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.
scd On 4/16/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> 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 port 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 is (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 ? Thanks Shawn _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- Steve Dickey Who is John Galt?
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