Re: [Asterisk-Users] call manager integration
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
Re: [Asterisk-Users] call manager integration
On Mon, 2006-03-06 at 15:00, Jerry Geis wrote: > I am getting this error from call manager (4.0) and asterisk 1.2.4 > > I have canreinvite=yes on the call manager setup. > > I can call into the asterisk box from call manager. THat seems to work. > When I am calling out of the box using a call file I see > this entry from call manager... > > What might be the problem with my setup? > What is the output on the console with sip debug turned on? -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
