Hi All, Just curious if anyone else has seen this, I opened a JIRA. My Configuration: * Voip ISP w/ preferred CODEC PCMU:20 * Phone A - Cisco SPA504G w/ pref Codem PCMU and PTime 20 * Phone A - Cisco SPA504G w/ pref Codem PCMU and PTime 30 My ISP calls go through an IVR. Calling through the IVR to Phone A is perfect, PCMU:20 <->PCMU:20. Calling in through IVR to Phone B is still great, FS detects the Ansych PTIME and adjusts to PCMU:20<->PCMU:20. But if the call goes through the IVR to Phone A (PCMU:20<->PCMU:20), and THEN is (attended) XFERed to Phone B, the result is a badly delayed (3 to 5 second) and degraded audio stream PCMU:20<->PCMU:30. Calling direct to Phone B, FS says "Asynch ptime not allowed, changing our end from 30 to 20." Freeswitch does not detect the same asynch ptime mismatch at the end of the attended xfer! I understand changing Phone B resolves this problem, but it is still a bug (I think). Sean
