SIP phone cannot dial out. CLI displays:
chan_sip.c:10427 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
In GUI, encryption for the extension is disabled. In SiP phone (polycom vvx310) Enable/Offer SRTP = off.
Found no way to fix it. Deleted extension in GUI and recreated same extension number. After changing passwd in SIP Phone, was able to dial out. No changes to the Polycom phone config were made.
in CLI sip show peers does shows extensions beginning with ‘99’ followed by extension number. This ‘99’ extension has Encryption = Yes.
Seems like it is used for webRTC.
Is it the case that the extension I was dialling from was using the values of the 99extension?
This happened to me also. On FPBX 13 after upgrading from 12. Enabled WebRTC in user manager and then enabled DTLS settings in extension. This allowed webRTC phone to work well, but broke the hard yealink endpoint that shared the extension being used in the UCP and webRTC. Had to delete the extension in FPBX, and re-create it, and then the yealink extension worked again. The error I was getting before doing this even after disabling DTLS on this extension in FPBS was:
chan_sip.c: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
No changing of settings for the extension in FPBX gui fixed this behavior and no changes on the endpoint mattered either including a reset to factory defaults. I am using the FreePBX distribution and commercial EPM module also.