[asterisk-users] Fwd: Asterisk With Cisco Voice Router

[David Backeberg]

On Sat, May 16, 2009 at 7:46 AM, Timothy Smith <timotsmith at gmail.com> wrote:
> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
> and also a dialpeer to forward on the router to forward calls to my
> asterisk. It works properly but the problem is there is NO AUDIO! I
> have tried to change codec but no sucess!
> Has anyone had the above set up working successfully?

Yes.

You have been caught by a not-very-well-documented issue with setting
up voice routing on the 3845, and probably other similar Cisco gear.
And I'm not sure how you've done your test.
This is the closest I've ever seen to a document that explains your problem:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml

Did you have a SIP phone on one side of asterisk and a POTS phone on
the outside of the 3845?

If you did, and you could talk on both at the same time, I think you
would discover in fact that you do have some audio, in fact, one-way
audio to be precise. But I don't remember for sure, because it's been
a while since I've done this to myself.

At any rate, your problem is you have dial-peers to get voice packets
out from the 3845 to Cisco, but no dial-peers to get the packets from
SIP back to a physical circuit on the 3845. Think about this. What
should happen to a call inbound from asterisk, to the 3845? Should it
go out an E1 to the outside phones world? If so, you need to build a
dial-peer that does that. Until you do, you won't be getting two-way
audio.

you need another rule something like:
dial-peer voice 790792888 pots
map this back to the proper E1 circuit

A secondary problem could also be with the way you're managing your
DSPs. I don't know how many physical DSPs you have in your router, but
usually it's a GOOD thing to enable DSP farming.