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| 01:27.23 | *** 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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| 03:51.26 | *** 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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| 09:50.53 | fubber | Hi everybody. I'm new in the field of video conferencing - and I'm trying to decide if I should look into Asterisk PBX for this. |
| 09:51.53 | fubber | Is Asterisk primarily for VoIP? Or does it also handle SIP calls for video conferencing equipment? |
| 09:55.08 | file | it's primarily been audio in the past - but has support for an SFU video mechanism for WebRTC, and video switching |
| 09:55.10 | file | it doesn't do video mixing |
| 09:55.23 | file | and it can pass video through |
| 09:57.09 | fubber | So, Asterisk can mediate video calls between two different video conference locations - as long as they don't need video mixing? Basically, it could handle the peer discovery part? |
| 09:58.29 | file | if a call is placed through it with video streams, it will relay the video |
| 09:59.04 | fubber | @file: Thank you. Does it assist setting up peer-to-peer communications, or will it always relay? |
| 10:00.10 | file | noone has worked on making it go directly between endpoints, video is not something many have touched |
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| 12:42.39 | nehemiah | I couldn't find any information on it. So I assume it's not supported, but just to be sure, can I add a password hash in the pjsip.conf file as a password? |
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| 12:46.31 | [TK]D-Fender | yes. |
| 12:46.50 | [TK]D-Fender | If you couldn't find any information on it then you didn't look at the sample config * provides |
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| 15:21.08 | Ryushin | Looking at building a new gui drivin asterisk server to replace our old built from source command line only asterisk server. Anyone have any thoughts or insights about Nerd Vittles FreePBX or Wazo? |
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| 15:25.01 | igcewieling | Ryushin: use the FreePBX distro. |
| 15:25.31 | Ryushin | igcewieling: That is what I was leaning towards. |
| 15:26.26 | Ryushin | One thread said the Wazo was newer code and from what I was gathering it was going to be the future. But the FreePBX has been updated just this month. Still researching. |
| 15:28.01 | Ryushin | Not sure if the Incredible FreePBX is entirely free or if the Whole Enchilada has some licensing costs after a certain number of users. |
| 15:28.10 | Ryushin | This is for a small office. 25 people or so. |
| 15:28.33 | Samot | There is no Incredible FreePBX |
| 15:28.45 | Samot | There is IncrediblePBX which is forked from FreePBX.. |
| 15:28.50 | Samot | Completely different. |
| 15:30.35 | igcewieling | Remember, you won't get help here on forked distros |
| 15:31.14 | igcewieling | ^^^ I meant #FreePBX |
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| 15:32.16 | Ryushin | igcewieling: Understandable on not getting help. Samot: Yea, I meant the fork of FreePBX. But it looks like Nerd Vittles has a fork of FreePBX and Wazo. Just trying to narrow down which one will be the best choice. |
| 15:33.13 | Samot | OK, there is zero commercial support/modules/apps in IncrediblePBX (the FreePBX fork) |
| 15:33.19 | Samot | No clue about Wazo. |
| 15:33.36 | [TK]D-Fender | Neither of those. Limited support & community of forked versions with less development. |
| 15:33.41 | Samot | If you're trying to choose between FreePBX and Wazo then just use one or the other... |
| 15:33.44 | Samot | Don't go with a fork. |
| 15:34.35 | [TK]D-Fender | Basically an official FreePBX build of some kind is just about the only real game in town for a free solution, which unless you have some sort of special needs should serve jsut fine. |
| 15:35.53 | Ryushin | [TK]D-Fender: Man, you are still here. Must have been more then a decade since I first saw you here. LOL |
| 15:36.20 | [TK]D-Fender | About to hit 15.... |
| 15:36.41 | [TK]D-Fender | That wins me a plushie off the middle shelf I believe... |
| 15:39.14 | Ryushin | Well, almost a decade. My asterisk irc log starts in 2009. |
| 15:40.33 | [TK]D-Fender | I started with * in Feb 2004. |
| 15:41.56 | Ryushin | I bet we both are feeling old right now. LOL |
| 15:44.27 | Ryushin | Seems that IncrediblePBX is based on Asterisk 13 while IncredibleWazo is based on Asterisk 15. Time to do some more digging. |
| 15:45.01 | tripleslash | Just go FPBX :) |
| 15:47.05 | Ryushin | Well, that seems to be the preference then. FreePBX then. |
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| 17:56.40 | anonymouz666 | file: sorry to ask you directly, but there's some option in PJSIP to make it add the fmtp for h264? |
| 17:56.57 | anonymouz666 | I saw in the code, but the outbound invite is being sent without the ftmp for the h264 |
| 17:57.45 | anonymouz666 | or it depends on what comes on SDP from the client? |
| 17:58.31 | file | if the module is loaded then it uses what came in |
| 18:00.02 | anonymouz666 | I got microsip sending a call through a PJSIP endpoint within fmtp on SDP |
| 18:00.30 | anonymouz666 | but there's a webrtc client using the same endpoint and the outbound invite goes without the ftmp for h264 |
| 18:03.48 | anonymouz666 | the res_format_attr_h264 is loaded |
| 18:11.58 | anonymouz666 | file, maybe i found the issue, i think asterisk expects the rtpmap of h264 coming in 96 rtpmap |
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| 18:12.11 | anonymouz666 | the damn client is sending the h264 on rtpmap 100. |
| 18:13.35 | anonymouz666 | makes sense? |
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