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
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