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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
So asterisk knows that it is going to dial a peer that supports only g729 when it gets an invite from a peer that supports both ulaw and g729. Using 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
[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 format found to offer ... - asterisk2billing.org
https://forum.asterisk2billing.org/viewtopic.php?t=3083
Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
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
Asterisk : No compatible codecs, not accepting this offer ...
https://community.asterisk.org/t/asterisk-no-compatible-codecs-not-accepting-this-offer/67820
Found RTP audio format 101 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_call: No audio format found to offer. Cancelling call to
https://groups.google.com/g/asterisk-es/c/Z2VN8vic2Uo
sip_call: No audio format found to offer. Cancelling call to ... una tarjeta X100P en mi Asterisk y un par de canales sip internos. Puedo llamar al exterior (el dialplan funciona aparentemente bien) pero al intentar recibir una llamada me encuentro con: ... load => codec_g729.so ; G729/PCM16 (signed linear) Codec Translator - Requires N/A
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