[asterisk-dev] Start testing with res_sip

[Mark Michelson]

On 06/20/2013 11:04 AM, Kevin Harwell wrote:
>
> Hi Ron,
>
> I believe you will need to define an endpoint for 1000 in res_sip.conf 
> as well.  It is trying to find the endpoint to invite, but it hasn't 
> been defined.  Hopefully that will work.
>
>

That shouldn't be necessary if he's just doing a simple playback to 101. 
As long as there is an extension 1000 in context "testing" then Asterisk 
should be able to find the destination. Now if extension 1000 attempts 
to do something like Dial(GULP/1000) then yes, you'd need a 1000 
endpoint defined in res_sip.conf.

Mark Michelson