[asterisk-dev] [Code Review] Generic call forward api: ast_call_forward()

[Russell Bryant]

> On 2009-06-02 10:19:00, Russell Bryant wrote:
> > /trunk/main/channel.c, line 3947
> > <http://reviewboard.digium.com/r/271/diff/2/?file=5506#file5506line3947>
> >
> >     What about replacing the existing forwarding implementations with usage of this one?

Let me rephrase this a bit.

As a first pass, I think the current patch is fine.  Also, I think we should leave the existing code for release branches.  In trunk though, I think it would be a good idea to come back with another patch and make all of the call forward handling use the new API.  It may require some additional modifications to accommodate the special needs of the existing uses, but it should be doable.


- Russell


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On 2009-06-01 17:10:23, David Vossel wrote:
> 
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> 
> (Updated 2009-06-01 17:10:23)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string.  After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one.  I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial().  App_dial and app_queue already contain call forward logic specific for their application and options.
> 
> 
> This addresses bug 13630.
>     https://issues.asterisk.org/view.php?id=13630
> 
> 
> Diffs
> -----
> 
>   /trunk/main/features.c 198632 
>   /trunk/main/channel.c 198632 
>   /trunk/include/asterisk/channel.h 198632 
> 
> Diff: http://reviewboard.digium.com/r/271/diff
> 
> 
> Testing
> -------
> 
> I set up Opensips to respond to a SIP INVITE with a different user.  
> 
> 1. Tested ast_call_forward in feature_request_and_dial() by directing an attended transfer to the Opensips extension.
> 2. Tested ast_call_forward in ast_request_and_dial() by using originate to create a call between a sip phone and the Opensips extension.
> 
> 
> Thanks,
> 
> David
> 
>