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asterisk-list at puzzl...
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 Posted: Tue Mar 11, 2008 7:33 am Post subject: [asterisk-users] CCM 6 and Asterisk routing again |
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On Tue, 2008-03-11 at 06:11 -0600, Aaron Fransen wrote:
[snip]
| Quote: |
Here's my sip.conf if that helps...
[callman]
type=peer
context=incoming
insecure=very
host=(ip of my call manager server)
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes
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Without any logs it's difficult to say what's going on. Assuming that
you want the signaling & media from the CCM to the IP phones behind the
Asterisk box to go through your Asterisk box set canreinvite to "no"
instead of "yes".
Regards,
Patrick |
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ppauly at gmail.com
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| Posted: Tue Mar 11, 2008 7:38 am Post subject: [asterisk-users] CCM 6 and Asterisk routing again |
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I've noticed two differences in what you described and my working CM setup:
1. My sip trunk in CM is defined as 711alaw, you have ulaw.
2. My sip.conf defines CM as a type=friend instead of a peer.
Do you have any SIP phones connected to Asterisk (you could use a
softphone like the free xten)? Can you call the phone from
CallManager?
Peter Pauly
http://www.usbtests.com
On 3/11/08, Aaron Fransen <aaron.fransen at gmail.com> wrote:
| Quote: |
Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1,
Asterisk is running strictly VoIP over the network and using CCM as the
trunk.
Calls from the SIP phones connected to Asterisk work fine. They can call
both external numbers and any Cisco extensions attached to CCM.
Calls from CCM to Asterisk fail without any notification in Asterisk (and I
DID have this working at one point, but I suspect that our Telco may have
pooched the config somehow, since they're in the process of connecting us to
another CCM site).
I have verified: Media Termination point exists, Calling Search Space
exists, Trunk is properly defined (uLaw 711, UDP, ip address & port, etc),
and a route pattern exists to take calls to the right trunk.
The system will let me complete the dialing sequence to the Asterisk
server, but as soon as I enter the last digit I get a busy signal.
Thoughts anyone?
Here's my sip.conf if that helps...
[callman]
type=peer
context=incoming
insecure=very
host=(ip of my call manager server)
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes
Thanks! Aaron
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