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

[Joshua Colp]

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/trunk/res/res_pjsip_sdp_rtp.c
<https://reviewboard.asterisk.org/r/4438/#comment25126>

    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.



/trunk/res/res_pjsip_session.c
<https://reviewboard.asterisk.org/r/4438/#comment25128>

    If telephone-event is negotiated the DSP should be dropped since it will not be used and will needlessly look at inband.


- Joshua Colp


On March 1, 2015, 1:34 p.m., yaron nahum wrote:
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4438/
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> (Updated March 1, 2015, 1:34 p.m.)
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> Review request for Asterisk Developers.
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> Repository: Asterisk
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> Description
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> add auto-dtmf mode for pjsip
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> Diffs
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>   /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 
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> Diff: https://reviewboard.asterisk.org/r/4438/diff/
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> Testing
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> Thanks,
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> yaron nahum
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>

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