[asterisk-bugs] [Asterisk 0018020]: Realtime SIP Registration lost when performing a SIP RELOAD

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Sep 21 13:23:31 CDT 2010


The following issue has been CLOSED 
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https://issues.asterisk.org/view.php?id=18020 
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Reported By:                journo
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18020
Category:                   Addons/res_config_mysql
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.36 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             2010-09-20 19:27 CDT
Last Modified:              2010-09-21 13:23 CDT
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Summary:                    Realtime SIP Registration lost when performing a SIP
RELOAD
Description: 
I'm testing realtime via mysql for the sip peers.

However, whenever I edit the sip.conf file, for example to add a new
wholesale connection (register =>), I need to perform a SIP RELOAD.

Whenever I do the SIP RELOAD, it causes the realtime peers to lose their
registration.

I've looked through the existing bug reports but can only find similar
reports for 1.6, not the 1.4.36 which I'm using.

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---------------------------------------------------------------------- 
 (0127211) lmadsen (administrator) - 2010-09-21 13:23
 https://issues.asterisk.org/view.php?id=18020#c127211 
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This does not appear to be a bug, but rather a support issue. Please use
the asterisk-users mailing list for such issues.

The problem looks like your device has not re-registered after your 'sip
reload' which means it does not exist in memory, and thus causes Asterisk
to not know where to send the call. Your device needs to re-register after
a 'sip reload' in order for Asterisk to know where to send the call. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-09-21 13:23 lmadsen        Note Added: 0127211                          
2010-09-21 13:23 lmadsen        Status                   new => closed       
======================================================================




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