Singer,
Assuming that you have no issues with firewalls in the path regarding
the rtp ports, or hardware/firmware problems, take a look at this patch:
http://www.sineapps.com/news.php?rssid=1019
Please take note if * does not receive rtp packets for any reason, it
does not send either.
Jorge
I have bindaddr=0.0.0.0 in my sip.conf; what my major problem is that it
only happens 5-8% of the time..
On Wed, 2006-12-06 at 09:56 -0500, Ed Nuñez wrote:
> If you use both the public and private interfaces for VoIP in the Asterisk
> Server, make sure you don't specify one of them for the bindi
Singer
I would be interested to see the rest of your configuration pertaining to how
you are recording the calls. I am having trouble with this part.
Are you using monitor or MixMonitor from extensions.conf of are you using the
queues.conf or agents.conf monitor ?
Ed Nuñez
IT/Telecom Engineer
If you use both the public and private interfaces for VoIP in the Asterisk
Server, make sure you don't specify one of them for the binding in sip.conf
Example
bindaddr=0.0.0.0
will allow SIP traffic on any of your interfaces.
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
