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
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 ;
youwill see if they change port or
something like that
after a while.
Cheers,
Frederic
- Original Message -
From:
Geoff
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
Is there an easy fix?- Dan L.On 4/25/06, Philip Edelbrock [EMAIL PROTECTED] wrote:
I experienced this today.Doing a 'show channels' in Asterisk showed aZap line perpetually ringing the sip phone even though the sip phone wasreset a few times.Doing a 'soft hangup' on the stuck Zap and the Sip
