[asterisk-dev] [Code Review] 2938: res_rtp_asterisk: Address jittery DTMF events in RTP streams

[Mark Michelson]

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Ship it!



/branches/1.8/main/translate.c
<https://reviewboard.asterisk.org/r/2938/#comment19193>

    Use ast_debug here instead of ast_log. This is pretty low-level, so I'd recommend putting this at a higher number as well, such as 3 or 4.


- Mark Michelson


On Oct. 18, 2013, 8:04 p.m., jrose wrote:
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> https://reviewboard.asterisk.org/r/2938/
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> (Updated Oct. 18, 2013, 8:04 p.m.)
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> Review request for Asterisk Developers and Nitesh Bansal.
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> Bugs: ASTERISK-21170
>     https://issues.asterisk.org/jira/browse/ASTERISK-21170
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> Repository: Asterisk
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> Description
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> Posted this patch by Nitesh. I've made some minor style tweaks.
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> Diffs
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>   /branches/1.8/main/translate.c 401169 
>   /branches/1.8/res/res_rtp_asterisk.c 401169 
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> Diff: https://reviewboard.asterisk.org/r/2938/diff/
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> Testing
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> Added some additional logging to verify time values were sensible. Confirmed that the patch doesn't break DTMF in obvious ways (used chan_sip inband DTMF for testing).
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> Thanks,
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> jrose
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