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| 01:10.29 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.0.0, 13.37.0, 16.14.0 (2020/10/20) Standard: 17.8.0 (2020/10/20); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 04:28.07 | Janos | question, is there a AMI event that updates the statistical data of a queue, like the number of processed calls ? or do I have to use the QueueStatus action periodically to get this data ? |
| 04:28.42 | Janos | For example there is a MemberStatus that is generated every time something changes on a member |
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| 13:27.21 | AL13N_lappy | I've seen that sip clients are able to redo codecs (is that the re-invite i see?) is there a way for an extension to trigger a reinvitation for codec stuff to the client? |
| 13:28.30 | AL13N_lappy | it may be a way to have playback messages in only one codec and no need for transcoding? |
| 13:37.43 | Samot | Redo codecs? |
| 13:38.01 | Samot | AL13N_lappy: What do you mean by "redo codecs"? |
| 13:39.38 | seanbright | he means renegotiate |
| 13:39.50 | seanbright | (and i think you knew that...) |
| 13:40.22 | seanbright | Samot: you're a kamailio genius, yeah? |
| 13:40.36 | Samot | I know it decently. |
| 13:40.51 | seanbright | i am going to send you a PM and devalue your time, one sec |
| 13:48.12 | Samot | Exit the processing. |
| 13:50.00 | Samot | What happens on the client side? |
| 13:50.50 | file | How unidirectional of a conversation |
| 13:51.28 | Samot | Wrong window |
| 13:51.40 | Samot | seanbright: ^^^ |
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| 13:57.21 | AL13N_lappy | so... is there a way to trigger that from an extension, to send like a re-invite with different set of codecs? |
| 13:57.41 | AL13N_lappy | or do i need to manually re-transcode all used messages on the server? |
| 13:58.58 | AL13N_lappy | i've looked into linphone and it looks like they have some kind of contact list; using RFC 5367; i was looking into something similar in asterisk, but i could only see subscribe= things? but that seems something else... |
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| 16:10.07 | igcewieling | IIRC, Some Digium and/or Sangoma cards used Xilinx chips. Is that correct, and if so, is it still the case? |
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| 16:16.55 | AL13N_lappy | file: any idea which callerid field i need to set, so that the sip contact line's user part is not "asterisk" ? |
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| 16:17.37 | seanbright | *CLI> config show help res_pjsip endpoint |
| 16:20.19 | igcewieling | AL13N_lappy: 1) the contact line is not related to authorization or authentication and if you really want to do that, 2) use use_callerid_contact=yes in [general] |
| 16:30.22 | AL13N_lappy | igcewieling: thanks |
| 16:31.07 | AL13N_lappy | apparently, the from_user= field on endpoint has effect on that, so i've used that, and that seemed to work |
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| 17:07.59 | laerus | hello, i'm using asterisk 16, trying to access channelvars from ARI doesn't work, i'm setting a variable in the dialplan but it doesn't appear in the channel object, any idea what is going on? |
| 17:09.19 | file | how are you trying to access channel variables? there is a specific route for retrieving from the channel |
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| 17:09.52 | laerus | i'm getting the channel object on StasisStart event, should i get it from the channel endpoint? |
| 17:10.43 | file | the channel object does not include dialplan variables by default, you have to use the route to retrieve them |
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| 17:19.36 | laerus | file, doesn't work too, channelvars key is missing from json |
| 17:21.03 | file | you're going to need to be specific about what exactly you are doing |
| 17:22.20 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Channels+REST+API#Asterisk18ChannelsRESTAPI-getChannelVar is what I was referring to |
| 17:22.50 | laerus | okay i was using the https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Channels+REST+API#Asterisk16ChannelsRESTAPI-get |
| 17:22.53 | laerus | thx |
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| 18:29.30 | igcewieling | IIRC, Some Digium and/or Sangoma cards used Xilinx chips. Is that correct, and if so, is it still the case? |
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| 20:38.20 | Derkko | Hi, im trying to send a call to a peer that's inside the local network of the machine. But asterisk keeps using the public IP. |
| 20:38.41 | Derkko | I been trying to research how to make it use the internal ip but im at a loss |
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| 21:12.45 | drmessano | Derkko: there's local network settings for both chan_sip and chan_pjsip |
| 21:13.13 | Derkko | I did add the local network on chan_sip |
| 21:13.24 | Derkko | but still ues the public ip |
| 21:34.31 | drmessano | Pastebin what you added for NAT |
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