[asterisk-dev] [Code Review] Prevent DSP from incorrectly triggering during CallWaiting spill

[Alec Davis]

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https://reviewboard.asterisk.org/r/978/
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Review request for Asterisk Developers.


Summary
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The dsp incorrectly detects a DTMF start, when call waiting spill is injected to an FXS port.
The result is the SIP call has continuous DTMF RTP packets sent to it.

This can be corrected by the FXS port pressing a DTMF button.


This addresses bug 18129.
    https://issues.asterisk.org/view.php?id=18129


Diffs
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  trunk/channels/chan_dahdi.c 291724 

Diff: https://reviewboard.asterisk.org/r/978/diff


Testing
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intial setup SIP -> FXS port (TDM800P)
2nd call from console -> same FXS port as above.

Call waiting beep heard, and audio resumes both directions.


Thanks,

Alec