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No audio format found to offer. Cancelling call - Asterisk ...
https://community.asterisk.org/t/no-audio-format-found-to-offer-cancelling-call/24048
When I call from the Telematrix phone I don’t have any issues, but when I call from the Polycom phone to the Telematrix I get the following error – Executing [3310@Back_Office:1] Dial(“SIP/6700-08372ee8”, “SIP/3310|10”) in new stack [Nov 11 10:57:24] WARNING[24820]: chan_sip.c:3024 sip_call: No audio format found to offer. Cancelling call to Room3310
WEBRTC no audio - Asterisk WebRTC - Asterisk Community
https://community.asterisk.org/t/webrtc-no-audio/90896
WEBRTC no audio. Asterisk Asterisk WebRTC. bobtarask December 11, 2021, 6:19pm #1. Hello there ! I’m trying to use WEBRTC with SIPJS and Asterisk (Asterisk GIT-16-89cf7899be) . It works very well with Chrome version ~50. Since I’m using chrome 96 on Android 10 audio seems not working at all…. Maybe codecs change?
Video but no audio - Poly Community
https://community.polycom.com/t5/Video-Endpoints/Video-but-no-audio/td-p/50039
The new HDX 8000 we added to our private network was able to transmit video but there is no sound. Both teh near and far sites see each other but we can't hear each other. I understand it may be a firewall issue but I was told that the video would be the problem if that was the case and not the audio. Any suggestions will be greatly appreciated.
No audio on inbound or outbound calls - General Help ...
https://community.freepbx.org/t/no-audio-on-inbound-or-outbound-calls/47303
I am running FreePBX with Asterisk version 15.1.5. I am using an old ObiHai 110 device as an FXO port and a Gigaset C530IP DEC station as a PJSIP extension. Both devices register with PBX and calls can be made and received but there is no audio in either direction. When calling the extension’s voicemail, the logs show that the proper audio files are played by …
no audio with asterisk 13 pjsip - Stack Overflow
https://stackoverflow.com/questions/37849325/no-audio-with-asterisk-13-pjsip
Show activity on this post. I cannot playback any wav files from the dialplan or get any audio at all with asterisk 13 pjsip. Server is not behind a nat, endpoints (clients) are. Chan_sip works perfectly just not chan_pjsip pjsip.ini: It seems that the rtp …
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