[asterisk-dev] [Code Review] Prevent DTMF incorrectly triggering during SAS+CAS (CallWaiting) signalling to an FXS port

Alec Davis sivad.a at paradise.net.nz
Thu Nov 4 06:49:21 CDT 2010


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https://reviewboard.asterisk.org/r/978/
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(Updated 2010-11-04 06:49:21.180701)


Review request for Asterisk Developers.


Changes
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Mute the conference as soon as the SAS+CAS signalling starts, and unmute when finished.

The symptom otherwise is that the other channels' dtmf detectors would hear the CAS acknowledgement (DTMF 'D') from the CPE, and would trigger a DTMF begin event, but with no corresponding DTMF end event, thus DTMF packets were sent continuously.

This also cleans up the beep that was heard by the other party/parties in the conference, when the FXS port is rung.


Summary (updated)
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The dsp incorrectly detects a DTMF 'D' start event, during the call waiting CAS acknowledgement from a CPE device that supports SAS+CAS connected 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 any DTMF button.


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


Diffs (updated)
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  trunk/channels/chan_dahdi.c 293886 

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




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