[asterisk-bugs] [Asterisk 0018542]: [patch] 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:                     acknowledged
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:              2011-01-17 17:23 CST
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Summary:                    [patch] 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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 (0130578) vmikhelson (reporter) - 2011-01-17 17:23
 https://issues.asterisk.org/view.php?id=18542#c130578 
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May,

Tested incoming calls. Received inconsistent results.

1. Call placed at 10:41 lasted more than 30 seconds and DTMF worked fine;
2. Calls placed at 1:48 and 10:49 were dropped at 33 seconds, DTMF was not
working;
3. Call placed at 12:56 was not interrupted and DTMF worked fine.

No changes were done on either Avaya or Asterisk sides in between the test
calls. All outgoing calls were processed fine.

The log is still showing some SIP information, now in a different format.

Another issue related to the previous one -- CDR shows CLID with "SIP
suffixes" like in the log, e.g. Vlad Mikhelson.r" <352> or Vlad
Mikhelsonpg" <352>.

Thank you,
Vladimir 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-17 17:23 vmikhelson     Note Added: 0130578                          
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