[asterisk-dev] [Code Review] 4438: add auto-dtmf mode for pjsip

[yaron nahum]

> On March 3, 2015, 7:10 p.m., Joshua Colp wrote:
> > /trunk/res/res_pjsip_sdp_rtp.c, lines 175-205
> > <https://reviewboard.asterisk.org/r/4438/diff/1/?file=71567#file71567line175>
> >
> >     This should only happen when the auto option is used. It can also be changed to set inband at the beginning and switch to rfc2833 if telephone-event is present.

I will Fix it. It's easy.


> On March 3, 2015, 7:10 p.m., Joshua Colp wrote:
> > /trunk/res/res_pjsip_session.c, lines 1107-1109
> > <https://reviewboard.asterisk.org/r/4438/diff/1/?file=71568#file71568line1107>
> >
> >     If telephone-event is negotiated the DSP should be dropped since it will not be used and will needlessly look at inband.

The question is how do I get access to session_media instance? 
In order to do so I need a way to add session_media instance to ast_sip_session function arguments, or maybe there is a way to access it from one of the other arguments (maybe pjsip_inv_session?).


- yaron


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On March 3, 2015, 7:11 p.m., yaron nahum wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4438/
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> 
> (Updated March 3, 2015, 7:11 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-24706
>     https://issues.asterisk.org/jira/browse/ASTERISK-24706
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> add auto-dtmf mode for pjsip
> 
> 
> Diffs
> -----
> 
>   /trunk/res/res_pjsip_session.c 431537 
>   /trunk/res/res_pjsip_sdp_rtp.c 431537 
>   /trunk/res/res_pjsip/pjsip_configuration.c 431537 
>   /trunk/res/res_pjsip.c 431537 
>   /trunk/res/res_musiconhold.c 431537 
>   /trunk/include/asterisk/res_pjsip.h 431537 
>   /trunk/channels/chan_pjsip.c 431537 
> 
> Diff: https://reviewboard.asterisk.org/r/4438/diff/
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> yaron nahum
> 
>

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