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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 …
vicidial.org • View topic - sip_call: No audio format ...
http://www.vicidial.org/VICIDIALforum/viewtopic.php?t=25966
[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 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
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.
[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…
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.
[Jun 22 10:08:38] WARNING[28684]: chan_sip.c:4607 …
https://pastebin.com/KFn7TfUj
[Jun 22 10:08:38] WARNING[28684]: chan_sip.c:4607 sip_call: No audio format found to offer. Cancelling call to 5228
No outgoing audio on SIP calls - General Help - FreePBX ...
https://community.freepbx.org/t/no-outgoing-audio-on-sip-calls/22162
Ok, first of all forgive me, I really can’t get my head around this NAT stuff. Basically, I’ve got outgoing/incoming calls connecting but no audio coming inbound when I initiate the call. However I have two way audio when the call is incoming from the external carrier. Asterisk 11.8.1 Freepbx 2.11.0.31 Carrier: Gamma Settings first, then a log at the bottom: …
vicidial.org • View topic - how to check if my server is hack?
http://www.vicidial.org/VICIDIALforum/viewtopic.php?t=39743
[Oct 10 16:13:06] WARNING[13398][C-0047e583]: chan_sip.c:6276 sip_call: No audio format found to offer. Cancelling call to 61386580543 Check the SDP dialog for the payload type ( you codec setting)
SIP - The Offer/Answer Model
https://www.tutorialspoint.com/session_initiation_protocol/session_initiation_protocol_the_offer_answer_model.htm
The use of SDP with SIP is given in the SDP offer answer RFC 3264. The default message body type in SIP is application/sdp. The calling party lists the media capabilities that they are willing to receive in SDP, usually in either an INVITE or in an ACK. The called party lists their media capabilities in the 200 OK response to the INVITE.
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