[asterisk-dev] [Code Review] 2455: Add support for DTMF via SIP INFO

[Joshua Colp]

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/team/group/pimp_my_sip/channels/chan_gulp.c
<https://reviewboard.asterisk.org/r/2455/#comment15968>

    As mentioned below - when you allocate this make sure it gets freed here appropriately.



/team/group/pimp_my_sip/channels/chan_gulp.c
<https://reviewboard.asterisk.org/r/2455/#comment15966>

    tdata memory leak



/team/group/pimp_my_sip/channels/chan_gulp.c
<https://reviewboard.asterisk.org/r/2455/#comment15967>

    No can do! Since you made this async now dtmf_data has to be allocated, and you also need to ensure it gets freed if pushing it fails.



/team/group/pimp_my_sip/channels/chan_gulp.c
<https://reviewboard.asterisk.org/r/2455/#comment15969>

    Remember to bump the ref for session up.


- Joshua Colp


On April 16, 2013, 8:26 p.m., Jason Parker wrote:
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2455/
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> (Updated April 16, 2013, 8:26 p.m.)
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> 
> Review request for Asterisk Developers.
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> Bugs: ASTERISK-21261
>     https://issues.asterisk.org/jira/browse/ASTERISK-21261
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> Repository: Asterisk
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> Description
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> Sending is done inside of chan_gulp, if the dtmf= setting in res_sip.conf is set to info.  Receiving is done as a separate module, since that made the most sense, to me.
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> Diffs
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>   /team/group/pimp_my_sip/channels/chan_gulp.c 385911 
>   /team/group/pimp_my_sip/res/res_sip_dtmf_info.c PRE-CREATION 
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> Diff: https://reviewboard.asterisk.org/r/2455/diff/
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> Testing
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> DTMF gets parsed properly.  Outgoing packet looks okay.
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> 
> Thanks,
> 
> Jason Parker
> 
>

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