[asterisk-bugs] [Asterisk 0012504]: Changing port number on SIP trunk is not reflected in Asterisk

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12504 
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Reported By:                voiptel
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12504
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.18 
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:             04-23-2008 09:42 CDT
Last Modified:              04-30-2008 11:21 CDT
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Summary:                    Changing port number on SIP trunk is not reflected
in Asterisk
Description: 
Changing the port number from 5060 to anything else (eg. 5061) and
reloading Asterisk (in the Advanced settings of the SIP trunk in Service
Providers) has no effect on the trunk in Asterisk, even after rebooting.
Even manually adding the trunk to the users.conf file with a different port
only results in it using port 5060 anyways (according to 'sip show
registry').

To further eliminate any other possibilities, I've changed all references
to port 5060 in all config files, including the bindport setting, but still
it remains locked to 5060.

This bug first appeared in a custom project based on an older version of
the Asterisk GUI, and was confirmed to exist in both version 0.9.6 and
1.0.2 of AsteriskNOW.
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---------------------------------------------------------------------- 
 oej - 04-30-08 11:21  
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Well, that was my very personal opinion. In an open source project with
multiple developers from multiple organizations involved, sometimes we have
different opinions on a module, and that's fine. We can't have an agreement
on everything in a complex toolkit like Asterisk. 

My opinion is that users.conf is a broken concept that doesn't scale
properly - there will have to be a lot of work to support all the features
in sip.conf, so it's to me just for very basic settings and very basic
configurations and something I don't want to spend my own time on. Sorry. 

Other developers might just love it and want to fix everything :-) 

Issue History 
Date Modified   Username       Field                    Change               
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04-30-08 11:21  oej            Note Added: 0086217                          
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