On Fri, Dec 5, 2008 at 2:05 PM, Brian West
<brian@freeswitch.org> wrote:
But you don't see the invite hitting FreeSWITCH? And you're behind
NAT? Make it register every 30 seconds instead of the default 3600
/b
On Dec 5, 2008, at 10:59 AM, MEHDi CHAABOUNi wrote:
> Actually, i did not mean that the line is dropped during a call...
> FS is configured to accept calls from the Junction Networks SIP
> trunk to make an audio conference.
> When I start FS and I dial the number all is working fine. But, if I
> wait for a couple of minutes and then make my call I get an error
> recorded message saying that the number is not in service...