[Asterisk-Users] One Way Audio....in the middle of a call

[Frederic Jean]

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 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  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?



------------------------------------------------------------------------------


  _______________________________________________
  --Bandwidth and Colocation provided by Easynews.com --

  Asterisk-Users mailing list
  To UNSUBSCRIBE or update options visit:
     http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060425/fde5f2a2/attachment.htm