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

[Joshua Colp]

> 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.
> 
> yaron nahum wrote:
>     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?).
>     
>

In the set_caps function when the presence of session->channel is checked and the channel is locked you can retrieve the DTMF mode using ast_rtp_instance_dtmf_mode_get. If auto is enabled on the endpoint and the DTMF is not inband then see if a DSP is present. If one is present and faxdetect is not enabled on the endpoint you can safely remove the DSP.


- Joshua


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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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