[c-nsp] 2600 SIP gateway question.

[Jared Mauch]

On Tue, Nov 08, 2005 at 08:17:06AM -0500, Jeff Crowe wrote:
> Hi all,
> 
> I am using a 2600 as a SIP gateway between the PSTN world and a VoIP switch.
> I have 2 PRI's coming into the 2600 and then being directed towards my VoIP
> switch over a SIP trunk.  Routing of inbound calls are not a problem. The
> PRI's that come into the 2600 are from geographically diverse calling areas
> and I would like to try and keep that traffic separate from each other. 
> 
> My question is:  Is there a way to route VoIP to PSTN traffic in the 2600
> based upon a SIP trunk?  I am trying to avoid using call routing in the 2600
> as it may change quite often and the VoIP switch I have can deal with it
> much easier.  
> 
> EG: if a call comes in from SIP trunk a, it goes out PRI 1, if a call comes
> in from SIP trunk B, it goes out PRI 2.

	you could always assign some dummy prepend code

	ie:

dial-peer voice 1234 pots
 destination-pattern 1234..........
 port 1/0/0

dial-peer voice 1235 pots
 destination-pattern 1235..........
 port 1/1/0


	that way you could prepend an "accounting" or authentication
code of random digits of variable length and
the router only signals out the PRI the wildcarded digits unless you
add 'prefix' as well.


-- 
Jared Mauch  | pgp key available via finger from jared at puck.nether.net
clue++;      | http://puck.nether.net/~jared/  My statements are only mine.