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| 06:05.43 | trz77 | Anyone know how to convert this to the wizard format? https://pastebin.com/sN3vbVV6 |
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| 06:52.15 | [TK]D-Fender | Show us your attempt |
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| 07:41.47 | perseiver | hello |
| 07:43.14 | perseiver | Have anyone tried to Connect Asterisk with Streaming Server. So that when SDP Session Progress, Peer User connected with Streaming Server directly and We can play the music |
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| 09:39.03 | war9407 | sibiria: thanks |
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| 10:29.00 | martinvw | Can I get Asterisk to dump the configuration parameters of a certain queue? Background: I've set announce-position=more and announce-position-limit=10 in my queue template that all queues derive from, but our main queue plays position announcements for <10 callers, and I don't understand why :/ |
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| 12:43.22 | niekniek | hello all. I need a way to get my sip registrations and other voip traffic from subnet x (phones) to subnet y (pbx). There cannot be routed between the subnets, but I can put in a host which has a connection to both networks. What would be the best way to accomplish this? I've been thinking about TURN? |
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| 12:44.12 | wakko | hi |
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| 15:25.35 | cresl1n | waves |
| 15:28.36 | Ai9zO5AP | big thanks to asterisk developers |
| 15:33.15 | igcewieling | some days I wish I could go back to bed and try again. |
| 15:34.54 | [TK]D-Fender | I wish I hadn't left mine #snowpocalypse |
| 15:37.09 | file | igcewieling: yes. |
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| 17:04.52 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.24.1 (2018/12/26) 16.1.1 (2018/12/26), Security Only: 15.7.1 (2018/12/26); 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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| 17:34.42 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.24.1 (2018/12/26) 16.1.1 (2018/12/26), Security Only: 15.7.1 (2018/12/26); 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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| 18:16.30 | Phil-Work | can you add custom headers to SIP responses in Asterisk 11? |
| 18:17.34 | Samot | Yes. |
| 18:17.44 | Samot | ER no Sorry, read that wrong. |
| 18:18.00 | Samot | You cannot manipulate or add anything to replies. |
| 18:18.31 | Samot | Regardless of Asterisk version. |
| 18:19.06 | Phil-Work | I recall Matt (F) having largely implemented it at Kamailio World a couple of years back - I guess it never made it into a version :( |
| 18:19.11 | Phil-Work | thanks for the confirmation, Samot |
| 18:19.24 | Samot | Yeah, that is Asterisk + Kamailio |
| 18:19.33 | Samot | Because Kamailio is a SIP router, Asterisk is a toolkit. |
| 18:19.51 | igcewieling | Kamailio is a toolkit too |
| 18:19.53 | Phil-Work | pretty sure he did it in Asterisk itself but had some issues |
| 18:20.02 | Samot | Then he wrote something |
| 18:20.10 | Phil-Work | indeed |
| 18:20.22 | igcewieling | Phil-Work: no, he didn't. Maybe you mean add headers to outgoing invites -- that is trivial. |
| 18:20.23 | Samot | igcewieling: How so? |
| 18:20.38 | Phil-Work | igcewieling, no - responses |
| 18:21.24 | Samot | igcewieling: How is Kamailio a toolkit? |
| 18:21.36 | igcewieling | Samot: As much work goes into setting up Asterisk as you do Kamailio |
| 18:21.49 | Samot | That doesn't make it a toolkit. |
| 18:21.58 | igcewieling | What makes Asterisk a toolkit then? |
| 18:22.15 | Samot | It supports numerous connections types. |
| 18:22.29 | Samot | It can run as a media application server without having any real "phones" connected to it. |
| 18:23.01 | Samot | It has voicemail, queues, conference bridges, etc. |
| 18:23.18 | Samot | Kamailio has none of those. It is a router and a registrar. |
| 18:23.47 | igcewieling | voicemail queues etc are required for it to be a toolkit? |
| 18:24.00 | Samot | As a Telephony Toolkit, sure. |
| 18:24.11 | igcewieling | OK. |
| 18:24.12 | Samot | As an application to run voice services, yes. |
| 18:24.17 | igcewieling | I disagree, but that is nothing new. |
| 18:24.28 | Samot | Kamailio lets you register devices, route locations and make calls. |
| 18:24.43 | Samot | It doesn't touch the media |
| 18:25.05 | Samot | You'd be making endpoint to endpoint calls with nothing in the middle. |
| 18:25.41 | Samot | In the grand scheme of things, Asterisk is not a SIP Router. |
| 18:26.36 | Phil-Work | I knew I wasn't going mad: https://www.youtube.com/watch?v=d0zHiLkHDyY |
| 18:26.38 | Phil-Work | 32m 30s |
| 18:27.58 | Samot | And there you go, he modified the code. |
| 18:28.02 | Samot | To the sip stack. |
| 18:28.17 | Phil-Work | indeed |
| 18:28.22 | Samot | And he broke the ability to read incoming sip headers. |
| 18:28.26 | Samot | So it's not worth a thing. |
| 18:28.28 | Phil-Work | indeed |
| 18:29.12 | Phil-Work | I was holding out some vague hope that he'd have finished it and put it into a released version |
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| 19:59.37 | veesh | what tools are used to stream online radio for MOH? |
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| 20:07.25 | ais-admin | mpg123 can be used. i.e. application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 2048 -y -@ http://199.180.75.116:80/stream |
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| 20:41.48 | veesh | is there a perl ARI library? |
| 20:43.36 | sibiria | no but there are perl websocket modules |
| 20:44.01 | veesh | i guess i'll write as I go |
| 20:44.09 | veesh | i use mojolicious already (it's the best) |
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| 21:32.37 | tm1000 | Sangoma Astricon Survey: https://www.surveymonkey.com/r/KZD6GTH |
| 21:36.01 | seanbright | what is a 'core developer?' |
| 21:38.15 | seanbright | i guess if you have to ask, you aren't one |
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