The various iterations are as follows:
- If we place a call to another Asterisk server, running identical versions Asterisk, the h323 channel driver, and OpenH323 - we get no audio.
- If we call a Cisco PSTN gateway, no audio
- If we place a call to a Linux workstation running GnomeMeeting - the call works just fine! Audio is passed successfully, and everything is precisely as expected.
The dial strings in Asterisk for all of these incantations are identical (although we have of course experimented with many variations). As per a trace of the h.323 debugging, the codecs being negotiated are always G.711u. As an example, the dial strings we've been using are often formatted like this (some IPs and numbers have been changed to protect the innocent):
exten => 2999,1,Dial(H323/[EMAIL PROTECTED]/9195735483)
What is different when calling another Asterisk server, or a Cisco gatekeeper, vs calling a GnomeMeeting workstation? Why does one function and not the other? For reference purposes, the GnomeMeeting client can call the same remote Asterisk server, with exactly the same number (for example: [EMAIL PROTECTED]) and get a valid extension. The GnomeMeeting client can also call a SIP extension on either Asterisk server, so I'm certain that Asterisk is capable of gatewaying between SIP and H.323. I've heard the audio go through myself -- but only when originated by GnomeMeeting, as opposed to originated from Asterisk.
A colleague of mine discovered this (http://bugs.digium.com/bug_view_page.php?bug_id=0000562) over the weekend, which we think may be directly related. We're going to be implementing this patch to test it this morning. From the sounds of it, his problem may be directly or indirectly related to ours, and although it's not an identical problem, his fix may actually encompass both problems.
If anyone has any insights or thoughts as to why this doesn't work, or how to go about fixing it, it would be greatly appreciated.
-- Aaron S. Joyner System Administrator Intrex.net Internet Services (919) 573-5488 x102
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