[asterisk-bugs] [Asterisk 0014514]: [patch] SIP Forking Feature

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Mar 10 10:13:32 CDT 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14514 
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Reported By:                Shivaprakash
Assigned To:                russell
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Project:                    Asterisk
Issue ID:                   14514
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.6.0.5 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 177657 
Request Review:              
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Date Submitted:             2009-02-20 00:32 CST
Last Modified:              2009-03-10 10:13 CDT
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Summary:                    [patch] SIP Forking Feature
Description: 
This feature enables a user to register from multiple locations and there
by allows forking of a SIP call.

Registration entries are maintained as a list in peers.
Registration refreshes of the same user are checked and updated
accordingly. New registrations of the same user from different location
will be added as a list in sip_peer structure.

An outgoing call is checked against the number of peers registered from
peer list and the invite is forked to all the location from where the user
is registered. Only the fisrst successful 2xx will establish a call, and
other forked calls will be terminated.
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---------------------------------------------------------------------- 
 (0101471) Shivaprakash (reporter) - 2009-03-10 10:13
 http://bugs.digium.com/view.php?id=14514#c101471 
---------------------------------------------------------------------- 
Thanks for the update.

My idea was from SIP point of view, as this is specific to SIP (dont know
whether other channels drivers support it) and also as the forking needs to
be done for a 'single' user registered from multiple locations, unlike
wherein PBX layer the call is forked to 'different' users, i thought it
would be better to handle at SIP channel driver

But these are valuable comments and let me see how best it can be
implemented at pbx layer 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-10 10:13 Shivaprakash   Note Added: 0101471                          
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