[asterisk-dev] [Code Review] 2466: Pimp my SIP: Call forwarding & diversion headers

[Kevin Harwell]

> On April 25, 2013, 3:41 p.m., Joshua Colp wrote:
> > /team/group/pimp_my_sip/res/res_sip_diversion.c, line 205
> > <https://reviewboard.asterisk.org/r/2466/diff/2/?file=36393#file36393line205>
> >
> >     Having this return a value is silly as it only ever returns 1.

This function implements the ast_sip_session_supplement function pointer incoming_request which expects a return value.  Did you mean change it on the supplement struct and for all supplements?


> On April 25, 2013, 3:41 p.m., Joshua Colp wrote:
> > /team/group/pimp_my_sip/res/res_sip_diversion.c, lines 241-248
> > <https://reviewboard.asterisk.org/r/2466/diff/2/?file=36393#file36393line241>
> >
> >     Did you swipe this from pjsip source code?

Ooops yeah forgot about that.  Can I use it and add a comment or do I need to do something else?


> On April 25, 2013, 3:41 p.m., Joshua Colp wrote:
> > /team/group/pimp_my_sip/include/asterisk/res_sip.h, lines 331-332
> > <https://reviewboard.asterisk.org/r/2466/diff/2/?file=36390#file36390line331>
> >
> >     chan_sip has an option called "promiscredir" which follows the explicit SIP URI and does not just go to the user portion within the dialplan. Is this something we want to do as well here? (albeit maybe not the same way)

I can add this option...and a call to ast_channel_call_forward_build right?  Or is that what you meant by not the same way?


- Kevin


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On April 24, 2013, 8:09 p.m., Kevin Harwell wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2466/
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> 
> (Updated April 24, 2013, 8:09 p.m.)
> 
> 
> Review request for Asterisk Developers, Joshua Colp and Mark Michelson.
> 
> 
> Bugs: ASTERISK-21426
>     https://issues.asterisk.org/jira/browse/ASTERISK-21426
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> Adds call forwarding support (Josh's patch) to the new SIP work being done in Asterisk.  This also includes the ability to add a diversion header to an outgoing response/request when appropriate.  The diversion header feature can be turned off by setting the send_diversion=false (defaults to true) on an endpoint within the configuration file.
> 
> 
> Diffs
> -----
> 
>   /team/group/pimp_my_sip/channels/chan_gulp.c 386451 
>   /team/group/pimp_my_sip/include/asterisk/res_sip.h 386451 
>   /team/group/pimp_my_sip/res/res_sip.c 386451 
>   /team/group/pimp_my_sip/res/res_sip/sip_configuration.c 386451 
>   /team/group/pimp_my_sip/res/res_sip_diversion.c PRE-CREATION 
>   /team/group/pimp_my_sip/res/res_sip_session.c 386451 
> 
> Diff: https://reviewboard.asterisk.org/r/2466/diff/
> 
> 
> Testing
> -------
> 
> Wrote a few testsuite tests to make sure calls are being forwarded and the diversion header is added/propagated appropriately.
> 
> 
> Thanks,
> 
> Kevin Harwell
> 
>

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