[Asterisk-Users] PSTN to SIP gateway

[Nick Kartsioukas]

On Fri, Jul 15, 2005 at 01:28:45AM -0400, Jose Raborg wrote:
> Do you want to route the calls depending on the caller id? Or Do you
> want to assign a DID to a SIP?

The remote SIP device will route the calls appropriately based on the
information sent to them (the "*ANI*DNIS*" sent as an "extension"), I
just want to take any incoming calls from the PSTN to get forwarded to
the other SIP gateway.

So far I've gotten Asterisk to say:
	    -- Extension 'XXXXXXXXXX' in context 'pstn' from '' does not
	    exist.  Rejecting call on channel 0/23, span 1
(where XXXXXXXXXX is the phone number I dialed)
So, that's a start, I guess ;)

sip.conf contains:
[voip]
type=peer
host=remote.sip.gateway

zapata.conf contains:
[channels]
language=en
switchtype=national
context=pstn
signalling=pri_cpe
group=1,24
channel => 1-23

extensions.conf contains:
[pstntosip]
exten => _X,1,Dial(SIP/*${CALLERIDNUM}*${DID}*@voip)

I'm sure I'm missing a vital config option elsewhere...

-- 
Nick Kartsioukas
Sky Way Networks, LLC