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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
the offer / answer model it would look like this: Peer -> Invite (SDP:ulaw,g729) -> Asterisk Peer <- 100 Trying (w/ SDP -- g729 only) <- Asterisk Peer -> 200 OK (w/ SDP g729) -> Asterisk I understand your point about not knowing what may happen after initial call setup, but the same implementation would apply in the event of a reinvite.
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.
[asterisk-dev] Sip call consciously without audio
https://asterisk-dev.digium.narkive.com/EQ2it8YM/sip-call-consciously-without-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 ... but what an Asterisk should do in an MSRP call beats me. Like with XMPP, we can start chat bots, but that hasn't really happened during all the years we've had XMPP ...
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
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"
No Sound on External SIP in Asterisk|FreePBX
https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
No sound with SIP | No Sound With Asterisk| No Sound With NAT. To correctly troubleshoot your issue, make sure that the following steps are taken. On your router, open up the following ports and forward them to your asterisk box. Your asterisk box may be in the form of (FreePBX, AsteriskNOW, or TrixBox)- They all use the same format.
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