[asterisk-bugs] [Asterisk 0011261]: [patch] chan_h323 with H323Plus for TRUNK (SVN rev. 89183)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11261 
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Reported By:                vhatz
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11261
Category:                   Channels/chan_h323
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     new
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 89183 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2007-11-15 11:46 CST
Last Modified:              2008-12-04 14:59 CST
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Summary:                    [patch] chan_h323 with H323Plus for TRUNK (SVN rev.
89183)
Description: 
This is a modification to the trunk version of SVN rev. 89183 chan_h32.
This modification enables chan_h323 to compile and use the latest H323Plus
library instead of the OpenH323 library.
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Relationships       ID      Summary
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duplicate of        0013888 Cannot compile SVN with patch from bug ...
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 (0095794) cclemama (reporter) - 2008-12-04 14:59
 http://bugs.digium.com/view.php?id=11261#c95794 
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I needed to connect to our Avaya switch using h323 and found this post. I
am running on FC10 using the most current version of asterisk
(asterisk-1.6.0.1) and the latest versions of h323plus code
(h323plus-v1_20_2 & ptlib-v2_0_1-4). Based upon the notes, I thought that
the 1.6 code would compile without any issues. It would not. So I manually
applied the pvhatz patches to the 1.6 code and got everything to compile. I
was very pleased with how easy it was to get the h323 connection working
with our Avaya switch. However, after more testing I think I found an
issue. 

When using a call file (i.e. autodial), control is passed on to the
context/extension/ before the outgoing h323 call is answered. This behavior
does not happen when I outbound dial a Sip, IAX2, or Zap channel. 

I am new to the asterisk code and voip protocols, but I am willing to
learn. I also have a system that we can use to test with. I have uploaded
my changed source code, config files, and logs. Can one of you guys help
direct me? 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-04 14:59 cclemama       Note Added: 0095794                          
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