Repost: Re: [asterisk-dev] How does RFC2833 get indicated to the SIP peer

[BJ Weschke]

On 2/17/06, Ed Greenberg <edg at greenberg.org> wrote:
> Can somebody who understands chan_sip.c please explain this to me? THanks.
>
> --On Thursday, February 16, 2006 6:20 AM -0800 Ed Greenberg
> <edg at greenberg.org> wrote:
>
> > Back in Asterisk 1.0.5, when we sent our SDP packet to the distant end,
> > we sent
> >       m=audio <port> RTP/AVP <codec> 101
> > where the 101 which indicated that we wanted to get RFC2833 DTMF from our
> > other end.
> >
> > Now it's missing, and my peer (level3) is sending me inband DTMF.
> >
> > It's not obvious to me from reading channels/chan_sip.c (in either the
> > old 1.0.5 or the current 1.2.4) how this 101 gets on the end of my Media
> > Description line or how else the peer is supposed to know that I need
> > rfc2833 DTMF.
> >
> > Can somebody please explain?

 Do you have dtmfmode=rfc2833 in sip.conf for this peer? If so, let's
get a sip debug and open a bug on bugs.digium.com.

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