[asterisk-bugs] [Asterisk 0015106]: Routing Extensions between 2 asterisk servers in 2 directions fails with "482 Loop Detected"

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15106 
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Reported By:                timeshell
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15106
Category:                   General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0.9 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-05-14 09:35 CDT
Last Modified:              2009-05-15 15:34 CDT
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Summary:                    Routing Extensions between 2 asterisk servers in 2
directions fails with "482 Loop Detected"
Description: 
A* has a user created by asterisk-gui2.
B* has a trunkA created to connect to user on A*.
Calling rule for trunkA on B* allows exts connected to B* to call exts on
A*.

Now, I want exts connected to A* to be able to call exts on B*.

Scenario 1
Create SIP trunk user on B* with asterisk-gui2.  Connect trunk from A* to
user on B* and appropriate outgoing rule.
Result:  calls from one or both sides will get error in CLI reporting "482
Loop Detected"

Scenario 2
Create SIP trunk using asterisk-gui2 on A* but leave hostname, username
and secret blank.  Change registersip to = no.
Manually edit extensions.conf and change the global for the new SIP trunk
to be something like "trunk_1=SIP/A*Ext" where A*Ext is the trunk extension
for B* to connect a trunk to A*.
Create outgoing rule on A* to route B* exts to B*.
Create incoming rule on B* to accept incoming calls on trunk to A* and set
it to route to ${EXTEN}.
Result:  Calls may or may not work in one direction and the other
direction will not work reporting "482 Loop Detected"


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 (0104846) mmichelson (administrator) - 2009-05-15 15:34
 https://issues.asterisk.org/view.php?id=15106#c104846 
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The sip debug you attached shows a legitimate loop. The telltale sign is
this here:

<--- SIP read from UDP://127.0.0.1:5060 --->
INVITE sip:7000 at 206.248.146.138 SIP/2.0

Right above this, you sent an INVITE out, and then immediately you read an
INVITE on the loopback address. This means that you are routing the INVITE
to yourself and thus have a loop. This means that you either configured
your routes incorrectly, or the Asterisk-GUI has a bug in its configuration
setup.

Regarding other scenarios, I haven't looked too deeply into it, but I have
committed a patch just yesterday to solve issue
https://issues.asterisk.org/view.php?id=12215, which may be
related. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-05-15 15:34 mmichelson     Note Added: 0104846                          
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