Hi everybody
in new versions of Asterisk the RTP on SIP pass only througt the Asterisk, not directly between the endpoints like olders versions.
What happened whit this feature? (reinvite) Can you help me?
The the two legs of the call are using different codecs then reinvites won't work. If you are using t or T option to dial (maybe others), Asterisk has to stay in the media stream to listen for the #. Check your codecs.
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