Asterisk Sip_Call No Audio Format Found To Offer

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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

No audio format found to offer. Cancelling call. Asterisk. Asterisk General. Osiris123d November 11, 2008, ... 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.

[SOLVED] Asterisk realtime - No audio format found ...

https://community.asterisk.org/t/solved-asterisk-realtime-no-audio-format-found/44204

SIP phones from sip.conf working fine. I can even call to “mysql” phone - and it works for incoming calls too, but when I’m trying to make a call from “mysql” phone this error occurs: Errors in Asterisk cli: [Oct 18 14:43:50] WARNING[19430]: chan_sip.c:6023 sip_call: No audio format found to offer. Cancelling call to 1999

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

[asterisk-dev] Sip call consciously without audio

https://asterisk-dev.digium.narkive.com/EQ2it8YM/sip-call-consciously-without-audio

There is a similar check for outgoing calls in chan-sip.c function 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;

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.

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.

vicidial.org • View topic - sip_call: No audio format ...

http://www.vicidial.org/VICIDIALforum/viewtopic.php?p=90518

("No audio format found to offer")" Code: Select all [Sep 28 18:30:37] VERBOSE[4304] logger.c: [Sep 28 18:30:37] -- Executing [618566244425@default:2] Dial("Local/618566244425@default-b217,2", "SIP/xcast/18566244425||Ttor") in new stack [Sep 28 18:30:37] WARNING[4304] chan_sip.c: No audio format found to offer. Cancelling call to …

Asterisk : No compatible codecs, not accepting this offer ...

https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820

Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 [Aug 24 00:06:01] NOTICE[15303][C-00005e94]: chan_sip.c:10753 process_sdp: No co mpatible codecs, not accepting this offer! <— Reliably Transmitting (no NAT) to 198.101.50.4:5060 —> SIP/2.0 488 Not acceptable here

Problems with trunks after updating all the modules ...

https://community.freepbx.org/t/problems-with-trunks-after-updating-all-the-modules/57961

[email protected]

Process_sdp: Failing due to no acceptable offer found ...

https://community.asterisk.org/t/process-sdp-failing-due-to-no-acceptable-offer-found/90344

Hi, I’ve an Asterisk version 13.10.0. We’ve a situation in which Asterisk reject a call once is answered in destination with the following error: “process_sdp: Failing due to no acceptable offer found” The Sip.conf of the outgoing trunk is the following: [TRUNKSIP-SBC_ACCESS] type=peer host=x.x.x.x context=incoming-SBC_ACCESS disallow = all allow = …

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