I finally have the solution, so thought I would post back to the list for completeness.
It ended up being a series of changes. First, on the gateway, set "Disconnect on Broken Connection" to false. Then, for the Polycom phones, set voIpProt.SIP.allowTransferOnProceeding to 1 in the sip.cfg. Next, set progressinband=yes in sip.conf. Finally, in my dialplan, I had to remove calls to Answer() before calling dial. With all of this, the gateway is working brilliantly! Thanks, James _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
