[asterisk-users] Issue with SIP & QSIG phones in MeetMe conf room

[Richard Kenner]

My system is linked to a legacy PBX via Q-SIG and most of my tests so
far have been from that PBX.  I created a number of MeetMe conference rooms
and they work fine when called from the legacy PBX.  However, when there's
a MeetMe room with a legacy caller and a SIP phone, the SIP phone can 
hear the legacy caller.   But the legacy caller can't hear the SIP phone.
However, "meetme show <conf>" does show the SIP caller as "talking" when
they do.

Here's the current channels when the conference is up in that configuration:

asterisk*CLI> core show channels
Channel              Location             State   Application(Data)             
DAHDI/23-1           201 at Conferences:2    Up      MeetMe(201,cosT)              
DAHDI/pseudo-1338070 s at default:1          Rsrvd   (None)                        
SIP/150-b444d988     201 at Conferences:2    Up      MeetMe(201,cosT)     

What should I be looking at to debug this?