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| 00:19.03 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0-rc3 (2018/09/20), Security Only: 15.6.1 (2018/09/20); 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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| 14:18.12 | Chotaire | Hi, is there any variable that would display the technology used by the channel? E.g. PJSIP or Local |
| 14:23.09 | file | Check the channel dialplan function |
| 14:24.30 | Chotaire | oh, ${CHANNEL}? |
| 14:24.44 | Chotaire | then something else gets me a status UNKNOWN. |
| 14:25.24 | file | No, there are multiple options that can be passed |
| 14:25.28 | file | Documented on wiki |
| 14:26.27 | igcewieling | "core show function CHANNEL" and go read more about Asterisk |
| 14:26.42 | Chotaire | possibly, thanks for that. But you just made me realize the problem is completely elsewhere, I've already correctly retrieved the technology. |
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| 14:33.58 | Chotaire | missing quotation marks. classic. |
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| 20:12.09 | niekniek | hello! Is it possible there's any SIP going on between 2 devices during a call? For keep alive purposes and such? |
| 20:15.19 | jjg | niekniek: if you can, use wireshark to take a look |
| 20:31.01 | Samot | niekniek: SIP is happening when there is a call. If you mean other messages outside of the INVITE, yes they still happen. NOTIFY, SUBSCRIBE, REGISTER will continue will an active call is happening. |
| 20:31.26 | Samot | Along with any other SIP messages such as OPTIONS |
| 20:46.44 | niekniek | Samot: I know those, but no things to keep the call alive right? For example a 2 hour call with only silence, and no disconnect on broken rtp will stay online for that long right? |
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| 21:01.00 | Samot | Well that is your RTP timeout |
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| 21:37.48 | niekniek | Samot: I learned about sip session timers now, that's what I was searching |
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