[Asterisk-Users] call manager integration

[Greg Oliver]

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