[Rtcomm] [Rtcomm] [Bug 5505] DTMF tones are not sent through SIP

Fri Oct 16 20:25:17 EEST 2009

https://bugs.maemo.org/show_bug.cgi?id=5505

maemo at subs.maneos.org changed:

           What    |Removed                     |Added
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                 CC|                            |maemo at subs.maneos.org

------- Comment #7 from maemo at subs.maneos.org  2009-10-16 20:25 GMT+3 -------
(In reply to comment #6)
> The fmtp attribute from Asterisk looks odd:
> 
> a=rtpmap:99 telephone-event/8000
> a=fmtp:99 0-16
> 
> It misses the events= parameter name

Should that be there at all?  It doesn't seem to agree with RFC2833 3.9.  BTW,
Diablo (working fine) doesn't send an fmtp line for telephone-event, maybe
that's relevant?

> and the range is 0-16 vs. our 0-15.

That should be fine, it just means asterisk also supports the flash event in
addition to 0-9, *, # and A-D.

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