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| 00:19.25 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12) 16.0.0-rc1 (2018/08/08), Standard: 15.5.0 (2018/07/12); 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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| 11:35.59 | jkroon | short of restarting asterisk (13), is there a way to reload the xmldocs? |
| 11:36.31 | file | no. |
| 11:37.46 | jkroon | right ... so core restart when convenient it is ... and then we hope pjsip recovers it's realtime endpoints this time round. |
| 11:39.06 | stefan27 | hello asterisk people |
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| 13:49.59 | n3t | Is there a way to dynamically set `fromuser` while dialing using SIP? I'm currently browsing documentation but I can't find anything related. I guess I don't use proper keywords to search. |
| 13:54.28 | n3t | If I understand correctly, this feature has been implemented in Asterisk 14, https://issues.asterisk.org/jira/browse/ASTERISK-25803. |
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| 13:57.56 | n3t | I removed `fromuser` from `sip.conf` and set caller ID appropriately in `extensions.conf`. Seem to work. |
| 13:59.05 | n3t | Even better, it's possible to use `o` Dial's option :) |
| 13:59.23 | n3t | (Which is in my case expected behaviour.) |
| 14:04.00 | stefan27 | yeah, whether fromuser is set in sip.conf and what CALLERID(num) was set to will matter... There's also the channel variable called SIPFROMDOMAIN which at least asterisk-13.15's chan_sip.c would parse... That may help if you also want to alter the domain of the from-uri of the outgoing SIP INVITE... But it's all a puzzle :) |
| 14:04.21 | stefan27 | glad you could find a solution |
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| 15:06.44 | Samot | v14 and above allow you to specify the TO and FROM URIs. |
| 15:07.10 | Samot | In the Dail() string |
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| 16:11.58 | simbalion | Hi, can anyone point me to instructions for recording all outgoing calls? |
| 16:12.44 | simbalion | I've got a 1-line 2 extension setup with a simple SIP trunk so I'm guessing it's pretty easy |
| 16:16.12 | [TK]D-Fender | "core show application monitor", "core show application mixmonitor" |
| 16:20.28 | simbalion | I've got mixmonitor setup to record queues in queues.conf, do I just add monitor-type=MixMonitor to extensions.conf under the outbound section? |
| 16:21.09 | [TK]D-Fender | Add it wherever you want it |
| 16:21.52 | [TK]D-Fender | "Sections" only exist or have meaning if you created them that way |
| 16:22.48 | simbalion | Okay, I assumed different conf files have different syntaxes since they're used by different modules |
| 16:22.56 | simbalion | er, applications |
| 16:24.20 | [TK]D-Fender | Queue's are about the only thing that have a built-in reference to recording |
| 16:24.29 | [TK]D-Fender | Everything else = dialplan and is your job |
| 16:25.00 | [TK]D-Fender | Maybe conferences too actually ... not 100% sure on that, but I can imagine it... |
| 16:27.39 | simbalion | hrm looks like it cannot go there |
| 16:27.47 | simbalion | on reload it said unknown directive, ignoring |
| 16:29.36 | [TK]D-Fender | then you did it wrong |
| 16:30.49 | file | there is no built-in "option" in dialplan to turn on call recording, it's configured to execute/do things, one of which may be to invoke an application that does call recording |
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| 16:57.28 | psim | hi all, question about conferences w/ video and sfu |
| 16:58.07 | psim | say we set up a webrtc conference using sfu and then join a standard sip client |
| 16:58.38 | psim | sip client should get all audio. what about video. can the sip client get at least switched video from the conference (based on talker) ?? |
| 16:58.44 | file | not currently. |
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| 16:59.11 | file | the SFU doesn't know what is a WebRTC client and what is one, and it doesn't allow a mix of strategies like that |
| 16:59.15 | file | er what isn't one |
| 16:59.59 | psim | @file thanks, I see. does the sip client then get all video streams (and maybe pick the first one in the SDP to display) or does it get no video offers? |
| 17:00.14 | file | it doesn't get any video forwarded to it currently |
| 17:01.05 | file | you could configure PJSIP to offer it additional streams, but no idea what would happen with it... |
| 17:01.44 | rmudgett | It would be up to the endpoint how it handled any offered video streams. |
| 17:01.53 | psim | if the sip client sends a video stream to the conference would that get relayed to the webrtc users?? |
| 17:02.04 | file | it would |
| 17:02.17 | psim | hmm, interesting. so we could get one way video from the sip client at least. |
| 17:02.25 | rmudgett | The conference doesn't treat any participant differently than any other. |
| 17:02.33 | simbalion | [TK]D-Fender: You told me that I could do it that way, so I guess "wrong" is in the eye of the beholder. |
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| 17:07.36 | psim_ | i apologize, lost my connection. last was: [13:02] <@rmudgett> The conference doesn't treat any participant differently than any other. - was there any more follow up on topic ? |
| 17:07.59 | rmudgett | no |
| 17:08.04 | psim_ | ty |
| 17:08.55 | psim_ | would like to be able to get video switched by talker. is this in the dev roadmap by any chance? no need for full mcu for sip clients, just some minor video support |
| 17:09.18 | file | I know of noone working on such a thing |
| 17:09.21 | file | or planning to |
| 17:09.48 | file | you're the first person that I've personally seen wanting it |
| 17:13.02 | avb | file: i think that confbridge works just fine with video |
| 17:13.18 | file | I'm not sure what your statement is about |
| 17:13.31 | file | I was referring to the mixed behavior and configuration that psim_ was talking about with the SFU |
| 17:13.40 | avb | ah |
| 17:13.43 | avb | sorry |
| 17:13.57 | avb | i think i havent read all the conversation |
| 17:14.45 | avb | i have finally tried video on asterisk-15 as you suggested. Great job! everything is works better |
| 17:15.06 | file | video killed the radio star |
| 17:15.06 | avb | my guy is fixing a softphone to add variable bitrate to the video encoding |
| 17:15.34 | avb | am I writting that bad? :) |
| 17:15.49 | psim_ | alternatively i would be happy to find an mcu product/software that will send and receive streams to asterisk's sfu. an mcu-sfu gateway if you will |
| 17:15.52 | file | it's a song, I think I've mentioned it in relation to the video work publicly? maybe |
| 17:16.10 | file | I use code words internally for things and they don't always go public |
| 17:16.29 | avb | haha :) |
| 17:16.40 | [TK]D-Fender | <simbalion> [TK]D-Fender: You told me that I could do it that way, so I guess "wrong" is in the eye of the beholder. <- monitor & mixmonitor work. If you want so help in getting it working show us what you did following my fconfirmations that it's up to your dialplan.... |
| 17:16.50 | file | https://www.youtube.com/watch?v=Iwuy4hHO3YQ |
| 17:17.04 | avb | file, i just were asking you a direct question one day here. and seems thats where you failed with your code words |
| 17:17.07 | avb | lol |
| 17:21.32 | igcewieling | idly wonders of a Mexican restaurant specializing in pork should be called Pig Latin |
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| 18:31.06 | Slade | the theme would have to end there |
| 18:32.02 | Slade | no one wants to order their ollopa urrittoba with a side of uacamolega |
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| 19:12.48 | simbalion | [TK]D-Fender: Well I looked at the help info but what I could really use is some examples or something, because I'm still kind of a newb when it comes to the syntax of the extensions application. I assume the best way is to start recording as an outgoing call is made, I didn't do this for incoming calls because I didn't want to record hold music |
| 19:13.14 | [TK]D-Fender | you don't have to |
| 19:13.17 | [TK]D-Fender | that's an option |
| 19:13.21 | [TK]D-Fender | read the instructions |
| 19:13.31 | simbalion | I tried searching online but every example has like half a dozen lines or more, that seems like a lot, but maybe I don't know |
| 19:13.53 | simbalion | I'm using version 13 btw, not the latest, I just installed using apt-get |
| 19:16.22 | [TK]D-Fender | I don't understand "searching online".... |
| 19:16.30 | [TK]D-Fender | this is a dialplan app. the instructions are right there in CLI |
| 19:16.36 | [TK]D-Fender | This isn't something you have to go out to get |
| 19:16.49 | [TK]D-Fender | It can be down to ONE line. |
| 19:17.46 | simbalion | Great! If you web search for it you find stackoverflow and similar pages where people are recommending a half dozen lines. |
| 19:22.57 | [TK]D-Fender | wiki.aserisk.org |
| 19:23.06 | [TK]D-Fender | Also no need to look for external docs. |
| 19:23.20 | [TK]D-Fender | it's always sitting right there in the official web documentation |
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| 20:00.20 | simbalion | [TK]D-Fender: Thanks for your guidance, I was able to get it working by adding the following to my outgoing section same => n,MixMonitor(${EXTEN}-${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}.wav,v(1)) |
| 20:00.29 | simbalion | had to boost the volume a bit for the outside line |
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| 20:02.13 | [TK]D-Fender | Glad you've got it figured out |
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| 20:16.12 | simbalion | Me too! I should have done it weeks ago.. had a bad time with a rude receptionist today and realized I didn't have the proof to share with her organization... that's what got me started heh. |
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