[Asterisk-Dev] SIP canreinvite=yes Broke?

[Dave Packham]

So I have many Cisco 7960's that are running the latest 5.1 Cisco SIP code and I cannot get the phones to talk/RTP to each other.  jtodd has had this problem in the past with the 186's.  Just wondering if anyone has a reason why "Cisco sometimes poop on reinvite"  is the Cisco code broke?  if so we can push on Cisco to fix it.  the U is a MAJOR Cisco shop so we have some puhs there.  if its * code,  I will offer up anything (within reason) to work this out.

This prob would be a major issue in rolling * out further if every call HAS to go thru the * server fro bridging.

Do other SIP hardsets have this problem?  sniff you calls to another SIP hardset and check ot see if RTP is coming from the * server of the other phone?

Thanks for any info I can get on this

Dave P