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

[Asterisk Bug Tracker]

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 04:36 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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 (0101452) oej (manager) - 2009-03-10 04:36
 http://bugs.digium.com/view.php?id=14514#c101452 
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To allow the channel driver to fork is the wrong way to solve this issue.
We can't have forking on two layers. Asterisk is a multiprotocol PBX that
forks in the PBX layer. 

While I appreciate that you contribute to Asterisk and I understand that
this patch solves your issue, I don't approve of it. We can't have two PBX
layers. 

The correct way is to create another peer, and have the core PBX being
aware of two different sip peers and implement a variable - maybe reuse the
hint structure - that holds all the names of SIP peers to call, like
sip/olle#01, sip/olle#02 for a peer name sip/olle. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-10 04:36 oej            Note Added: 0101452                          
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