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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
    – 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 – 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

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
    [Aug 2 15:00:31] VERBOSE[1918] chan_sip.c: Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - …

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

    http://www.vicidial.org/VICIDIALforum/viewtopic.php?t=25966
    ("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 …

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...

[asterisk-dev] Sip call consciously without audio

    https://asterisk-dev.digium.narkive.com/EQ2it8YM/sip-call-consciously-without-audio
    chan-sip.c in release 11 and 12 is dropping sip calls in which no common audio media is negotiated, regardless of if other media are negotiated. There are very valid cases when a call is wanted with any combination of video and text, but no audio. One place where calls without audio are blocked is in the function: sip_request_call

[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…

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