[OpenSIPS-Users] RE-INVITEs being sent to original contact doesn't properly adjust RTP ports on transfer?

[Tyler Merritt]

Hi,

We've got three parties for this example:  A, B, C

A - Asterisk end-point Polycom

B - Asterisk end-point Polycom

C - Outside end-point Uniden

OpenSIPs sits in front of the Asterisk servers and communicates with a
carrier C5 switch directly (same local area network inside a lab facility)

A calls C - call completes - talk, no issues.

C presses the transfer button, which is a flash-hook putting A on hold.  C
dials B.

B answers the call - both parties talk.

C presses the flash-hook button again in order to complete the transfer.

A can hear B - B cannot hear A.

The RTP debug from Asterisk shows that RTP packets from B are still going to
C.

B didn't get the RE-INVITE apparently - but I cannot figure out where the
packet is.  It's not showing up in OpenSIPs sip_trace, and it's definitely
not getting to Asterisk.

I don't have control of the Carrier-side C5 to check, and they have been
slow to provide me with a wireshark trace.

Is there anything else I could do in OpenSIPs to determine if the RE-INVITE
is not being handled properly besides what I've already mentioned?

Thanks in advance.

Tyler
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20110204/50c64b57/attachment.htm>