[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 12:02 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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 (0134194) davidw (reporter) - 2011-04-27 12:02
 https://issues.asterisk.org/view.php?id=19190#c134194 
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It is behaving as intended.  After the redirect, there is no longer any SIP
protocol running on the outbound side.  Your SIP provider is actually being
nice by not charging you for the call.

The incoming call is not running on a local channel.  A local channel has
been substituted for the SIP channel on which the original outbound call
was attempted, which is on the other side of the Dial application from the
incoming channel.

A local channel has to be used because Asterisk wants to interpret the
redirected address in the name-space (context) associated with incoming
calls from that peer, so it has to run dialplan to translate that number.

The final result may never end up on a channel, e.g. it might end up on a
voice announcement, so you can't say that there is any physical channel
technology associated with it.

If it does end up on a physical channel, that may not be SIP.  When it
does and that channel is answered, the Local channel should be optimised
out.

There does seem to be a problem with your SIP provider, if they provide
DID for incoming calls, in that they should have redirected to the DID, but
you should not be trying to disect SIP headers; you should be looking at
the target ${EXTEN}.  In any case, the redirected address does not appear
in a To header, it will be in a Contact header, but that SIP session will
be completely dead by the time that the new target extension gets run.  All
this paragraph is really about support issues. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-04-27 12:02 davidw         Note Added: 0134194                          
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