[OpenSIPS-Users] re-INVITE (Hold) looses audio

[Mark Farmer]

Hi all

I'm trying to solve an issue where if the call is placed on hold via a
re-INVITE, my end stops the audio at that point so when another re-INVITE
is received to resume the call, the audio does not restart. I can still
hear the other party but they cannot hear me because I'm not sending any
RTP any more.

I'm really struggling to pin this one down. Can anyone give me any clues as
to why this might be happening?

I'm using OpenSIPS 2.4.5 with RTPProxy 2.0

Many thanks
Mark.


-- 
Mark Farmer
farmorg at gmail.com
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