[asterisk-bugs] [Asterisk 0019190]: When calling myself with a SIP TRUNK with ITSP provider, incoming call is considered diverted and channel converted to Local/

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=19190 
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Reported By:                albersag
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19190
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   trivial
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.17.3 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-04-27 10:35 CDT
Last Modified:              2011-04-27 12:39 CDT
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Summary:                    When calling myself with a SIP TRUNK with ITSP
provider, incoming call is considered diverted and channel converted to Local/
Description: 
Asterisk connected to SIP Provider, seems to bridge SIP Channels or Console
Channels as a call forwarding when they are incoming INVITES related to
outbound call made with same SIP provider account. ( I call myself).

When asterisk make this threatment, i could not use SIP functions s as
SIP_HEADER, because it is not a SIP CHANNEL as it has been considered a
call diverted.

I attach sip trace
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 (0134197) davidw (reporter) - 2011-04-27 12:39
 https://issues.asterisk.org/view.php?id=19190#c134197 
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Actually, in this case, the call hasn't been redirected by the SIP
provider, so they are trying to charge you.  Asterisk has been clever
enough to detect that there is a loop in the routing and has inferred the
redirect, to optimise out that loop.

I can actually see why you are trying to parse out the To header, now, but
that is a bug in the downstream system.  It has rewritten the user part of
the INVITE parameter, which it should not do if it is providing proper DID
service.

You shouldn't of course, be allowing such a loop to form in the first
place - you should recognize and handle your own number without passing it
to your service provider.

If you want me to go further, please take this to the Asterisk Support
forum. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-04-27 12:39 davidw         Note Added: 0134197                          
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