[asterisk-users] Difference between SCCP and Cisco Call Manager
steve.dickey at gmail.com
Mon Apr 16 05:56:43 MST 2007
Call setup/teardown is handled with the SIP protocol while the actual call
audio is handled with RTP I think. Check the config of your NAT devices
relative to RTP.
On 4/16/07, shawnl at up.net <shawnl at up.net> wrote:
> I'm wondering about the difference between Cisco Call Manager and
> SCCP(2) network traffic. I'm working on getting a Cisco 7960 phone to
> speak through a NAT to an asterisk box, without having to do a bunch of
> forwarding on the NAT device.
> Without the nat, everything works fine.
> If the phone is behind a cisco pix that is doing the natting, it works
> fine (fixup protocol).
> If the phone is behind a more generic nat device, such as a linux box
> running ipfilter. Then it can dial out, but there is no audio. The
> interesting part is that this same phone, behind the same NAT works just
> fine if it is talking to a Cisco Call Manager box instead of an
> asterisk server. So, I'm wondering what the difference in the protocols
> (I no longer have access to the call manager box, so I can't look @ the
> traffic). In a perfect world, I'd like to have the phone pretty much just
> work wherever it's plugged in as long as it can see the asterisk server.
> Any ideas ?
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