[asterisk-bugs] [Asterisk 0011225]: URI direct dialing to target domain : call rejected if source extension exist in the destination dialplan.

noreply at bugs.digium.com noreply at bugs.digium.com
Mon Nov 12 16:49:15 CST 2007


The following issue has been RESOLVED. 
====================================================================== 
http://bugs.digium.com/view.php?id=11225 
====================================================================== 
Reported By:                FRANCE IP
Assigned To:                qwell
====================================================================== 
Project:                    Asterisk
Issue ID:                   11225
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     resolved
Asterisk Version:            1.2.18  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             11-12-2007 16:37 CST
Last Modified:              11-12-2007 16:49 CST
====================================================================== 
Summary:                    URI direct dialing to target domain : call rejected
if source extension exist in the destination dialplan.
Description: 

When calling from an SIP user Agent, if the calling party is using an
extension number used inside the called party dialplan, then asterisk wants
to authenticate the call instead of considering the call as anonymous.

This is not desirable, for example if using Xlite and the target domain
uri dialing option, the extension number sent inside the sip body is the
one from the regsitered on the calling party asterisk machine. If this
extension does exist on the destination asterisk, but with a different
password, then the call will get a 403 forbidden, instead of being
considered an anonymous call.


This is a serious problem when using such softphones in an asterisk world,
when we send out URI calls directly using the target domain IP ou SRV
record.


====================================================================== 

---------------------------------------------------------------------- 
 qwell - 11-12-07 16:49  
---------------------------------------------------------------------- 
1.2 is in security maintenance mode.  Please attempt to reproduce this
issue with 1.4, and reopen if you are able to. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-12-07 16:49  qwell          Status                   new => resolved     
11-12-07 16:49  qwell          Resolution               open => fixed       
11-12-07 16:49  qwell          Assigned To               => qwell           
11-12-07 16:49  qwell          Note Added: 0073551                          
======================================================================




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