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

David Backeberg dbackeberg at gmail.com
Sun May 17 07:44:08 CDT 2009


On Sat, May 16, 2009 at 10:22 AM, Timothy Smith <timotsmith at gmail.com> wrote:
> I have finally managed to get voice working. I both parties can hear
> each other. The problem was nating. Our network is fairly big and
> these machines are atleast 2 switches from each other. I just enabled
> it (nat=route or nat=yes) and it worked.
>
> It's not yet done however. When I redirect a call to any Asterisk
> application, it just hangs up! I have read some history and archives,
> but none of the solutions has worked for me. e.g ip inspect udp
> idle-time 900. My router (or IOS) doesn't have thet command.
>
> Could you please assist point to what could be causing this and how to
> solve it? Below are some logs and attached is the router log.
>
> ; This is the extension conf. Enter the extension you want to reach
> now (something like auto attendant).
> exten => _X.,1,Read(NUM,beep,4,2,3)
> exten => _X.,n,Dial(SIP/${NUM})
>
> ; This is all i get when i call and the call hangs up!

Did you ever set up that reverse dial-peer? If not, do that first.

You put a three second timeout on the Read(). By any chance, is the
call hanging up 3 seconds after you call? That would be expected
behavior. Well, actually you give it two tries. So it should be
beep
three second wait
beep
three second wait
hangup

If you're actually entering numbers on your dialpad and they're not
getting read, you have a misconfiguration on your DTMF. If you enable
sip debugging on your asterisk side you can see exactly what's coming
over the wire from the Cisco side. There are a lot of choices for DTMF
on the asterisk side and the Cisco side, and they need to agree for
the button presses to be encoded and passed correctly. You can pass
them in-line as real audio, or you can convert them to a special dtmf
sip encoding. You'll notice all those choices when you go to configure
the Cisco dial-peer.

My personal preference:
on the Cisco dial-peer side
 dtmf-relay rtp-nte

on the asterisk side
I left the dtmf config blank, and I don't remember which default you
end up with, but it worked in the default config for me.



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