[asterisk-bugs] [Asterisk 0010356]: chan_ooh323 don't work correctly
noreply at bugs.digium.com
noreply at bugs.digium.com
Thu Aug 9 23:41:35 CDT 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=10356
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Reported By: malufrj
Assigned To:
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Project: Asterisk
Issue ID: 10356
Category: Addons/chan_ooh323
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.4.8
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 08-01-2007 10:04 CDT
Last Modified: 08-09-2007 23:41 CDT
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Summary: chan_ooh323 don't work correctly
Description:
Hi people,
I wanna work with Asterisk 1.4, and need work with 2 channels, SIP and
H.323.
Because I have one Proxy SIP, OpenSER, and other Proxy H.323, GnuGK.
My tests with chan_h323 wasn't good and I compiled chan_ooh323 in
replace.
But until now, I still don´t get one perfect call. The RTP traffic don't
pass in the two directions or the call is broken.
When I call H.323 to SIP, my sip softphone ringing and my h.323 terminal
show the message "XXXX is ringing", but when I answer, the call in SIP stay
active while the call in H.323 is broken (hangup).
When I call SIP to H.323, the call is established but the RTP traffic just
work in H.323 to SIP direction while in the other direction become mute,
i.e., without RTP traffic.
I am sniffering the messages SIP and H.323 in the clients and in the PROXY
to help us.
My software version is: Asterisk 1.4.8 and Asterisk-addons 1.4.2
thanks in advanced,
Thiago Maluf.
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jeanneth - 08-09-07 23:41
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I have the same problem with ooh323 so i change to openh323 and it is
better, now is working!!!!:)
Issue History
Date Modified Username Field Change
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08-09-07 23:41 jeanneth Note Added: 0068698
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