[asterisk-dev] chan_ooh323 issue
Vlasis Hatzistavrou Mailing Lists Account
asterisk at kinetixtele.com
Wed Aug 23 14:27:37 MST 2006
Hello,
I believe that this is related to a known bug of chan_ooh323 where Asterisk
does not respond/handle properly the CONNECT message it receives from the
far end.
I didn't have time to look into this in depth, so I don't know if changing
the settings of ooh323.conf will be of any help...
Best regards,
Vlasis Hatzistavrou.
-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Juan Carlos
Castro y Castro
Sent: Τετάρτη, 23 Αυγούστου 2006 11:03 μμ
To: asterisk-dev at lists.digium.com
Subject: [asterisk-dev] chan_ooh323 issue
Greetings. Although this message may not initially look like a -dev
worthy post, keep reading because I believe it is.
I have the following problem: I have an Asterisk+GnuGK installation, in
which the brand of H.323 ATAs we use (obscure OEM; I believe the maker
is Aristel) are unable to receive calls. The phone rings, when I pick it
up it's all mute and - that's the weird part - the originator keeps on
hearing ring tones as if nobody had picked up. When the receiving (mute)
end hangs up, the call is terminated. Calls in the other direction don't
have this problem.
It's not an obvious ATA bug, neither a GnuGK bug -- two ATAs can call
each other just fine. More: Asterisk can call an H.323 softphone
(SJPhone) fine too. The traffic from the latter looks mighty different
from that from a call to an ATA. Here's links to Ethereal analyses of
three cases:
Asterisk to SJPhone (OK):
http://img78.imageshack.us/img78/9452/asterisktosoftphoneka1.png
Asterisk to ATA (Rings but doesn't complete):
http://img78.imageshack.us/img78/8003/asterisktoatasq5.png
ATA to ATA (OK):
http://img168.imageshack.us/img168/1830/atatoouratajd6.png
The big difference between Asterisk-to-ATA and ATA-to-ATA is, Asterisk
(or, rather, chan_ooh323) never sends terminalCapabilitySet or
masterSlaveDetermination packets. And, of course, there's no RTP coming
from Asterisk.
All captures were made at the gatekeeper. It has two IP addresses (has
to because of Linux-HA). Switching to it having only one IP didn't help.
Yes, I have the .cap files that originated those.
I fear there will have to be some hacking in chan_ooh323 (hence me
posting here) in order for it to speak properly with the ATAs like other
ATAs do. If someone would like to test this, I can register my H.323 ATA
with the gatekeeper you're using, and keep on making and receiving calls
at your leisure. Plus chatting via MSN or the messenger of your choice,
or IRC.
If this wasn't -devvy enough, I'll post to -users instead. Should it
also go to bugs.digium.com?
Regards,
Juan
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