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

Rich Adamson radamson at routers.com
Sat Feb 18 07:23:48 MST 2006


> > > 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.

Might also do another update as that was removed by Olle about a week ago,
and then restored a few hours later.





More information about the asterisk-dev mailing list