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| 00:21.12 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 07:56.05 | dan_j | Hi. Where can I find the mysql schema for meetme? |
| 07:58.16 | dan_j | found it |
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| 09:27.56 | wyoung | dan_j: Isn't meetme deprecated? |
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| 09:31.55 | samwierema | I'm trying to set up Asterisk for WebRTC through PJSIP. It all seems to work (I get the signals, Asterisk executes the dialplan), but there's no audio played back. Looking at wireshark, it does seem like I'm getting RTP packets, but no audio. PJSIP is configured like this: https://gist.github.com/samwierema/b6458d67be166a4a788bbced625e8532. Is there anything I'm missing (in terms of NAT setups, or anything else)? |
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| 09:47.29 | dan_j | wyoung: i wasnt aware. Whats replaced it for conference calling? |
| 10:06.11 | dan_j | I can't seem to find anywhere that says meetme is deprecated but can see there is confbridge. |
| 10:06.24 | dan_j | Please can someone confirm which one I should be using if I'm going to use one? |
| 10:21.27 | wyoung | dan_j: Ask [TD]Fender |
| 10:25.55 | file | ConfBridge is the current recommended option |
| 10:39.42 | wyoung | that's the one, thnx @file |
| 10:40.21 | wyoung | @file: Are you a perl scripter by chance?? |
| 10:40.27 | file | um, no |
| 10:40.42 | wyoung | ok, @file is used a lot in perl :) |
| 10:41.00 | wyoung | (for IO) |
| 10:50.15 | dan_j | is realtime available for confbridge or should it all be set up in the dialplan? |
| 10:50.55 | wyoung | dan_j: ask [TD]Fender |
| 10:51.10 | dan_j | I would but he's not on right now. |
| 10:51.47 | file | there is no explicit realtime support in it |
| 10:52.04 | wyoung | dan_j: My second choice would be @file |
| 10:52.10 | dan_j | :) |
| 10:52.27 | wyoung | :P |
| 10:52.34 | wyoung | <3 @file, you are great |
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| 12:37.54 | jkroon | dan_j, most of the parameters for confbridge can be controlled from dialplan, so you could use func_odbc in combination with a handful of strategic profiles to do quite intense things. |
| 12:39.09 | dan_j | Just thinking about the dmtf options while in the conference. |
| 12:42.46 | jkroon | so what I did is I created a few profiles with the different options that we want, and then select them dynamically using dialplan. |
| 12:43.43 | jkroon | having too many layouts etc is outright insane. granted you may only want to be able to edit the various options, in which case realtime loading the config file may be adequate, but i'll be the first to admit that whilst I've seen that certain config files can be directed to database, I haven't actually dont that just yet. |
| 12:44.07 | jkroon | I'm mostly latching onto sip and iax2 realtime, and then doing things "dynamically" using func_odbc. |
| 12:44.32 | jkroon | realtime can also latch onto res_odbc so it should be possible to utilize the same db connection that func_odbc uses. |
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| 14:12.12 | Samot | dan_j: What DTMF options in the conf bridge? |
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| 18:19.22 | salviadud | Has anyone here installed wanpipe on gentoo? |
| 18:19.30 | salviadud | I'm wondering if I should install yate instead |
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| 19:59.45 | dan_j | Hi. I've got an ongoing problem since upgrading to v13. I've set a queue member to ringinuse=yes, but the debug still says "app_queue.c:5299 is_our_turn: There are 0 available members" when the agent is already on the phone. |
| 19:59.48 | dan_j | Is this a bug? |
| 20:02.24 | dan_j | I also moved to pjsip when I moved to 13 |
| 20:02.53 | dan_j | The agent is still able to receive calls from other queues. Just not the same one that he's already talking to. |
| 20:03.03 | dan_j | agent = sip endpoint |
| 20:04.09 | Samot | I know we talked about this last week. |
| 20:04.49 | Samot | @file: Can you clarify this from the queues.conf.sample : "(Note: only the SIP channel driver currently is able to report 'in use'.)" <-- Does not mean Chan_PJSIP does not report 'in use' properly? |
| 20:04.56 | Samot | And only Chan_SIP does? |
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| 20:06.31 | file | it does report it, that comment is old |
| 20:06.45 | Samot | Thanks |
| 20:07.41 | [TK]D-Fender | devices state is device state |
| 20:07.46 | [TK]D-Fender | * checks whatever it is given |
| 20:08.30 | dan_j | device state should be ignored if I've set it to 'ringinuse' |
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| 20:09.10 | dan_j | [TK]D-Fender: 'ringinuse' partially works because the agent can still receive calls from other queues. |
| 20:09.29 | dan_j | file: does this sound like a bug to you? |
| 20:10.35 | [TK]D-Fender | sounds like a queue won't call members on calls from THAT queue |
| 20:10.43 | file | probably? I don't know the option or behavior |
| 20:11.02 | dan_j | [TK]D-Fender: well put |
| 20:11.12 | [TK]D-Fender | Yes, this is the "support" queue, but if the "EMERGENCY" queue sends a call then it'll allow it |
| 20:11.14 | dan_j | couldnt figure out how to simplify it into a sentence! |
| 20:11.16 | [TK]D-Fender | but not another "support" call" |
| 20:11.53 | dan_j | in v11, 'ringinuse' functionally would allow multiple 'support' queue calls to the same agent |
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| 20:15.00 | dan_j | file: You said you dont know what options. Are there any options that would affect 'ringinuse'? |
| 20:15.15 | file | I really don't know, app_queue is not something I deal with |
| 20:15.18 | file | asking me is useless |
| 20:15.45 | dan_j | ok |
| 20:15.48 | dan_j | noted |
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| 20:19.14 | Samot | Are you running the queue in RealTime? |
| 20:19.19 | dan_j | yes |
| 20:19.26 | Samot | Do you really need to? |
| 20:19.47 | Samot | And have you tested if this is an issue when not using RealTime? |
| 20:19.57 | dan_j | I suppose I can generate the queues.conf automatically, but the queuemembers needs to be in realtime |
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| 20:20.27 | Samot | It's a queue. |
| 20:20.37 | Samot | How often are the actual queue settings going to change? |
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| 20:21.13 | dan_j | No idea. Clients update their settings online and I have code that writes the various conf files or updates the asterisk db |
| 20:21.40 | dan_j | pjsip uses conf files now along with extensions.conf which i also took out of realtime |
| 20:21.42 | Samot | Well I would test if this issue happens when using static. |
| 20:21.49 | dan_j | will give it a go |
| 20:25.57 | dan_j | It seems (using realtime), if i set the wrapup time to 0, it resolves the issue, which is weird |
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| 20:28.15 | dan_j | I suppose wrapuptime should be ignored if ringinuse is on, since an endpoint that is busy wrapping up is considered in-use, and ringinuse is set. |
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| 21:25.16 | salviadud | I'm having issues compiling wanpipe |
| 21:25.19 | salviadud | shrugs |
| 21:25.42 | salviadud | Anybody done it in gentoo? |
| 21:28.43 | salviadud | Well, I'm just gonna downgrade my friggin' kernel then |
| 21:29.35 | Samot | salviadud: You could try explaining the _actual_ issue you have compiling wanpipe. |
| 21:29.51 | salviadud | let me pastebin it... |
| 21:30.41 | salviadud | https://bpaste.net/show/22d4f1fb6af5 |
| 21:32.11 | Samot | Well I won't be able to help. |
| 21:32.23 | salviadud | :) |
| 21:32.24 | Samot | I don't use wanpipe or anything that would require it in Asterisk. |
| 21:32.33 | salviadud | It's for a sangoma card... |
| 21:32.37 | Samot | But just saying you're having issues... |
| 21:32.37 | salviadud | what a pain. |
| 21:32.54 | Samot | And not actually saying what they are, will not get people jumping out to help. |
| 21:33.13 | salviadud | Well, I think my issue is that wanpipe wants a 3.x.x kernel |
| 21:33.18 | salviadud | I'm running a 4.x |
| 21:33.29 | Samot | Sigh. |
| 21:33.44 | salviadud | It's my first time |
| 21:33.59 | salviadud | I've used asterisk on SIP and IAX without any analog stuff |
| 21:34.18 | salviadud | never had to compile anything besides asterisk itself. |
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| 23:02.33 | polysics | hey! will SayDigits with SAY_DTMF_INTERRUPT only jump to an extension, or is there a way to have it set a value somehow? |
| 23:03.56 | *** join/#asterisk cmendes0101 (~cmendes01@124-169-140-249.dyn.iinet.net.au) |
| 23:06.52 | *** part/#asterisk kharwell (kharwell@nat/digium/x-zxpkhvokbasavfpu) |
| 23:21.44 | *** join/#asterisk matrix1233 (~matrix123@41.230.49.131) |
| 23:22.13 | *** join/#asterisk robmal (r@wporzo.pl) |
| 23:58.15 | *** join/#asterisk polysics (~polysics@47.151.146.151) |
| 23:58.19 | polysics | hello again! :) |
| 23:58.31 | polysics | do I need to enable anything to get URI support in Playback? |
| 23:58.50 | Samot | Asterisk 14 |
| 23:58.55 | polysics | ok, I do have that |
| 23:59.07 | Samot | Then you're good to go. |
| 23:59.10 | Samot | Works great. |
| 23:59.29 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Project+-+URI+Media+Playback |
| 23:59.38 | [TK]D-Fender | https://www.google.ca/#q=asterisk+playback+uri |
| 23:59.51 | Samot | That's the wiki page I used and it worked fine. |