[asterisk-users] Mixing SIP/TDM in MeetMe

[Steve Davies]

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