[Freeswitch-users] freeswitch, openzap, dahdi issues...

Tom Christensen pavera at gmail.com
Thu Mar 18 13:28:02 PDT 2010

In my quest to replace an unstable and deadlock prone asterisk solution I
was able to get calls working over my PRI last night using freeswitch,
native openzap, and dahdi 2.2.1.  However, in order to make it work, I had
to disable all echo cancellation in dahdi.  If any of the 4 echo
cancellation algo's were specified in /etc/dahdi/system.conf then there was
no audio at all on the calls.  Signaling worked, and calls could be
made/received, but no audio in either direction.  As soon as I disabled EC,
audio worked (although, there was a good amount of echo).

Also, I had intermittent (like 1 out of 2 or 1 out of 3) calls with very
poor voice quality (robotic sounding, distortions, dropping in and out).

This experience leads me to believe that possibly openzap isn't ready for
prime time?  Would I have been better served to include libpri in the
technology stack?

At this stage my "solution" looks like leaving asterisk where it is, and
using it as a media gateway, passing calls off to freeswitch via SIP or IAX
and using freeswitch as the app/registration/feature server.  I don't know
if that will necessarily be better, but I've had good luck with asterisk and
call quality tuning (the PRI here has always had pretty bad echo, but I've
been able to get asterisk EC working quite well on it), and it seems like
asterisk has most of its issues with stability in
transferring/parking/hold/voicemail/music on hold...

I'm running a current FS SVN from 3 days ago, dahdi 2.2.1, and openzap using
a digium te110p.  The PRI is an NI2 t1 if that matters...

Any ideas or recommendations of things to try would be greatly appreciated.
I'd really like to avoid the dual server situation if at all possible.

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