:56.36 | okei | mailbox |
| 19:56.40 | okei | password is incorrect |
| 19:56.43 | p3nguin | The voicemail context is "test"? |
| 19:56.48 | okei | yes |
| 19:56.58 | p3nguin | And you specified it as such in extensions.conf? |
| 19:57.22 | okei | p3nguin: hm yes, |
| 19:57.25 | ChannelZ | VoiceMailMain(1001@test) |
| 19:57.25 | p3nguin | Show me. |
| 19:57.30 | ChannelZ | etc |
| 19:57.33 | okei | ok wait |
| 19:58.13 | okei | p3nguin: http://pastebin.com/EMgg4SWw |
| 19:58.45 | p3nguin | exten => 250,1,VoiceMailMain(${CALLERID(num)}) |
| 19:58.54 | p3nguin | That will certainly fail. |
| 19:59.03 | okei | why? |
| 19:59.11 | okei | lol |
| 19:59.14 | p3nguin | Your caller ID number does not have a voicemail context on it. |
| 19:59.22 | okei | 1001 |
| 19:59.24 | okei | ? |
| 19:59.28 | Gugge | unless your callerid is 1001@test |
| 19:59.34 | Gugge | but that would be a strange callerid :) |
| 19:59.37 | p3nguin | If your caller ID is 1001, then you are calling VoiceMailMain(1001) which is 1001@default. |
| 19:59.58 | okei | ah sure |
| 20:00.02 | p3nguin | (1454.34) <p3nguin> Then you made a mistake in your dial plan. |
| 20:00.03 | okei | i need static entry yea? |
| 20:00.33 | Gugge | if you think you can trust your callerid, you can use VoiceMailMain(${CALLERID(num)}@test) |
| 20:00.38 | okei | exten => 250,1,VoiceMailMain(1001@test) |
| 20:00.39 | okei | right? |
| 20:00.49 | p3nguin | If all calls to 250 are going to be for the voicemail context of test, sure. |
| 20:01.08 | okei | yep yep i trying |
| 20:02.39 | okei | p3nguin: Gugge that's working thanks guys :) |
| 20:03.14 | p3nguin | I would have used something like exten => _86XXXX,n,VoiceMailMain(${EXTEN:2}@test) if all calls to 86XXXX are for the test voicemail context. Then to check 1001, you'd call 861001 from the phone. |
| 20:03.16 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-075-145-073.pools.arcor-ip.net) |
| 20:04.30 | p3nguin | You can use another prefix for every voicemail context. |
| 20:05.03 | okei | yes |
| 20:05.42 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 20:08.17 | JasonL | Hello.. I'm running into issues with asterisk voicemail.. while checking new messages it will sometimes stop accepting DTMF... the voicemail is still asking for input but its not getting my dtmf. If I hangup and call back it starts to work again. I googled it and found other people talking about that problem but so far no one posted a resolution. |
| 20:13.43 | p3nguin | Have you tried the various DTMF types? |
| 20:14.51 | *** join/#asterisk nmjnb (~nmjnb@c-567e72d5.026-18-73746f23.cust.bredbandsbolaget.se) |
| 20:17.17 | *** join/#asterisk garymc (~chatzilla@host109-145-110-205.range109-145.btcentralplus.com) |
| 20:18.08 | tyman | I have a user that when you call the user when what should play the unavail greeting (the unavail.wav exists) it plays Allison default. If I set his voicemail option to with the 'u' param, it plays the correct message. He has had a temp greeting before, but there is no longer a temp greeting showing in his mailbox. What could be the problem? |
| 20:22.32 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 20:23.10 | _Corey_ | JasonL: FWIW, I had some version of 1.4 installed somewhere last year that was doing that intermittently. It was resolved with an Asterisk upgrade. |
| 20:26.21 | jmls | hey! |
| 20:26.31 | jmls | where's my LANGUAGE gone ? |
| 20:26.34 | *** join/#asterisk devmikey (~irc@96.46.249.230) |
| 20:26.48 | jmls | playing with the * demo menu, try option 3 |
| 20:26.50 | jmls | bam! |
| 20:27.08 | devmikey | Help! My sip calls are failing |
| 20:27.14 | jmls | [Jul 18 21:24:42] ERROR[20206]: pbx.c:3588 ast_func_write: Function LANGUAGE not registered |
| 20:27.23 | jmls | how the hell do I register it, then ? |
| 20:27.41 | devmikey | Basically, it connects ok but then disconnects me in the middle of my call |
| 20:28.11 | devmikey | I can only log the client side |
| 20:28.33 | devmikey | Only thing unusual is: I get 2 Call 0 state=CALLING |
| 20:28.35 | devmikey | messages |
| 20:28.52 | devmikey | then i get 2 Call 0 state=EARLY messages |
| 20:29.26 | devmikey | But basically as soon as my client starts timing the call, it disconnects |
| 20:30.03 | ectospasm | ~thebook |
| 20:30.03 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 20:30.48 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 20:31.16 | nmjnb | Hi, I'm trying to change the keyboard layout through CLI but no luck so far. Any ideas? |
| 20:31.33 | raden | anyone ever use a2billing ? |
| 20:32.48 | raden | anyone play with polycom 335's ? |
| 20:32.57 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 20:35.36 | devmikey | HELLO??? |
| 20:43.07 | wdoekes2 | shouting always helps |
| 20:44.19 | *** join/#asterisk angryuser_laptop (~angryuser@80.214.8.7) |
| 20:45.18 | devmikey | I hope so |
| 20:45.24 | devmikey | I'm pretty pissed |
| 20:47.28 | garymc | How is everyone. Ive not been on for a while. Just like to say Hello :P |
| 20:50.43 | Bob_Pierce | raden: we use Polycom 335's |
| 20:50.57 | sunfone | Me too... lots of 335s |
| 20:51.04 | sunfone | great phone |
| 20:51.10 | *** join/#asterisk Syrex (~syrex@dsl-146-17-03.telkomadsl.co.za) |
| 20:51.11 | Bob_Pierce | Very nice phone! |
| 20:51.32 | sunfone | HD Voice enabled is unbelievable |
| 20:51.54 | _Corey_ | The LCD leaves a little to be desired. I think it's worth a few more bucks for the 450 |
| 20:52.16 | sunfone | Agree... depends on the deployment though |
| 20:52.27 | sunfone | A lot of our clients prefer the slimmer design |
| 20:52.36 | anonymouz666 | this phone is too expensive here |
| 20:52.47 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 20:52.56 | anonymouz666 | 335 supports two lines? |
| 20:53.04 | sunfone | Is there a 33x series with gigabit yet? |
| 20:53.12 | sunfone | that has been a recent problem |
| 20:53.41 | sunfone | yes, two lines on the 335 |
| 20:55.03 | anonymouz666 | almost $230 here |
| 20:55.08 | anonymouz666 | what about there? |
| 20:57.55 | _Corey_ | less than $150 |
| 20:58.11 | *** join/#asterisk radic (~radic@tmo-096-54.customers.d1-online.com) |
| 20:58.33 | anonymouz666 | inflation here then |
| 20:58.41 | _Corey_ | where are you? |
| 21:01.57 | anonymouz666 | br |
| 21:02.34 | _Corey_ | Probably taxes/duties |
| 21:02.52 | _Corey_ | I know a lot of people from BR who come here (USA) to buy stuff and take back in their luggage... ;) |
| 21:07.08 | *** join/#asterisk radic (~radic@tmo-096-54.customers.d1-online.com) |
| 21:07.48 | anonymouz666 | _Corey_: me too |
| 21:16.11 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 21:17.49 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 21:18.06 | neurosys | Is there a way in the CLI to see what time asterisk believes it is? |
| 21:19.04 | ChannelZ | it should believe whatever the system believes |
| 21:19.26 | ChannelZ | core show version |
| 21:20.07 | neurosys | ChannelZ: That would show m ethe time it was built :P |
| 21:20.35 | ChannelZ | oh yah |
| 21:21.27 | ChannelZ | NoOp(${STRFTIME(${EPOCH},,%d-%b-%Y %I:%M:%S %p)}) |
| 21:21.41 | ChannelZ | (first thing in my dialplan) |
| 21:24.00 | *** join/#asterisk ixx (~ixx@cpe-173-174-58-199.austin.res.rr.com) |
| 21:25.18 | *** join/#asterisk garryfre (~gfreemyer@mxgw.videomaker.com) |
| 21:32.32 | neurosys | <PROTECTED> |
| 21:43.53 | *** join/#asterisk esperanto (~rusty@p57ACCABA.dip0.t-ipconnect.de) |
| 21:44.23 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 21:44.39 | tyman | I've got a menu item (http://pastebin.com/v6d7YtzP) to be used a 3rd party virtual receptionist. Looking to save caller ring time when the called party is on the phone? The pasted option is not working. |
| 21:44.43 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 21:44.52 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 21:45.06 | neurosys | ugh bad /etc/localtime zoneinfo |
| 21:45.08 | neurosys | fixed. |
| 21:45.19 | tyman | s/used a/used by a/ |
| 21:45.31 | *** part/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 21:49.30 | tyman | the "TMA1" and "TMA2" globals correspond to the hard & soft phones, respectively, listed in line 5 |
| 21:49.34 | ChannelZ | tyman: have you just NoOp'd DEVICE_STATE to see what it's set to? And what version of asterisk are you running? |
| 21:49.48 | tyman | 1.8.5.0 |
| 21:50.20 | *** join/#asterisk MarKsaitis (~MarKsaiti@78-61-148-80.static.zebra.lt) |
| 21:50.43 | ChannelZ | Do you have callcounter turned on in sip.conf? |
| 21:51.07 | tyman | no...i'm looking that up now |
| 21:51.19 | tyman | is it required here? |
| 21:51.35 | ChannelZ | Pretty sure |
| 21:52.08 | tyman | should i be using extension_state with hints? |
| 21:53.02 | johnnyasterisk | Hi all, I am on a mac and trying to browse to issues.asterisk.org however it seems to require a local certificate to be validated.... anyone else have this issue? |
| 21:53.51 | WIMPy | Not exactely, but from this machine I cannot fetch anyting under /jira/ |
| 21:54.27 | WIMPy | But then my browser is not from this century. |
| 21:55.05 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 21:55.57 | johnnyasterisk | hmmmm - same on both safari and firefox - must try on my windows machine...... ughhhhh windows |
| 21:56.11 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 21:56.31 | tyman | ChannelZ: i've enabled callcounters...same result...asterisk thinks is false when my line is in use |
| 22:02.10 | ChannelZ | tyman: It works here. Either your TMA variables are not set to what you think they are and you're querying the wrong thing or there is some other error in your syntax I'm not seeing at this second |
| 22:03.11 | ChannelZ | RE do a NoOp and see what DEVICE_STATE is actually saying. Verbose on the console should show you what the variables are expanding out to |
| 22:04.18 | tyman | ChannelZ: i will do that right now... those globals are the same globals i use all over my dialplan so pretty absolute on those..i'm checking out a NoOp now |
| 22:04.39 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 22:05.02 | FLeiXiuS | What's everyones opinion on adhersion? An Asterisk ruby API |
| 22:05.07 | ChannelZ | Did you turn callcounter on in the [global] section of sip.conf to turn it on for everything (and reload) |
| 22:05.21 | ChannelZ | or [general] sorry not global |
| 22:06.35 | tyman | i only enabled within the sip section of my extensions...i will move it to general |
| 22:07.53 | ChannelZ | well it should work per peer but I prefer the shotgun approach when debugging and then go narrower as discoveries progress :) |
| 22:09.25 | tyman | ChannelZ: That was it...moving it to general fixed it |
| 22:10.04 | ChannelZ | hmm |
| 22:11.12 | ChannelZ | was it set before on the peers you wanted to actually monitor? |
| 22:19.36 | tyman | ChannelZ: set to the to template of the peers monitioring? (sorry for delayed response...phone call) |
| 22:19.46 | tyman | s/?/./ |
| 22:22.15 | ChannelZ | hmm. Dunno maybe it's a bug in that case |
| 22:23.00 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 22:26.20 | tyman | ChannelZ: Yes. That's what it appears to be. The "book" says it can be set in either [general] or peer specific sections of sip.conf |
| 22:28.31 | samandiriel | hiya! I'm runnning the latest pbx in a flash, and I'm having a problem with AGI that I'm hoping someone can shed some light on. To me it looks like a permissions issue, but it seems unlikely. I am trying to run a script in agi-bin that is owned by asterisk with permissions rwxrwxr-w. However, when I try to run it, it simply doesn't execute. However, if I move this script to /var/www/html and change the path in my dial plan, it |
| 22:28.31 | samandiriel | <PROTECTED> |
| 22:29.11 | *** join/#asterisk andres833 (~andres833@190.26.210.155) |
| 22:29.16 | samandiriel | I don't really want to store all my php agi scripts in /var/www/html :P |
| 22:32.26 | p3nguin | I've got an SPA-3102 that I am trying to use to take a call from the connected PSTN line and send it into Asterisk. I have configured the PSTN-to-VoIP gateway on the device and it answers the call and sends it into Asterisk... the problem is that the caller ID is not being sent along with the call. I have enabled the "PSTN CID for VoIP CID" option, but I'm still not getting it. I also set the answer delay to 6 seconds to ... |
| 22:32.32 | p3nguin | ... allow the CID to be received before sending the call in to Asterisk. Anyone have any idea why the CID is not making it to Asterisk through the device? |
| 22:34.27 | p3nguin | With a callerid value on the device in sip.conf, I get that caller ID num. Without a callerid set in sip.conf, I get the device's sip name as the caller ID num. |
| 22:34.52 | ChannelZ | samandiriel: What does 'core show settings' show your AGI dir as? |
| 22:35.25 | ChannelZ | and/or have you specified the entire path to the script when you run it? (that's what I do) |
| 22:36.08 | p3nguin | I even tried changing the answer delay to 12 seconds so there was PLENTY of time for the CID data to be present from the PSTN before sending the call into Asterisk. No change. |
| 22:36.30 | samandiriel | ChannelZ: it shows AGI Scripts directory: /var/lib/asterisk/agi-bin |
| 22:37.07 | samandiriel | ChannelZ: and unfortunately specifying the full path makes no difference. what's truly strange is that I have another script in there that runs no problem. however, if I copy it and rename it... it won't run |
| 22:37.10 | ChannelZ | p3nguin: you set the answer delay on the SPA or in Asterisk? |
| 22:37.50 | p3nguin | I set the SPA device FXO answer delay to 12 seconds to allow for it to "pick up" the CID data from the PSTN. |
| 22:38.02 | *** join/#asterisk shine (~stroll@lamantin.achamo.net) |
| 22:38.23 | p3nguin | Watching the attached CID box, I see the CID being displayed, and then a few seconds later, the SPA device answers and Asterisk takes over. |
| 22:38.32 | ChannelZ | p3nguin: hmm. It used to work fine on my 3102 but I don't have a POTS line anymore |
| 22:38.47 | p3nguin | I thought it used to work for me, too, but that would have been many asterisk versions ago. |
| 22:40.01 | p3nguin | The first line of the dial plan coming from the device is verbosely showing the caller ID num, so I expected to see the CID from the PSTN. |
| 22:40.24 | ChannelZ | Have you tried setting the CID Prefix to see if that is even coming in? |
| 22:40.34 | *** join/#asterisk JuStIcIa_ (~artur0@190.167.126.113) |
| 22:41.08 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 22:41.51 | p3nguin | I have not tried the CID prefix on the SPA. I will try that now. |
| 22:42.13 | *** join/#asterisk lcb (~Affiliate@unaffiliated/lcb) |
| 22:43.47 | p3nguin | It didn't do anything different. |
| 22:44.06 | ChannelZ | Also look on the User 1 page for 'CID Setting' and 'Block CID Setting' |
| 22:44.07 | lcb | Hi all |
| 22:44.24 | ChannelZ | I'm not sure if they control anything important in this case |
| 22:44.48 | p3nguin | CID setting: yes Block CID setting: no |
| 22:45.02 | ChannelZ | hmm |
| 22:47.44 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 22:48.40 | p3nguin | The caller ID pattern settings are blank. |
| 22:48.51 | ChannelZ | is this the same box that used to work? what firmware is it running? |
| 22:50.21 | p3nguin | I'm quite sure it used to give me the CID num off the PSTN, but it has been so long since I would have used it for that, I just don't really remember if it positively did give me the CID num or not. I have never changed the firmware since I got it. It has 3.3.6(GW) on it. |
| 22:51.17 | JuStIcIa_ | guys for set the ip phone nortel 1220 i need have the firmware on the asterisk and use the tftp on the same box ? |
| 22:51.39 | p3nguin | I kind of remember the last time I used it for this, I wasn't getting the CID num... then I read a couple forum postings that talked about turning up the answer delay on the device so it had time to get the CID right before the second ring. |
| 22:52.03 | ChannelZ | yeah I've had mine set at 3 and it worked for me |
| 22:52.08 | p3nguin | I figure it probably worked after I turned up the delay so there was time for it to arrive. |
| 22:52.27 | p3nguin | If I turn it down to 2, I don't see the CID show up on the analog CID box. |
| 22:52.34 | p3nguin | -NO DATA- |
| 22:53.09 | p3nguin | But turning it up to 6 gave it enough time, and I even went up to 12 seconds, which gave it a full ring after the CID popped up on the box. |
| 22:58.55 | p3nguin | I even changed the PSTN CID for VoIP CID to no and got no change in the CID num that Asterisk received. |
| 22:59.59 | p3nguin | Isn't the callerid value in the sip definition only supposed to be used if there is no other CID value from the phone/device? If I take out that line, the call is from the User ID of the device. |
| 23:00.46 | WIMPy | No. That makes Asterisk ignore what it receives. |
| 23:02.33 | p3nguin | Ah, okay. Well, with that value set, asterisk is obeying it and it says the call is from that number. Without that set, the call comes from the device's User ID. |
| 23:02.49 | WIMPy | yes |
| 23:03.05 | *** join/#asterisk bchia (~Adium@24.42.227.160) |
| 23:03.23 | p3nguin | I don't want to see the device's user id as the caller id num... I expect to see the caller's caller id num. |
| 23:04.08 | WIMPy | Have you looked at a sip debug? |
| 23:05.39 | p3nguin | Yes, I have. The PSTN caller ID number value is not present in the sip debug anywhere that I saw. |
| 23:09.49 | p3nguin | I'm going to debug again just to double check it. |
| 23:11.15 | JuStIcIa_ | some one know about Nortel IP Phone 1220 for Asterisk ? |
| 23:12.32 | p3nguin | wimpy: Okay, I see the PSTN CID showing up in sip debug. It shows in the remote party ID. Do I need to trustrpid in the sip definition? |
| 23:13.08 | WIMPy | Possibly. But you know, I'm not the SIP guy. |
| 23:13.40 | p3nguin | Are you the PRI guy? |
| 23:14.03 | WIMPy | That's much more to my taste, yes. |
| 23:14.07 | p3nguin | Just asking for future reference. |
| 23:14.43 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:16.08 | *** join/#asterisk shine (~stroll@lamantin.achamo.net) |
| 23:18.12 | p3nguin | This is very puzzling to me. |
| 23:19.06 | p3nguin | Is there any way to use templating in sip.conf where the template forces callerid and the child can nullify the callerid setting? |
| 23:19.11 | garryfre | permissions difference? |
| 23:19.54 | garryfre | hopes they ignore me . .. after replying and THEN noticing I forgot to scroll down. |
| 23:20.09 | WIMPy | I've never tried "callerid=". |
| 23:23.16 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 23:24.54 | p3nguin | Okay, I'm glad I got that sorted. My sip peer template was interfering with the caller ID number from the PSTN. |
| 23:25.08 | p3nguin | Caller ID from the PSTN is, in fact, working correctly. |
| 23:29.29 | ChannelZ | oopsies |
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| 23:36.04 | ChannelZ | stares at the clock |
| 23:38.14 | ChannelZ | which aparently causes netsplits |
| 23:38.53 | WIMPy | Do not stare at that clock again. |
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| 23:51.34 | carrar | pokes Katty for no particular reason. |
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