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| 00:04.25 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 00:55.52 | braden80 | anyone familiar with "prodding channel 'XXXX' failed" error? can't make outgoing calls |
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| 02:52.49 | volga629 | Hello Everyone, how in pjsip enable Path header support ? |
| 02:53.21 | volga629 | I don't see anything in configuration |
| 02:53.50 | volga629 | based on this report should be fixed |
| 02:53.52 | volga629 | v |
| 02:53.54 | volga629 | https://issues.asterisk.org/jira/browse/ASTERISK-21084 |
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| 03:03.02 | IRCJOE | hello |
| 03:03.17 | IRCJOE | guys a good book to learn not only i ma a book person.. |
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| 03:44.48 | [TK]D-Fender | ? book |
| 03:44.56 | [TK]D-Fender | ~book |
| 03:44.56 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 03:44.57 | [TK]D-Fender | ^^^^^ |
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| 04:55.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 07:17.24 | jzu_ | when I'm forwarding a call from my deskphone to my mobile should "sendrpid=yes" send the remote caller id to my mobile? |
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| 07:22.43 | bobbymc | i got a question about meetme app |
| 07:23.04 | bobbymc | there is a option to customize the intro message with a Q flag but im not sure how to set that in the configs |
| 07:23.07 | bobbymc | can someone please help me? |
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| 07:48.48 | bobbymc | anyone home? |
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| 07:57.12 | jzu_ | hmm, right |
| 07:57.35 | jzu_ | my Cisco 7970 send busy when it does Call Forward that is activated from the phone button |
| 07:57.45 | jzu_ | is there a way to do call forward through Asterisk? |
| 07:57.53 | jzu_ | * something? |
| 08:11.29 | *** join/#asterisk areski (~areski@80.174.128.96.dyn.user.ono.com) |
| 08:14.31 | ChannelZ | bobbymc not sure what 'intro message' you are talking about. MeetMe is old and I don't have it compiled anymore so I don't have the syntax handy |
| 08:15.05 | bobbymc | hey i know but my app depedns on it its a open source php program called vicidial |
| 08:15.21 | bobbymc | i looked at the docs and it tells me there is a G flag that lets you set it |
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| 08:15.32 | bobbymc | but i dont get how, may i paste the link tot he doc here? |
| 08:15.59 | ChannelZ | jzu_, what do you mean it 'sends busy' when doing call forwarding? Probably the number you're trying to forward to is invalid and it's actually being rejected on the asterisk side |
| 08:16.32 | ChannelZ | bobbymc, yeah |
| 08:16.50 | bobbymc | ChannelZ https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_MeetMe is the link and it shows option G with what im talking about |
| 08:17.25 | jzu_ | ChannelZ: the call goes through, my own mobile starts ringing but the original call goes to voicemail on Asterisk |
| 08:17.29 | ChannelZ | exten => 123,1,MeetMe(1,G(somefile)) |
| 08:17.52 | jzu_ | ChannelZ: that's funny behavior |
| 08:17.55 | bobbymc | thank you ChannelZ =) |
| 08:18.54 | ChannelZ | jzu_, too short of a timeout on the Dial? |
| 08:19.12 | jzu_ | the problem could also be that I'm trying to send the call to a forwarded number through the same number that it comes in from |
| 08:19.32 | ChannelZ | uhm.. huh? |
| 08:19.59 | jzu_ | my external number from Localphone is 123, my internal phone is 456 and I'm trying to forward to external number 789 |
| 08:20.24 | jzu_ | call comes in from 123 to 456 which is behind my Asterisk, then I try to forward to 789 using my external number 123 which is already handling incoming call |
| 08:21.30 | ChannelZ | Well it doesn't really work like that, unless it was an analog phone line |
| 08:21.50 | jzu_ | yeah, it's all SIP in/out |
| 08:22.42 | ChannelZ | Need to see verbose console output of one of these forwarded calls that goes wrong, at minimum |
| 08:23.27 | jzu_ | yeah, I'll do that later today |
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| 08:27.51 | bobbymc | channelZ worked like a charm |
| 08:27.54 | bobbymc | thank you again |
| 08:28.19 | ChannelZ | To answer your actual question, yes you could build your own call forwarding with asterisk, it just takes constructing a bunch of dialplan. There's FollowMe |
| 08:28.27 | bobbymc | i was wondering if you have any experience using asterisk in a linux container or vm? |
| 08:28.32 | ChannelZ | bobbymc no prob |
| 08:28.38 | bobbymc | the biggest issue i encounter is timing |
| 08:28.38 | ChannelZ | Not personally, no |
| 08:29.08 | ChannelZ | Yes because MeetMe is old. What version of asterisk? |
| 08:29.11 | bobbymc | linux containers solve that issue because you can access dahdi but with xen server for example you cant |
| 08:29.22 | bobbymc | 11.15.0 |
| 08:29.52 | bobbymc | i was about to rewrite the app to use confbridge but it looks like that also needs a timing device |
| 08:30.16 | ChannelZ | Well there are software-based timing sources now but MeetMe still uses DAHDI for mixing |
| 08:30.34 | bobbymc | what software based timing sources? |
| 08:30.43 | ChannelZ | You might need to play with which timer works best, pthreads or timerfd |
| 08:32.10 | ChannelZ | It's probably using one of them already, I think dahdi_dummy (the old virtual timing device for machines without hardware) went away some time ago |
| 08:32.47 | bobbymc | yea dahdi_dummy become part of dahdi |
| 08:33.03 | bobbymc | they build it directly into it i believe |
| 08:42.00 | jzu_ | bobbymc: I run Asterisk currently in SmartOS KVM |
| 08:42.03 | jzu_ | bobbymc: works like a charm |
| 08:49.13 | bobbymc | smartOS kvm let me check it out |
| 08:49.30 | bobbymc | how do you access timing? |
| 08:51.28 | bobbymc | jzu_: do you have any iax trunks or meetme rooms? |
| 08:51.47 | bobbymc | either require good timing and my biggest issue with virtualizing asterisk is the timing problem |
| 08:53.51 | jzu_ | bobbymc: IAX2 configured, never tested it yet actually |
| 08:54.19 | ChannelZ | dump MeetMe. Into the trash where it belongs. ConfBridge can also run parallel to MeetMe so you can make a test conference and try it |
| 08:54.57 | ChannelZ | and on that bombshell |
| 08:54.59 | ChannelZ | goes to bed |
| 08:55.07 | bobbymc | yea i dont mind switching to confbrige but im worried the timing devices require kernel timing like dahdi does which make it the same problem i got with meetme |
| 08:55.40 | bobbymc | timerd i belive depends on the kernel timing |
| 08:56.06 | bobbymc | *timerfd |
| 08:58.04 | jzu_ | bobbymc: I'm able to test it in May :-) |
| 08:58.19 | bobbymc | test what exacly? |
| 08:58.22 | jzu_ | bobbymc: now IAX2 Trunk is just within my local LAN, I'm installing the second Trunk on remote site in May |
| 08:58.32 | jzu_ | bobbymc: how IAX2 works from KVM |
| 08:58.45 | bobbymc | the biggest issue you run into is when you have high load |
| 08:58.50 | jzu_ | bobbymc: the other side is KVM Linux on SmartOS and other side is Raspberry Pi running Linux :) |
| 08:59.06 | bobbymc | it may work fine with a few calls but quality goes down when you get high load |
| 08:59.11 | jzu_ | well, it will be one call at a time, it's "internal" phone system between my place and my parents |
| 08:59.25 | jzu_ | so I bet it will be fine |
| 08:59.25 | bobbymc | haha you wont have issues with that then =) |
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| 09:10.59 | CyberBob | Hey All, May I know what is this error for? ---->> [C-00009f94]: channel.c:4860 ast_prod: Prodding channel 'SIP/192.168.3.9-0000bbf6' failed ---->> [C-00009f89]: channel.c:4860 ast_prod: Prodding channel 'SIP/1007-0000bbe4' failed |
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| 09:28.49 | s4va | Hello to everyone! Could you please help me to get what kind of hardware i need for small bussiness office with 2-5 phones (12 calls/hr each)? Sorry for my english |
| 09:29.27 | s4va | Forgot. I want to use Asterisk |
| 09:36.10 | bobbymc | nohting special |
| 09:36.19 | bobbymc | a single quad core |
| 09:36.22 | bobbymc | at most should do |
| 09:36.27 | bobbymc | 2 gigs of ram |
| 09:37.08 | bobbymc | s4va: any basic server should do fine |
| 09:37.39 | s4va | thank you! |
| 09:45.52 | bobbymc | anyone used timerfd in a virtualized environment ? |
| 09:46.03 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-125.wall.uls.co.za) |
| 09:49.08 | jkroon | hi guys, i'm seeing a very weird one, automatic call forward, setup involves two servers, with an IAX/2 between them. Server A receives a call, Dial(IAX2/B/${EXTEN}) ... B receives it and in the dialplan forwards back to Dial(IAX2/A/${DIFFERNT_EXTEN}) ... On A transfer=no, on B I've tried with transfer=yes/no/mediaonly (I actually want mediaonly all over). When the call gets answered (A now passes it to SIP/another/${EXTEN}) the |
| 09:49.08 | jkroon | second channel pair on A goes Up, both channels on B goes Up, but the channel pair for the initial forwarded call remains Down/Ringing. |
| 09:59.31 | *** join/#asterisk CyberBob (~DC@bba429145.alshamil.net.ae) |
| 09:59.39 | CyberBob | Hey All, May I know what is this error for? ---->> [C-00009f94]: channel.c:4860 ast_prod: Prodding channel 'SIP/192.168.3.9-0000bbf6' failed ---->> [C-00009f89]: channel.c:4860 ast_prod: Prodding channel 'SIP/1007-0000bbe4' failed |
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| 10:57.08 | WIMPy | Hi jkroon. Long time no see. I haven't actually looked at the state, but I have the impression that mediaonly doesn't work. On the intermediate box the call always ands after some rathr random time |
| 10:57.15 | WIMPy | <1 Minute. |
| 10:57.27 | WIMPy | Not matter if it's still up or even if it was ended earlier. |
| 10:57.36 | jkroon | WIMPy, that's bad news. but even with transfer=no on both sides ... |
| 10:57.53 | jkroon | and iax2 debug on B still shows transfer stuff going on, so i'm trying to gather more info. |
| 10:58.17 | WIMPy | Maybe the lopo detection does something bad there? |
| 10:58.23 | jkroon | WIMPy, may be related to the fact that one of the call legs doesn't get answered and eventually ringing times out? |
| 10:58.56 | WIMPy | Or the transfer option is only for avtively initiating transfers, not for accepting them. |
| 10:59.06 | jkroon | on B both goes Up, but the initial channel on A remains down ... in particular, i also get hangups after random period (initial report), but if I phone without a timeout from the origin then the call stays up indefinitely. |
| 10:59.08 | WIMPy | s/lopo/loop/ |
| 10:59.38 | jkroon | read as such :p |
| 11:00.25 | WIMPy | Oh. So you thik it's the timeout of the Dial that determines when the call on the intermediate box ends? |
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| 11:14.49 | jkroon | jip. |
| 11:14.57 | jkroon | i can hang up from either end and it works. |
| 11:15.11 | jkroon | for some reason B is not sending the ANSWER, or A is discarding it. |
| 11:15.25 | jkroon | looks like the latter. |
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| 11:35.27 | jkroon | WIMPy, a second pair of eyes would be appreciated but it indeed looks like the latter: http://pastie.org/10067041 |
| 11:35.52 | jkroon | check around lines 167. |
| 11:39.53 | WIMPy | Do you have an idea what that 255 is? |
| 11:40.10 | jkroon | http://pastie.org/10067053 - different call, but taken on A, line 149 the ANSWER gets received. |
| 11:40.20 | jkroon | no idea, was heading for the code now. |
| 11:42.44 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 11:43.43 | jkroon | so afaik those CONTROL sub-class values match the asterisk AST_CONTROL_* defines. |
| 11:44.17 | WIMPy | I just check and I don't have an issue with Answer. But it's a "normal" 3 box thing. Will check if I can do it back to the same box. |
| 11:44.18 | jkroon | include/asterisk/frame.h doesn't have it defined/declared. |
| 11:44.59 | jkroon | ok, my exact call scenario here is SIP endpoint => P, talks IAX2 with A, talks IAX2 with B, talks IAX2 back with A, forwards to SIP. |
| 11:45.30 | jkroon | hmm, what's the chance that the 255 comes from a -1 error return value that isn't being handled? |
| 11:46.01 | WIMPy | Might fit. |
| 11:46.10 | jkroon | seen strange ones like that before. |
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| 11:48.05 | WIMPy | Nope. From A to B and back to A seems to do the same. |
| 11:48.23 | jkroon | meaning yours also doesn't answer? |
| 11:48.49 | WIMPy | Are you doing any manual Progrss() or such? |
| 11:48.56 | WIMPy | It does answer. |
| 11:49.02 | jkroon | no. |
| 11:49.20 | WIMPy | It's all ok, ecept for the CDR on B. |
| 11:51.05 | jkroon | ast version? |
| 11:51.10 | WIMPy | Hmm. Actually, this time the CDR on B was right. |
| 11:51.20 | WIMPy | 11.16.0 |
| 11:51.26 | jkroon | never had issues on B with respect to CDR, only on A. |
| 11:52.56 | WIMPy | On A? But that would keep an active call. |
| 11:53.22 | jkroon | well, there are two CDR entries on A, the first goes NOANSWER, the second is correct. |
| 11:54.01 | WIMPy | I can't reproduce that noanswer thing so far. |
| 11:54.43 | jkroon | hmm, interesting. |
| 11:54.53 | jkroon | perhaps that's already fixed upstream. |
| 11:55.20 | WIMPy | What version are you using? |
| 11:58.10 | WIMPy | Hmm. When I try A>B>C the CDR on B is wrong again. Looks like theres something to be discovered... |
| 11:58.57 | jkroon | some handling somewhere is foobar. |
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| 12:00.39 | WIMPy | Interesting. It only seems to happen when dundi is used. |
| 12:02.05 | WIMPy | wonders where that makes the difference. |
| 12:02.51 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:04.10 | jkroon | dundi ... i don't even have dundi loaded on B, and DUNDi isn't involved for these particular calls. |
| 12:04.21 | jkroon | hmm, perhaps I just need to double-check that statement ... |
| 12:05.29 | WIMPy | It's still only the CDR on the intermediate box that goes out of the media path. Otherwise the call still works as expected. |
| 12:10.02 | jkroon | yea, no dundi loaded on B. can't find any reference to DUNDi on A for this particular call (A will ask DUNDI if, and only if, it can't find the destination locally) |
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| 12:12.40 | jkroon | WIMPy, if I add an explicit Answer() on B before the Dial() everything works ... but the call goes Up early. |
| 12:12.56 | jkroon | at least it's a short-term solution for my client right now. |
| 12:13.16 | jkroon | definitely NOT ideal though. |
| 12:13.27 | WIMPy | Maybe a Progress() or Proceeding() would also help? |
| 12:13.41 | WIMPy | dundi on the 2nd call leg doesn't seem to do any harm. |
| 12:13.50 | eduzimrs | Hi guys, i`ve changed the audio that was being played on a queue (periodic-announce) reload the module, and still plays the old audio, any ideas? |
| 12:14.29 | WIMPy | So I only get a random duration in the CDR od B when the call goes from A to B because of a dundi switch. |
| 12:15.54 | jkroon | interesting idea... let's try. |
| 12:16.20 | jkroon | what's the difference between Progress and Proceeding btw? |
| 12:18.27 | WIMPy | Proceeding means that the call number looks valid and the calll is being forwarded (No more dialling allowed). Progress can mean about anything including the setup of early media. |
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| 12:19.15 | jkroon | either way, no go. |
| 12:19.34 | jkroon | proceeding relates more to overlap dialing then. |
| 12:19.55 | jkroon | WIMPy, different issues then. |
| 12:20.27 | WIMPy | Like SIP "Trying" I guess. |
| 12:20.34 | WIMPy | Yes, they seem to be different issues. |
| 12:28.03 | jzu_ | so yes, does this seem like busy being triggered because no answer happened within time? |
| 12:28.29 | jzu_ | http://pastebin.com/SVB2auhW <-- I'm doing call forward to my mobile, I guess the 20s timer start from the moment the call arrives my Asterisk and before it even is forwarded to my mobile |
| 12:28.53 | jzu_ | so I should increase the timer I guess, to 60s or so for the call forwarding not to go to Busy and Voicemail :) |
| 12:29.21 | eduzimrs | Hi guys, i`ve changed the audio that was being played on a queue (periodic-announce) reload the module, and still plays the old audio, any ideas? |
| 12:29.25 | WIMPy | Which voicemail? |
| 12:29.46 | eduzimrs | an asterisk restart is needed? |
| 12:30.21 | jzu_ | WIMPy: voicemail of the internal extension that forwards the call |
| 12:30.23 | jzu_ | WIMPy: SIP/100 |
| 12:30.55 | jzu_ | WIMPy: call arrives from external number through a SIP Trunk, it's destinated always to SIP/100 which is Cisco phone |
| 12:31.08 | jzu_ | WIMPy: I've done cfwdall on the Cisco phone to my mobile |
| 12:31.49 | WIMPy | Oh, on the phone. Yes, that would still count as the initial ringing then. |
| 12:31.54 | jzu_ | yeah |
| 12:32.46 | jzu_ | is dialplan for call forward how hard to do? one that would trigger on/off the forward |
| 12:33.00 | jzu_ | or maybe I'll just adjust ring time of that internal extension :) |
| 12:33.37 | WIMPy | The point is that dialplan works even if the phone doesn't or isn't reachable. |
| 12:33.43 | jzu_ | yeah :) |
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| 12:49.09 | anonymouz666 | good morning! |
| 12:49.44 | anonymouz666 | sends a whatsapp call (using OPUS) to coppice |
| 12:49.50 | anonymouz666 | he he |
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| 13:54.36 | Acilim | Hello all, i am trying to make outbound calls with asterisknow and cisco 2811. is there anyone has any experience with this configuration |
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| 13:58.44 | [TK]D-Fender | What problem are you having? |
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| 14:02.06 | Acilim | Hello Fender, I am trying to find document how to configure them |
| 14:03.06 | Acilim | do I need to add trunk in asterisk? |
| 14:03.12 | *** join/#asterisk mjt (~mjt@isrv.corpit.ru) |
| 14:03.31 | [TK]D-Fender | Clearly. |
| 14:04.04 | [TK]D-Fender | Asterisk won't know about your Cisco unless you configure it to... and then configure your calls to use that trunk. |
| 14:05.05 | Acilim | I am also little bit confused about #dial-peer voice configuration |
| 14:05.30 | Acilim | in cisco config, i didn't get the point how it works |
| 14:05.40 | *** join/#asterisk mjordan (mjordan@nat/digium/x-ihposuyryzpaetfs) |
| 14:05.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:05.56 | [TK]D-Fender | Get your * side right first |
| 14:06.13 | Acilim | * side? |
| 14:07.12 | anonymouz666 | would suggest the book |
| 14:07.43 | *** part/#asterisk bulkorok (~Benjamin_@85.183.61.47) |
| 14:08.18 | mjt | how to bind iax listener to a single IP address? I've set bindaddr=a.b.c.d in [general] in iax2.conf, the same as I did in sip.conf, but it still binds to *... |
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| 14:08.59 | [TK]D-Fender | * = Asterisk |
| 14:09.15 | Acilim | ahh ok. |
| 14:09.32 | [TK]D-Fender | mjt: There is no "iax2.conf" |
| 14:09.58 | mjt | yeah. it is iax.conf not iax2.conf |
| 14:10.25 | [TK]D-Fender | Show us your actual config |
| 14:10.28 | [TK]D-Fender | ~pb |
| 14:10.29 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:10.30 | [TK]D-Fender | ^^^ |
| 14:11.04 | mjt | http://paste.debian.net/164296/ |
| 14:12.56 | [TK]D-Fender | And the listen dump & interface listing... |
| 14:13.14 | mjt | listen dump? |
| 14:13.54 | mjt | # netstat -tulnp | grep asterisk |
| 14:13.54 | mjt | udp 0 0 0.0.0.0:4569 0.0.0.0:* 992/asterisk |
| 14:13.57 | mjt | udp 0 0 192.168.19.1:5060 0.0.0.0:* 992/asterisk |
| 14:14.00 | mjt | this one? :) |
| 14:14.55 | [TK]D-Fender | yes |
| 14:15.25 | [TK]D-Fender | And now ifconfig <- |
| 14:15.31 | mjt | that will be LARGE |
| 14:16.10 | mjt | 6: eth-lan: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc noqueue state UP group default |
| 14:16.13 | mjt | <PROTECTED> |
| 14:16.25 | mjt | that's one of about 40 interfaces |
| 14:16.54 | [TK]D-Fender | have you restarted * or reloaded the module entirely? |
| 14:17.00 | [TK]D-Fender | a config reload won't do it |
| 14:17.07 | mjt | yes, restarted it just a minute ago once more |
| 14:17.28 | mjt | udp 0 0 0.0.0.0:4569 0.0.0.0:* 14404/asterisk |
| 14:17.44 | [TK]D-Fender | Fill in bindport for good measure |
| 14:18.31 | [TK]D-Fender | mjtthat's one of about 40 interfaces <- actually.. show these... |
| 14:18.36 | [TK]D-Fender | I don['t care about "big" |
| 14:18.40 | [TK]D-Fender | I'll filter through... |
| 14:18.43 | mjt | udp 0 0 0.0.0.0:4568 0.0.0.0:* 17503/asterisk |
| 14:18.55 | mjt | i've set port to 4568 (not 9) |
| 14:19.04 | [TK]D-Fender | I suppose you changed it just to make thee diference obvious? |
| 14:19.10 | mjt | yup |
| 14:19.18 | [TK]D-Fender | ok, PB the full interface listing. |
| 14:19.18 | mjt | heh |
| 14:19.25 | [TK]D-Fender | you can PM it if you like |
| 14:19.26 | mjt | it is the typo |
| 14:19.33 | mjt | bindaddr = 192.168.19.2 |
| 14:19.39 | mjt | it is *1* not 2 |
| 14:19.40 | [TK]D-Fender | (still a PB.. don't flood in private either |
| 14:19.47 | [TK]D-Fender | I was suspecting that :) |
| 14:19.59 | [TK]D-Fender | bbut you said you had 40 so I was going to wait and see :) |
| 14:20.26 | mjt | udp 0 0 192.168.19.1:4569 0.0.0.0:* 18532/asterisk |
| 14:20.29 | mjt | ;) |
| 14:20.38 | [TK]D-Fender | Glad you've got it sorted... |
| 14:20.50 | mjt | thank you for the patience. It's a great shame for me |
| 14:21.21 | mjt | tho, I'd expext some sort of error message... ;) |
| 14:21.33 | mjt | (so I don't feel THAT ashamed :) |
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| 14:46.26 | avb | hey guys, is there is any ways to use variables i got in readsql in the writesql query with func_odbc? |
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| 16:13.44 | *** join/#asterisk Mango45 (~Mango45@d209-89-210-219.abhsia.telus.net) |
| 16:14.42 | Mango45 | I have received a call, and I would like to transfer it to a predetermined context,extension,priority when I press a DTMF digit. |
| 16:15.09 | Mango45 | I tried to use a dynamic feature like such: xfr => 5,peer/callee,Goto,"context,208,1" |
| 16:15.39 | Mango45 | That did not work. The CLI said "Goto (context,208,1)", but nothing happened on the call. |
| 16:15.52 | Mango45 | What did I do wrong please? :) |
| 16:32.14 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 16:33.35 | [TK]D-Fender | Mango45: Just because you execute something ... doesn't mean your call ENDS |
| 16:34.59 | *** join/#asterisk eppigy (~Dave@snugglenets.com) |
| 16:35.02 | eppigy | hello |
| 16:35.05 | eppigy | I am dave |
| 16:35.50 | [TK]D-Fender | yes |
| 16:35.57 | [TK]D-Fender | you are dave |
| 16:36.19 | pjensen00 | I doubt he's dave. |
| 16:36.33 | eppigy | thank you |
| 16:36.34 | simonpatapon_lun | i've seen a ot of daves |
| 16:36.44 | eppigy | simonpatapon_lun: YOU HAVE SEEN NOTHING |
| 16:36.53 | pjensen00 | It is April 1st. Eppigy is clearly not Dave. |
| 16:37.14 | eppigy | Very shrewd... |
| 16:37.43 | pjensen00 | You cannot throw something over my head. I am too fast. I will catch it. |
| 16:38.00 | eppigy | I thought mcdonalds doing breakfast all day was an Aprils fools joke |
| 16:38.05 | eppigy | luckily it is not |
| 16:38.20 | [TK]D-Fender | https://www.youtube.com/watch?v=JF1chLj1fro |
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| 16:46.45 | volga629 | Hello Everyone Is NOTIFY should preserve Record-Route from first SUBSCRIBE ? |
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| 17:13.30 | Mango45 | <[TK]D-Fender> Mango45: Just because you execute something ... doesn't mean your call ENDS |
| 17:13.35 | Mango45 | I see. What should I do instead? |
| 17:15.18 | [TK]D-Fender | Call a script that does an AMI redirect, eetc on the target channel |
| 17:15.49 | Mango45 | Cool, thanks. |
| 17:18.43 | *** join/#asterisk Demon_VoIP (~demon@ip253.net222.n37.ru) |
| 17:19.24 | Looserouting | hi. is there a diaplan function to read offered codecs in an inbound leg ? i would like to use these exact codec for the outbound leg useing SIP_CODEC_OUTBOUND |
| 17:23.02 | Mango45 | As far as I am aware, no. However, you can hack a similar result: |
| 17:24.41 | Mango45 | Actually, SIPPEER may do what you want. |
| 17:26.12 | Looserouting | looks good. i'll try it. Thank you |
| 17:26.38 | Mango45 | If that doesn't work, come back. |
| 17:27.10 | Looserouting | I think this is the exact codec list as in sipf.conf |
| 17:27.32 | [TK]D-Fender | It won't |
| 17:27.46 | [TK]D-Fender | that lists what * offers.. not what the other side offered |
| 17:27.53 | Mango45 | Ok. In that case: |
| 17:28.11 | Mango45 | same => n,NoOp(${IF($[${REGEX("g722" ${SHELL(/usr/bin/rasterisk -x "sip show channel ${SIPCALLID}" | grep "Their Codec")})}]?g722 supported:g722 unsupported)}) |
| 17:36.42 | Looserouting | hmmm... i'll take a look at the code. Maybe I can make something like CHANNEL(theircodecs) |
| 17:37.32 | Looserouting | i appriciate your help |
| 17:37.59 | Mango45 | Ah-ha. I can use a macro for my issue. |
| 17:38.05 | Mango45 | with ChannelRedirect. |
| 17:41.26 | putnopvut | Has anyone ever had an issue with PJSUA python tests in the testsuite always sending SIP 488 responses to calls? If so, how did you fix it? |
| 17:41.48 | putnopvut | I now have two dev boxes that are doing this, and it's incredibly frustrating. |
| 17:42.31 | putnopvut | Specifically, I have Asterisk placing an outbound call to a PJSUA instance. I have an account callback that is supposed to answer with a 200 OK response, but PJSUA sends 488 instead. |
| 17:45.47 | mjordan | and I'm assuming it doesn't say why? |
| 17:46.18 | putnopvut | No, the error on the exception thrown is just "Not Acceptable Here" which is the status text that goes along with a 488 |
| 17:46.35 | putnopvut | experiments more |
| 17:48.03 | anonymouz666 | putnopvut: usually 488 regards to codec offer, isn't? |
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| 17:48.53 | putnopvut | anonymouz666: yep. The issue here is that I had a working test, then I did something to screw up my PJPROJECT installation, so I reinstalled it (thinking I reinstalled it the same way as it was before), and after reinstallation, the previously-working test has 488s every time. |
| 17:49.20 | mjordan | putnopvut: I wonder if pjproject thinks that all the codecs should be provided externally |
| 17:49.25 | putnopvut | So my assumption here is that there is something different about the PJPROJECT installation, but I don't know what it is. I was hoping that someone who had experienced this before would have an answer. |
| 17:49.26 | mjordan | as that is usually the case when built for Asterisk |
| 17:50.22 | putnopvut | Well, my initial thought was that I may have left out the --with-external-gsm flag, but if I do specify that, then configure errors out because I don't have libgsm. So what that tells me is that I didn't have that configure flag option specified previously. |
| 17:50.45 | mjordan | yeah |
| 17:51.52 | putnopvut | My second thought was that maybe when I initially installed PJPROJECT, I didn't specify "--disable-sound" when configuring. But leaving that out leads to a total wtf error, telling me to recompile with -fPIC. I compile with -fPIC and I see the same error still. |
| 17:52.36 | putnopvut | <PROTECTED> |
| 17:53.14 | putnopvut | So the disabled sound may or may not be the issue, but trying to compile without --disable-sound is being a PITA |
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| 18:02.05 | Demon_VoIP | Hello. Asterisk 13.2. How to get multiple lines Message(body) via AGI? |
| 18:02.15 | Demon_VoIP | https://issues.reucon.com/browse/AJ-47 This is the verdict? |
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| 18:04.05 | Demon_VoIP | Sorry, issue is from foreign tracker. |
| 18:04.33 | [TK]D-Fender | Asterisk variables alone have a length limit of 255 or some such.. |
| 18:04.45 | [TK]D-Fender | And yes, escaping would be required |
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| 18:06.40 | Demon_VoIP | [TK]D-Fender, escaping on GET? Escaping (or base64) variable and putting it by agi call param.. |
| 18:09.34 | Demon_VoIP | And why asterisk still does not have support for offline messaging? in addition strongly inhibits its simple implementation |
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| 18:32.39 | mjordan | Demon_VoIP: er... what do you mean by "offline messaging"? |
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| 18:36.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:39.37 | Demon_VoIP | mjordan, endpoint is offline and have incoming SimpleMessage... The message is lost? |
| 18:41.06 | Demon_VoIP | mjordan, for example, ejabberd has modules for offline message storage. |
| 18:41.15 | robmal | Anyone here using asterisk calendar integration? I can't figure out why my radicale doesn't show all events. |
| 18:42.15 | Demon_VoIP | mjordan, And now I realize my desired behavior through the storage of messages sent was not successful. And sending messages when endpoing is logged on the server |
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| 19:07.09 | marceloamorim | I don't know if here is the right place to ask this, but here we go. I'm trying to build chan_dongle from googlecode and I had problem with aclocal |
| 19:18.52 | mjordan | Demon_VoIP: You're basically talking about building a 'voicemail' for text messages. That's all fine and dandy, but I wouldn't build that in Asterisk. |
| 19:19.15 | mjordan | Demon_VoIP: If you want to have some notion of store & forward, I'd just use ARI |
| 19:19.25 | *** join/#asterisk v0ip-wolf (~wilfred@2610:1c0:0:2:1d60:b92f:1b1e:5de2) |
| 19:20.17 | mjordan | You can listen for received text messages by subscribing to the endpoints resource - at which point, you'll get TextMessageReceived events |
| 19:20.18 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+REST+Data+Models#Asterisk13RESTDataModels-TextMessageReceived |
| 19:20.29 | mjordan | And - if the destination endpoint isn't available, you store them |
| 19:20.33 | *** join/#asterisk areski (~areski@80.174.128.96.dyn.user.ono.com) |
| 19:20.45 | mjordan | and then send them laster using the sendMessage operation: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Endpoints+REST+API#Asterisk13EndpointsRESTAPI-sendMessage |
| 19:20.57 | mjordan | the nice part about that, as well, is that it completely bypasses the dialplan |
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| 19:21.33 | mjordan | if that doesn't do it for you, well... Asterisk is open source. Patches welcome :-) |
| 19:22.08 | mjordan | but really, when you get into offline storage mechanisms followed by some sort of online detection mechanism, you're tackling a problem that is not as trivial as you might think. |
| 19:22.48 | mjordan | personally, I'd rather just own that problem myself outside of Asterisk, and use Asterisk to do the hard bit of turning JSON blobs into SIP MESSAGE requests (or XMPP messages, etc.) |
| 19:27.55 | v0ip-wolf | is there any info on picture messaging? I was able to do sms using this tutorial http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html |
| 19:28.47 | mjordan | does your upstream provider support MMS? |
| 19:28.54 | mjordan | or have a SIP to MMS gateway? |
| 19:32.15 | v0ip-wolf | i just wanted to sms/mms to other sip on the same network |
| 19:32.17 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 19:32.31 | v0ip-wolf | not over pstn or cellular |
| 19:37.37 | Demon_VoIP | mjordan, My opinion: rudimentary signaling mechanisms in SIP is a barrier to mass users. Google understood this and created a Jingle. It is a pity that they changed their minds. They buried Hangouts. |
| 19:37.53 | v0ip-wolf | would it be possible for mms to sip? |
| 19:40.04 | robmal | o_O But why would you do any of these things? I understand cellphone similarity, but even they use two channels, one for voice (in this case SIP) and one for thata (in this case XMPP/e-mail/pidgeons) |
| 19:40.47 | mjordan | v0ip-wolf: generally, for that, you want to use MSRP, which Asterisk doesn't have a stack for |
| 19:40.47 | robmal | s/thata/data/ |
| 19:41.25 | v0ip-wolf | thanks for the info, I will search for that |
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| 19:49.27 | mjordan | robmal: SIP is just a signaling protocol. It can convey anything. You can use it to establish text only sessions. |
| 19:50.03 | robmal | Of course. |
| 19:50.19 | mjordan | a good example of someone who cares about these kinds of things is the Blink SIP client |
| 19:50.26 | mjordan | they've done a lot of interesting work in this area |
| 19:53.42 | robmal | Looks great, i'll give it a go. |
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| 20:16.39 | acidfoo | what tool exist to loop through every files of a directory in the dial plan ? thank you ! |
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| 20:22.02 | [TK]D-Fender | acidfoo: There is none. |
| 20:23.01 | acidfoo | ok ! thank you, I'll write my thing via the agi |
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| 20:29.32 | [TK]D-Fender | packs up to head home |
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| 23:06.08 | priorfra | Hi all; just a question: Asterisk 13 need webrtc2sip installed on server to do video calls ? (or this was only with Asterisk 11 ?) |
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| 23:38.09 | bobbymc | hi room |
| 23:38.36 | bobbymc | priorfra: you wont need webrtc2sip, ast13 already has webrtc support |
| 23:38.51 | bobbymc | i believe even 11 does video also |
| 23:39.13 | bobbymc | does anybody in this room know much about asterisk timing sources? |
| 23:39.30 | bobbymc | i wonder if in asterisk 11 does meetme use the timefd timer |
| 23:43.02 | rmudgett | bobbymc: meetme uses DAHDI for audio mixing. Since you have to have DAHDI installed anyway you might as well use DAHDI timing. |
| 23:45.26 | bobbymc | well i dont want to use dahdi because it requires access to the kernel for timing and the asterisk box is on a xen vm |
| 23:46.08 | bobbymc | so in order to use vms i was hoping to take advantage of confbridge maybe which uses timefd, but im not 100% sure if timefd also depends on the kernel for timing |
| 23:46.58 | file | it does. |
| 23:49.28 | bobbymc | file: so basically the timing limitation will be the same between confbridge and meetme ? |
| 23:49.45 | file | I'm not sure what you mean by limitation |
| 23:50.28 | file | something has to provided timing on a system, no matter what in a virtualized environment that can be a little... shaky |
| 23:50.46 | file | but it's better these days |
| 23:50.56 | bobbymc | well on xenserver i cant use astersk if i depend on iax trunks or meetme because the timing is off since the vm doesnt have direct access to the hardware timing surce |
| 23:52.16 | file | people are running Asterisk in VMs these days in production and it seems to be fine for them |
| 23:52.27 | bobbymc | if works ok if there is only 1 vm on the xen host, but if i have multiple asterisk servers on the same xen host the timing is off and call quality degrecates |
| 23:52.52 | bobbymc | yea but i bet most of them dont depend on meetme and iax trunks as much |
| 23:53.27 | bobbymc | if both arent required it works great, but with asterisk 1.4 and lower i always had issues putting them in vms because the timing isnt on par |
| 23:53.48 | file | I'm fairly certain people are using ConfBridge at the very least in virtualized environments |
| 23:53.55 | file | heck, I've done it |
| 23:54.09 | bobbymc | i recently switched to asterisk 11 and unless i put it in production in a vm i have no way of testing if it really works well or not and i dont want the customers to be a test |
| 23:54.24 | file | I did it for the AstriDevCon conference bridge |
| 23:54.24 | bobbymc | but with high call volumes? |
| 23:54.36 | file | define high call volumes |
| 23:54.53 | bobbymc | lets say 20 cps and a total of 200 concurrent calls |
| 23:55.00 | bobbymc | all calls are via a iax trunk |
| 23:55.06 | file | I don't use IAX, so no |
| 23:55.26 | bobbymc | yea only iax trunks and conf rooms require proper timing, everyhting else deosnt really |
| 23:55.52 | file | playback does, but the jitterbuffer on the endpoints can compensate |
| 23:56.05 | file | as do things like server side tone generation |
| 23:56.37 | bobbymc | im not a expert but here is what we do |
| 23:56.47 | bobbymc | we generate a big amount of calls on box a, |
| 23:56.50 | bobbymc | its a dialer |
| 23:56.58 | bobbymc | so we build and break down tons of calls |
| 23:57.14 | bobbymc | the calls that answer we use iax trunk to send to a server b where agents are registered on |
| 23:57.29 | bobbymc | the agents on server b sit in a meetme room currently |
| 23:57.53 | bobbymc | in the past when i put both servers on a vm on asterisk 1.4 the quality was really bad when we have high cps |
| 23:58.22 | bobbymc | i found out after that iax and meetme require timing which a vm isnt abl to do very well because it deosnt have access to the hardware bus |
| 23:58.58 | bobbymc | i might be wrong about some of this, excuse me if i am but thats what i found out so far, so now at this point we are rewriting the app and supporting asterisk 11 |
| 23:59.27 | file | I can only speak for Asterisk, in which case it does require a stable timing source for those - yes. |
| 23:59.29 | bobbymc | and my goal is to have another attempt at virtulaizing these boxes, i hoped confbridge doesnt require the same type of timing like dahdi does |
| 23:59.37 | file | it requires timing |
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