[Asterisk-Users] sip <-> h323 audio problem

[Stewart Nelson]

> i have a audio problem between sip and h323.

> First my installation:

> Debian Sarge
> Asterisk 1.0.1
> Gnugk 2.0.8

> Asterisk register a prefix to gnugk.

> Communication from sip to sip and h323 to h323 is working.

> When i now call from the siphone (three tested) the h323 phone (also 
> three tested) the connection is coming up and everything seems to be ok 
> (no errors, no debug info).  But there is no audio in both directions. 
> Also when i call voicemail, i hear nothing one the h323 phone.

> I have tested different codecs.

> Has anybody a hint for me, where to continue my search for the problem?

> Greats,

> Andre Peitz

I assume, since you didn't mention it, that there are no NATs or
firewalls in the path.

I would think that H.323->voicemail would be easiest to debug.
Run Ethereal on the Asterisk machine.  Is * sending audio?

If not: did * send a Connect, and did the Open Logical Channels
happen correctly?

If yes: Is it sending to the correct IP, correct port, using
the right codec and correct payload size?  If any routers are
involved (including gnugk proxy), run Ethereal at the H.323
phone to be sure that the audio is really getting there,
and that no unwanted packet mangling is happening.

Can you configure an H.323 phone to call * directly (without
a GK)?  Also, try turning Fast Start on (or off).  Likewise with
H.245 tunneling.

Good luck,

Stewart