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Hi Geoff,
You might want to try tcdump, specifying the source
and destination IP (to minimize the info)
and see where are the RTP packets going ;
you will see if they change port or
something like that
after a while.
Cheers,
Frederic
----- Original Message -----
Sent: Tuesday, April 25, 2006 17:37
Subject: [Asterisk-Users] One Way
Audio....in the middle of a call
We had a user report that they were on a SIP <---> PSTN
call for about 4.5 minutes before the call went to on-way audio. The user
called the person back and they reported being able to hear my user, but my
user couldn't hear them. The audio condition persisted for about 15 seconds
before the user hung up.
Where do I start to troubleshoot one way
audio that occurs during a call?
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