On connection to an incoming call via PSTN where
asterisk [11.7.0] is Dialing an internal extension
on answering the call there is about 6-7 seconds before
audio is heard on either side.
When looking at the CLI traces when I answer the incoming call that
asterisk extensions were dialing, I see immediately upon answering
>0xhexnumber -- Probation passed - setting RTP source address to
192.168.1.11:portnumber
then not until about 6 seconds later I see this
>0xhexnumber -- Probation passed - setting RTP source address to
192.168.1.11:diffportnumber
and immediately hear audio
what appears to be an issue is that the RTP link(audio) setup is delayed.
Anyone have suggestions on how to fix this issue?
pc
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