- Dan L.
On 4/25/06, Philip Edelbrock <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED]>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> <mailto:asterisk-users@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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