On Mon, 2006-03-06 at 15:42, Jerry Geis wrote: > here is some of the output. I am no longer the to spcifically do sip > debug but this is what I have. > along with my sip.conf snip. > > The call to extension 3726 never rings. so it never gets answered. >
Are you sure your sip trunk and route pattern are in the same partition/CSS by chance? Without more info (AGI script and SIP debug), I really can't be much more help. Your sip.conf entry is good though. Your callmanager context from extensions.conf will help as well. -Greg _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
