Hi,<div><br></div><div>We've got three parties for this example: A, B, C</div><div><br></div><div>A - Asterisk end-point Polycom</div><div><br></div><div>B - Asterisk end-point Polycom</div><div><br></div><div>C - Outside end-point Uniden</div>
<div><br></div><div>OpenSIPs sits in front of the Asterisk servers and communicates with a carrier C5 switch directly (same local area network inside a lab facility)</div><div><br></div><div>A calls C - call completes - talk, no issues.</div>
<div><br></div><div>C presses the transfer button, which is a flash-hook putting A on hold. C dials B.</div><div><br></div><div>B answers the call - both parties talk.</div><div><br></div><div>C presses the flash-hook button again in order to complete the transfer.</div>
<div><br></div><div>A can hear B - B cannot hear A.</div><div><br></div><div>The RTP debug from Asterisk shows that RTP packets from B are still going to C.</div><div><br></div><div>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.</div>
<div><br></div><div>I don't have control of the Carrier-side C5 to check, and they have been slow to provide me with a wireshark trace. </div><div><br></div><div>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?</div>
<div><br></div><div>Thanks in advance.<br clear="all"><div><br></div><div>Tyler</div>
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