[asterisk-users] asterisk 11.7.0: Delayed audio

[gm1]

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