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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
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.
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 [Sep 28 21:32:42] Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
Problems with trunks after updating all the modules ...
https://community.freepbx.org/t/problems-with-trunks-after-updating-all-the-modules/57961
[2019-04-15 18:12:38] WARNING[29506][C-00000032]: chan_sip.c:6274 sip_call: No audio format found to offer. Cancelling call to 81. Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 6400 ms (Method: INVITE) Audio is at 18570. Adding codec 100002 (gsm) to SDP. Adding non-codec 0x1 (telephone-event) to SDP
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)
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.
Solved: incoming sip call with no audio drop after 10 ...
https://community.cisco.com/t5/ip-telephony-and-phones/incoming-sip-call-with-no-audio-drop-after-10-seconds/td-p/2158090
Incoming sip calls are disconnecting after 10 sec and there is no audio for either side. the other end is hearing only call progress tone even after my side answers the call. and out going call choosing a different dial-peer than what it should use.
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: …
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