[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]

The following issue has been CLOSED 
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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:                     closed
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:              
Resolution:                 reopened
Fixed in Version:           
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Date Submitted:             2011-04-27 10:35 CDT
Last Modified:              2011-04-27 13:02 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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Issue History 
Date Modified    Username       Field                    Change               
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2011-04-27 13:02 pabelanger     Status                   feedback => closed  
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