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

[Sean Bright]

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/trunk/main/channel.c
<http://reviewboard.digium.com/r/271/#comment2004>

    Code was added to app_dial.c in all the branches to make sure that an empty FORWARD_CONTEXT was treated like a missing one.  So that change should go here as well:
    
        if (ast_strlen_zero(forward_context)) {
            forward_context = NULL;
        }


- Sean


On 2009-06-01 15:41:26, David Vossel wrote:
> 
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> (Updated 2009-06-01 15:41:26)
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> 
> Review request for Asterisk Developers.
> 
> 
> Summary
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> 
> 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
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> 
> Diffs
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>   /trunk/include/asterisk/channel.h 198632 
>   /trunk/main/channel.c 198632 
>   /trunk/main/features.c 198632 
> 
> Diff: http://reviewboard.digium.com/r/271/diff
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> 
> Testing
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> 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.
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> 
> Thanks,
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> David
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>