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

[Dan Levy]

Is there an easy fix?

- Dan L.

On 4/25/06, Philip Edelbrock <phil at netroedge.com> wrote:
>
>
> 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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