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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 Sound on External SIP in Asterisk|FreePBX
https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
If you have No Audio - It's to do with your ROUTER or FIREWALL. If you've configured the ports as I've shown above, there are only a few things left to check. 1. Internal firewall. If you're using a soft-phone on a machine, make sure the firewall is disabled for testing.
Asterisk 16.4 & WebRTC = no audio - Asterisk WebRTC ...
https://community.asterisk.org/t/asterisk-16-4-webrtc-no-audio/83624
Hello there, we’ve just tried our fist webrtc test with Asterisk 16.4.0 The endpoints get registered well and it’s able to send calls without problems … but without audio. This is the Asterisk CLI: -- Executing […
Asterisk - One way audio with PJSIP over PRI - Stack …
https://stackoverflow.com/questions/31031273/asterisk-one-way-audio-with-pjsip-over-pri
3) The problem occurs only on outgoing calls over PRI (no problem when using IAX2 trunk). 4) The issue does not occur when I originate the call via AMI. 5) Asterisk Version - 13.4.0 Freepbx Version - 6.12.65-26 PJSIP version - 2.2 DAHDI version - 2.10.0.1 I know PJSIP is experimental but I have been able to use PJSIP over PRI before.
PJSIP no audio on calls - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/pjsip-no-audio-on-calls/85165
Everything worked perfectly on chansip. I moved to PJSIP and I can’t hear audio on any of my calls. NAT was nat=force_rport,comedia for extensions 200 and 300 and nat=no for 100. It worked well and I used the python script to convert sip.conf to pjsip.conf. If I enable direct media, I’m able to hear one-way. PJSIP config: removed RTP debug shows no log, nothing.
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