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No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
No audio on Asterisk SIP call. Ask Question Asked 10 years, 10 months ago. Active 1 year, 10 months ago. Viewed 34k times 9 I almost managed to init a 2 sided call (click to call): 1st to my office and the 2nd to my cell using Michal Niklas answer (thanks Michal) on Asterisk click to call. The major ISSUE is that the 2 call participants can not ...
No audio - Fresh install - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/no-audio-fresh-install/83257
Hello everyone, I’m fairly new to Asterisk and have a problem regarding audio. I have recently set up a VPS with CentOS 7 and Asterisk 16.9.0. It’s a basic, fresh install I did by compiling Asterisk myself. I used the instructions from the book “Asterisk: The Definitive Guide (Fifth Edition)” to setup everything. Endpoints are setup using a realtime connection with a …
No audio on chan_skinny 7941G - Asterisk Support ...
https://community.asterisk.org/t/no-audio-on-chan-skinny-7941g/37040
Hi all, as many others, first post 😃 I’ve got some problem with my Cisco 7941G and Asterisk (1.8.7.0). Before listing my problem, I want to explicit that I have a Cisco 7941G (firmware: SCCP41.8-4-2S), a SIP phone (Polycom IP320 SIP or Zoiper Free, but this is not influent) and Asterisk on a Ubuntu Server. Below there is my skinny.conf: [general] disallow=all …
directrtp=on - no audio · Issue #405 · chan-sccp/chan …
https://github.com/chan-sccp/chan-sccp/issues/405
Skinny Client Control Protocol (SCCP). Release: 4.3.2 (HEAD - 558d2b6M) As far as I can tell DirectRTP between SCCP endpoints on asterisk (7912/7940/7960) working correctly. But there is No DirectRTP in call between SCCP and PJSIP endpoint. RTP flow goes through the PBX for each call leg. Is it possible at all, in case of different signaling ...
SIP -> Local -> SCCP one way audio because of wrong …
https://github.com/chan-sccp/chan-sccp/issues/420
Install revision c461215. Make a call with local channel from a sip phone with (PJSIP) allow: (g722|alaw|ulaw|opus|gsm|g726|g729) and with g722 as first priority in the invite. Asterisk is then sending G722 to the 7965 and no sound. Expected behavior: Before c461215 there was sound in both directions.
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