[asterisk-users] Audiocodes Mediant 1000, Polycom, and no ringback on transfer

[james.texter at cox.net]

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