asterisk users - Jan 2007 - Audiocodes Mediant 1000, Polycom, and no ringback on transfer

I just put in a Audiocodes Mediant 1000, which seems to be working well except
for one annoyance.  I am using Polycom 501's and 601',s and if I do a
supervised transfer of a PSTN call where I complete the transfer before the 3rd
party has answered, the PSTN party hears dead air until the call is answered or
goes to voicemail.  I'm not really sure where to start my troubleshooting. 
Any one have any experience with this type of setup?

Thanks,

James

On 1/15/07, james.texter@cox.net <james.texter@cox.net>
wrote:
>
> I just put in a Audiocodes Mediant 1000, which seems to be working well
> except for one annoyance.

I don't have any experience with an Audiocodes Meidant 1000, but I'll
try to
help you

> I am using Polycom 501's and 601',s

We have a lot of these

and if I do a supervised transfer of a PSTN call where I complete
the
> transfer before the 3rd party has answered,

I don't think you can do that. Here's why: on the Polycom's, the
Transfer
button doesn't reappear until the transferree picks up the phone. Unless
something changed in the firmware recently. But, if you're completing it
before the 3rd party answers, it's not an attended transfer.

 the PSTN party hears dead air until the call is answered or goes
to
> voicemail.

I would start by making sure the Music on Hold actually works, and that the
SIP phones are properly configured to use a MOH context that actually
exists. If those things check out, I would try using a blind transfer and
see what happens, try transferring when the 3rd party answers (VM or
whatever), and watch the console carefully with as much verbosity as
possible.

 I'm not really sure where to start my troubleshooting.  Any one have
any
> experience with this type of setup?

Hope this helps,
David

Thanks,
>
> James
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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