[asterisk-dev] [Code Review] 2938: res_rtp_asterisk: Address jittery DTMF events in RTP streams
svnbot
reviewboard at asterisk.org
Wed Oct 23 12:30:34 CDT 2013
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2938/
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(Updated Oct. 23, 2013, 12:30 p.m.)
Status
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This change has been marked as submitted.
Review request for Asterisk Developers and Nitesh Bansal.
Changes
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Committed in revision 401619
Bugs: ASTERISK-21170
https://issues.asterisk.org/jira/browse/ASTERISK-21170
Repository: Asterisk
Description
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Posted this patch by Nitesh. I've made some minor style tweaks.
Diffs
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/branches/1.8/main/translate.c 401169
/branches/1.8/res/res_rtp_asterisk.c 401169
Diff: https://reviewboard.asterisk.org/r/2938/diff/
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).
Thanks,
jrose
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