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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
[Nov 11 10:57:24] WARNING[24820]: chan_sip.c:3024 sip_call: No audio format found to offer. Cancelling call to Room3310 – Couldn’t call 3310. Why doesn’t it skip the 722 codec and move on to the next Codec that I have listed in the sip.conf
No audio format found to offer. Cancelling call to.....
https://forum.asterisk2billing.org/viewtopic.php?t=3083
Found user 'xxxxxxx' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.103:2226 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101
[SOLVED] Asterisk realtime - No audio format found ...
https://community.asterisk.org/t/solved-asterisk-realtime-no-audio-format-found/44204
Hello. I have problems when trying to make call from 1 sip user to another sipuser, I'm using Asterisk Realtime SIP and Extension with mysql. SIP phones from sip.conf working fine. I can even call to "mysql" phone - a…
sip_call: No audio format found to offer. Cancelling call to
https://groups.google.com/g/asterisk-es/c/Z2VN8vic2Uo
[Jun 10 14:54:35] WARNING[1933]: chan_sip.c:3001 sip_call: No audio format found to offer. Cancelling call to 202 -- Couldn't call 202 == Everyone is busy/congested at this time (0:0/0/0) No recibo la llamada porque parece haber un problema de formato de audio. En sip.conf tengo: [general]... disallow=all allow=alaw allow=gsm...
Codec negotiation issue (no audio format found to offer)
https://asterisk-users.digium.narkive.com/eGEktuMi/codec-negotiation-issue-no-audio-format-found-to-offer
And then this, no INVITE goes out to callwithus at all: [Aug 2 15:00:31] WARNING[1315] chan_sip.c: No audio format found to offer. Cancelling call to ***** [Aug 2 15:00:31] VERBOSE[1315] app_dial.c: -- Couldn't call SIP/CallWithUs/***** Similarly, if I set the Grandstream FXO trunk to ulaw only, the call fails as well.
[asterisk-dev] Sip call consciously without audio
https://asterisk-dev.digium.narkive.com/EQ2it8YM/sip-call-consciously-without-audio
sip_call : /* If there are no audio formats left to offer, punt */ if (!(ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_AUDIO))) {ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username); res = -1; I suggest that the check in both these places is replaced with a check for any common supported media and …
webrtc - sipjs and asterisk voice call no audio issue ...
https://stackoverflow.com/questions/40396973/sipjs-and-asterisk-voice-call-no-audio-issue
I am using sipjs 0.7.5 version and asterisk 13.12.1 to established call between 2 sipjs call through webRTC. ... NOTICE[64407][C-00000008]: chan_sip.c:10315 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 55294 UDP/TLS/RTP/SAVPF 109 9 0 8 Found RTP audio format 109 Found RTP audio format 9 Found …
vicidial.org • View topic - sip_call: No audio format ...
http://www.vicidial.org/VICIDIALforum/viewtopic.php?p=90518
[Sep 28 21:32:42] WARNING[4100]: chan_sip.c:3346 sip_call: No audio format found to offer. Cancelling call to 17275551212 [Sep 28 21:32:42] -- Couldn't call xcast/17275551212
No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
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 hear one each other, I used from-internal context for both of them.
upgraded to 3.3 and getting call errors! - GOautodial Open ...
https://goautodial.org/boards/1/topics/3283?r=3287
yes codec problem "[Mar 2 16:50:38] WARNING8363: chan_sip.c:6033 sip_call: No audio format found to offer. Cancelling call to 441246861496"
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