Howdy global friends,
I have SIP reinvite working fine from asterisk 11.5 with snom 3xx series on 7.4.x.x firmware, no problems. reinvite=yes, dtmfmode=info, nat=no. Very nice audio actually, love the low latency audio.
Now I enable the extensions with TLS and SRTP, making the changes to the extension in asterisk and on phone. Excellent, I now get secure locked symbol calls between the Snoms.
BUT, the SRTP stream is going via Asterisk, and not directly to the other peer extension.
Review of asterisk verbose has:
Connected line update to SIP/305-00000c5d prevented.
I’m curious if others have found this, or if its a limitation within asterisk, or a protocol or rule issue somewhere.
Update: Have gone through the asterisk debug and it appears there indeed may not be code written in the devicestate to handle SRTP been bridged… Havent checked into the source code thou… maybe someone can help out here with some comments.