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.<br><br>scd<br><br><div><span class="gmail_quote">On 4/16/07,
<b class="gmail_sendername"><a href="mailto:shawnl@up.net">shawnl@up.net</a></b> <<a href="mailto:shawnl@up.net">shawnl@up.net</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I'm wondering about the difference between Cisco Call Manager and<br>SCCP(2) network traffic. I'm working on getting a Cisco 7960 phone to<br>speak through a NAT to an asterisk box, without having to do a bunch of port
<br>forwarding on the NAT device.<br><br>Without the nat, everything works fine.<br><br>If the phone is behind a cisco pix that is doing the natting, it works<br>fine (fixup protocol).<br><br>If the phone is behind a more generic nat device, such as a linux box
<br>running ipfilter. Then it can dial out, but there is no audio. The<br>interesting part is that this same phone, behind the same NAT works just<br>fine if it is talking to a Cisco Call Manager box instead of an<br>asterisk server. So, I'm wondering what the difference in the protocols is
<br>(I no longer have access to the call manager box, so I can't look @ the<br>traffic). In a perfect world, I'd like to have the phone pretty much just<br>work wherever it's plugged in as long as it can see the asterisk server.
<br><br><br>Any ideas ?<br><br><br>Thanks<br><br><br>Shawn<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list
<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>
-- <br>Steve Dickey<br>Who is John Galt?