[asterisk-users] Call Manager as trunk

[Aaron Fransen]

I can't believe I fixed the problem, but here's what I did:

1. Checked the "Use Media Termination Point" in the profile for the SIP
trunk in Call Manager.
2. Split the SIP config for Call Manager into separate inbound and outbound
settings like so:
3. Added the "insecure=very" to the "callmanout" section.

[callmanout]
type=peer
context=incoming
insecure=very
host=(ip of server)
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes

[callmanin]
host=(ip of server)
type=user
context=incoming

And suddenly it's working great!

Aaron

On Thu, Mar 6, 2008 at 9:54 AM, Aaron Fransen <aaron.fransen at gmail.com>
wrote:

> I have Asterisk 1.4 tied via SIP to a Cisco Callmanager 6.1 system. Calls
> between the systems (ie. extension to extension) work perfectly.
>
> However when I attempt to make an outside call from an Asterisk extension
> through Call Manager to the outside world, it connects but only for a few
> seconds, and on the Asterisk console I get:
>
> Got SIP response 503 "Service Unavailable" back from (ip of call manager)
>
> Coming the other way, if I call into the Call Manager system (from my cell
> to be exact), then transfer my call to the Asterisk SIP phone (an Aastra
> 57i), on the cell I can hear the voice on the Aastra, but the Aastra can
> only hear the Asterisk music on hold! As I mentioned though, going the other
> way and calling out from Asterisk to my cell works perfectly...for between 5
> and 10 seconds (it varies), then disconnects with the above error.
>
> My sip.conf looks like this:
>
> [callman]
> type=friend
> context=incoming
> host=(ip of call manager)
> disallow=all
> allow=ulaw
> allow=alaw
> nat=yes
> canreinvite=yes
> qualify=yes
>
> I've tried experimenting with the "externip" and "localnet" parameters to
> no effect.
>
> Any ideas?
>
>
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