[asterisk-bugs] [Asterisk 0018542]: OOH323 Outgoing Calls Fail with Asterisk 1.8.1.1

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18542 
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Reported By:                vmikhelson
Assigned To:                may213
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Project:                    Asterisk
Issue ID:                   18542
Category:                   Addons/chan_ooh323
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.8.1.1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-12-28 11:26 CST
Last Modified:              2011-01-04 07:49 CST
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Summary:                    OOH323 Outgoing Calls Fail with Asterisk 1.8.1.1
Description: 
1. Upgraded to Asterisk 1.8.1.1 from Asterisk 1.6.2.15.

2. All outgoing calls started to fail.

3. "ooh323 show" was not recognized as a valid command. Auto-complete
worked though.

4 Rebooted the system.

CLI:

pbx*CLI> ooh323 show peers
Name             Accountcode      ip:port                  Formats
avaya            h3230101         172.17.135.2:1720        0x4 (ulaw)

FreePBX 2.8.0.4
AsteriskNOW 1.7

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 (0130122) may213 (manager) - 2011-01-04 07:49
 https://issues.asterisk.org/view.php?id=18542#c130122 
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Vladimir,

main change is progress sending by ooh323 channel, 1.6 addons version
doesn't provide progress signal, current version send progress on
ast_control_progress frame or by 1st rtp packet from asterisk core.
Btw, if you setup globally progressinband=yes in sip.conf then ringing
tone will generate by asterisk even you don't use r dialing option.
But 'r' option generate fictive ringing (alerting) signal which is first
real signalling packet from called side (call proceeding generate by ooh323
for any accepted by channel driver call). In this we have 1st ringing
(caused by 'r' option), progress (caused by 1st sound packet) and 2nd
ringing (caused by alerting from called endpoint). But possible it's
vice-versa, 1st ring go from endpoint and 2nd is fictive by 'r'.
In any case i think you can try to remove 'r' for testing for short time
and see on result.
I will see more accurately on app_dial code, i think root of this trouble
is here and we must generate right packet stream to avaya for solving
trouble.
Also you can replace 'r' by 'm' option, then asterisk will generate
background music sound instead of ringing tone. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-04 07:49 may213         Note Added: 0130122                          
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