[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
Fri Nov 5 06:23:11 CDT 2010
> On 2010-11-04 10:31:28, rmudgett wrote:
> > Looks ok to me. Just have a couple minor commenting items below.
Further testing revealed that muting the conference stopped the FXS port detecting the CPE's CAS response DTMF 'D', and thus doesn't send the CallWaiting CallerID.
Need to find another way.
- Alec
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On 2010-11-04 06:49:21, Alec Davis wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/978/
> -----------------------------------------------------------
>
> (Updated 2010-11-04 06:49:21)
>
>
> Review request for Asterisk Developers.
>
>
> Summary
> -------
>
> 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
> -----
>
> trunk/channels/chan_dahdi.c 293886
>
> Diff: https://reviewboard.asterisk.org/r/978/diff
>
>
> Testing
> -------
>
> 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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