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

[Mark Michelson]

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/team/group/pimp_my_sip/res/res_sip_dtmf_info.c
<https://reviewboard.asterisk.org/r/2455/#comment15973>

    I believe that PJSIP should send a "500 Not handled by dialog usages" response if an in-dialog response is not handled by us, but have you double-checked that a response to the INFO is actually sent if we return -1 here?



/team/group/pimp_my_sip/res/res_sip_dtmf_info.c
<https://reviewboard.asterisk.org/r/2455/#comment15972>

    If an INFO arrives with no signal= line in the body, then we'll end up queuing a frame with 0-length DTMF. I have no idea what the result of this would be, but it's a good idea to make sure that the event is not '\0' before entering this if/else block.


- Mark Michelson


On April 17, 2013, 6:18 p.m., Jason Parker wrote:
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> (Updated April 17, 2013, 6:18 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,
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> Jason Parker
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>

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