[Asterisk-Users] SIP registration/dialing problem.

[Scott Laird]

On Nov 3, 2004, at 4:16 PM, Ben Greear wrote:

> Hello!
>
> I have a Grandstream and a Cisco SIP phone, and I'm trying to make
> a call between them.  I added this to my sip.conf:
>
> ; Grandstream
> [1001]
> type=friend
> host=dynamic
>
> ; cisco phone
> [1002]
> type=friend
> host=dynamic

First, what's in your extensions.conf?  That controls the flow of calls 
once they get into the system.  There should be a context that has 
extensions for 1001 and 1002, and sip.conf should direct calls into 
that extension via a 'context =' line.

Running an Asterisk console in verbose mode (asterisk -vvvvvr will 
connect to a running server) provides a lot of useful debugging 
information.


Scott