[Asterisk-Dev] Bug 14 in bug tracker (fwd)

[Mark Spencer]

It's a little of both probably.  Asterisk's handling of DTMF in RTP isn't
proper either -- it is made around an attempt to be robust in handling
good implementations, but that makes it sensitive to odd implementations.

Mark

On Fri, 25 Jul 2003, Peter Grace wrote:

> I've done a workaround in rtp.c, but I know I'd prefer for the
> solution to be concrete.  What exactly is wrong, so we can give the
> xten people a good idea of what needs to be done?  From your message I
> gather that the duration field is not being properly populated?
>
> On Fri, 25 Jul 2003, Chris Heiser wrote:
>
> > This actually appears to be a bug with the X-Lite Sip Phone.
> > I've done some packet captures, and the X-Lite violates RFC2833.
> >
> > For the Payload of the RTP packet for DTMF, the RFC states:
> >
> > "   duration: Duration of this digit, in timestamp units. Thus, the
> >            event began at the instant identified by the RTP timestamp
> >            and has so far lasted as long as indicated by this parameter.
> >            The event may or may not have ended."
> >
> > X-Lite never increments the duration field of the RTP packets for DTMF
> > digits.  I'm sure this is confusing Asterisk.  I'd say e-mail the X-Lite
> > folks and tell them they have bugs.
> >
> > --Chris Heiser
> >
> > > -----Original Message-----
> > > From: asterisk-dev-admin at lists.digium.com [mailto:asterisk-dev-
> > > admin at lists.digium.com] On Behalf Of Chris Heiser
> > > Sent: Thursday, July 24, 2003 6:47 PM
> > > To: asterisk-dev at lists.digium.com
> > > Subject: Re: [Asterisk-Dev] Bug 14 in bug tracker (fwd)
> > >
> > > Peter,
> > >
> > > Can you send a tcpdump of the session between asterisk and the X-lite
> > > phone?
> > >
> > > Something like: tcpdump -s 0 -w xlite-phone-dump host [x-lite ip address]
> > > Make sure you run it while you're not logged in or otherwise connected to
> > > Asterisk (so the tcpdump only has the packets between the X-lite phone and
> > > Asterisk)
> > >
> > > I'd say just e-mail it to me direct as the file will probably be a little
> > > big to go out on the mailing list.
> > > I've had to debug other DTMF issues with other devices.  (I've come across
> > > a
> > > few sip UA's that ignore the SDP info that Asterisk sends out).  Maybe I
> > > can
> > > be of some help.
> > >
> > > --Chris Heiser
> > >
> > > ----- Original Message -----
> > > From: "Peter Grace" <pgrace at fierymoon.com>
> > > To: <asterisk-dev at lists.digium.com>
> > > Sent: Thursday, July 24, 2003 3:54 PM
> > > Subject: [Asterisk-Dev] Bug 14 in bug tracker (fwd)
> > >
> > >
> > > > Oops.  Helps if you have the list address right.. See below!
> > > >
> > > > ---------- Forwarded message ----------
> > > > Date: Thu, 24 Jul 2003 15:04:44 -0400 (EDT)
> > > > From: Peter Grace <pgrace at fierymoon.com>
> > > > To: asterisk-dev at digium.com
> > > > Subject: Bug 14 in bug tracker
> > > >
> > > > Guys,
> > > >
> > > > I just fired in a bug report, but I'll give the rundown:
> > > >
> > > > http://bugs.digium.com/bug_view_page.php?bug_id=0000014
> > > >
> > > >
> > > > X-Lite's DTMF problems with voicemail may not be X-lite's fault...  I
> > > was
> > > > tracking DTMF between rtp.c and channel.c and it looks like the system
> > > is
> > > > ignoring the first couple digits.  Check the bug report for full debug
> > > > messages...
> > > >
> > > > Thanks,
> > > > Pete
> > > >
> > > > _______________________________________________
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> > > > Asterisk-Dev at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-dev
> > > >
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> >
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