[asterisk-bugs] [Asterisk 0010313]: Chan SIP and ACL Source Based Routing

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Sep 14 05:00:52 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=10313 
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Reported By:                bfsworks
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   10313
Category:                   Channels/chan_sip/Registration
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.9  
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 69944 
Request Review:              
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Date Submitted:             2007-07-26 18:22 CDT
Last Modified:              2010-09-14 05:00 CDT
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Summary:                    Chan SIP and ACL Source Based Routing
Description: 
If asterisk is connected to two ISP connections statically. SIP
registrations fail and ACL reports cannot connect. Asterisk uses the
routing table and does not pass a source ip to the table and all routing
will default to the default route or gateway. If a SIP registration comes
in on ISP1 and the default route is for ISP2 then any and all responses go
out ISP2. 
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---------------------------------------------------------------------- 
 (0126919) lancey (reporter) - 2010-09-14 05:00
 https://issues.asterisk.org/view.php?id=10313#c126919 
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I'm having a machine which has a secondary IP address used for voice
traffic only. When I have chan_sip bind to it, all works fine. But now I
have to add a dedicated VLAN for direct communication with one of our
peers. In order for asterisk to use that, I have to tell it to bind to
0.0.0.0, but then some of the other peers connecting through the dedicated
voice IP address fail, because asterisk replies from the wrong IPs.
Is there any progress on the multiple bind issue, or is there anything
else I can do in this case (despite running another instance of asterisk
and connect both of them)? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-09-14 05:00 lancey         Note Added: 0126919                          
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