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

[Philip Edelbrock]

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:geoffm33 at gmail.com>
>     *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>     <mailto:asterisk-users at lists.digium.com>
>     *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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