[Asterisk-Dev] SIP / H.323 Gateway with Asterisk
Aaron S. Joyner
asjoyner at intrex.net
Mon Sep 6 07:15:11 MST 2004
We are using SIP phones to make calls through Asterisk, out an H.323
connection. Depending on the end point, some times these calls do not
pass audio. The call setup appears normal, everything about the
negotiation appears to be correct, but no audio is passed out of
asterisk, in either direction. From a packet dump we can see that audio
is coming in from both sides, but no audio is being sent out from Asterisk.
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/XXXXXXXXXXX at 111.222.333.444/9195735483)
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: 600 at hostname.of.machine) 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
More information about the asterisk-dev
mailing list