I experienced this today. Doing a 'show channels' in Asterisk showed a
Zap line perpetually ringing the sip phone even though the sip phone was
reset a few times. Doing a 'soft hangup' on the stuck Zap and the Sip
allowed 2-way audio to resume.
Phil
Frederic Jean wrote:
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 -----
*From:* Geoff Manning <mailto:[EMAIL PROTECTED]>
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
<mailto:[email protected]>
*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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