On Mon, 9 Feb 2009, Jeff LaCoursiere wrote: > > I've never used "reinvite" in systems I have installed to date, and I have > finally run across a situation where it would be preferred. > > A remote office has a flaky Internet connection. With G729 encoding the > calls to the central office over the 'net are tolerable. One Linksys 2102 > drives two phones at this location, and when the first one calls the second > one it travels to the central office and back, which is no longer tolerable. > > For each sip peer I have "canreinvite=yes", but I am a bit confused as to the > correct options on the 2102 to use this feature. Is anyone doing this with > 2102s that can give me some pointers? >
I have been playing around with this in my "lab" and cannot seem to make it work as expected. I have a remote asterisk server on a public IP - 1.4.22-3 on Centos 5. I have two Polycom IP501s on a local LAN behind a NAT gateway. Both Polycom's register with the remote server and can call each other without issues. Both SIP contexts have nat=yes, canreinvite=yes. The caller is 223, the callee is 222. eth0 is the outside (public) interface, XXX is my dynamic IP. I trapped a conversation on the asterisk server with: tcpdump -nli eth0 -s 0 -w /tmp/reinvite.debug host XXX and not port 22 While this was running I made a call between the two extensions for a few seconds then hungup. I opened this capture in etherreal and can see the following: 223->AST INVITE 2...@ast AST->223 407 Proxy auth required 223->AST ACK 223->AST INVITE 2...@ast, with proxy-auth info AST->223 100 Trying AST->223 200 OK 223->AST ACK Then I see the RTP traffic begin back and forth. I am confused on two fronts - first where is the INVITE from AST to 222? Not sure how I missed capturing that side of the conversation. And of course where is the AST "reinvite"? It isn't occurring since I can clearly see the RTP traffic flowing via the asterisk server. Any ideas? Cheers, j _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
