asterisk users - Oct 2009 - Mixing SIP/TDM in MeetMe

I sent a query about this before, but have some further information and am
hoping somebody has a suggestion as to what to try next to debug this.

I'm using an Asterisk box primarily for MeetMe conferencing.  There are
two sources: TDM via two Q.SIG T1's and SIP phones.  Conferencing works
fine between TDM channels.  But when a SIP phone calls the conference,
there's no voice path *to* the conference.  It can hear the conference
and its indicator changes appropriated from "not talking" to
"talking",
but nothing from it gets bridged into the conference (the entering and
leaving tones work fine).

Calls from the SIP phone to a TDM are fine.  I tried the experiment of
having the SIP phone dial across the T1 to the PBX which will then tandem
the call back to Asterisk.  When I do that, I have sound just fine.
"core show channel" look the same for both the Dahdi and SIP channels.

This is very frustrating.  Does anybody have any ideas?

2009/10/16 Richard Kenner <kenner at gnat.com>:
> I sent a query about this before, but have some further information and am
> hoping somebody has a suggestion as to what to try next to debug this.
>
> I'm using an Asterisk box primarily for MeetMe conferencing. ?There are
> two sources: TDM via two Q.SIG T1's and SIP phones. ?Conferencing works
> fine between TDM channels. ?But when a SIP phone calls the conference,
> there's no voice path *to* the conference. ?It can hear the conference
> and its indicator changes appropriated from "not talking" to
"talking",
> but nothing from it gets bridged into the conference (the entering and
> leaving tones work fine).
>
> Calls from the SIP phone to a TDM are fine. ?I tried the experiment of
> having the SIP phone dial across the T1 to the PBX which will then tandem
> the call back to Asterisk. ?When I do that, I have sound just fine.
> "core show channel" look the same for both the Dahdi and SIP
channels.
>
> This is very frustrating. ?Does anybody have any ideas?
>
As a complete guess, I would check what codecs you are using on the
SIP phone, and what transcoding paths are possible, particularly if
you are using licensed codecs.

Regards,
Steve