[asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy

[Jason Ma]

Buddies,
Thanks for  you response.
I have resolved the issue,it was not the DTMF mismatch between
Asterisk and Cisco proxy.
In fact,there is a Convedia media box behind Cisco proxy as conference
bridge,after checked the whole trace through the patch,I found that my
asterisk send video codec information in the SDP of invite,but the
Convedia media box doesn't support video,it got the request but did
not reject,just add a blank IP address 0.0.0.0 in SDP of 200 OK,so
there were two sections in 200 OK SDP,one is audio section with audio
IP address and port,the other is video section with a 0.0.0.0 IP
address.
When Asterisk got 200 OK,it was strange that it treated the video IP
0.0.0.0 as audio address,so the call was established but could not go
on,that was why I input anything it did not work,I think.
When I disabled the video support in Asterisk,it worked.

On 6/26/07, Ed Nuñez <enunez at netoneint.com> wrote:
> To configure the Cisco for RFC 2833 add the following line to the desired
> dial-peer
>
> dtmf-relay rtp-nte
>
> Hope this helps.
>
> Ed Nuñez
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric
> "ManxPower" Wieling
> Sent: Tuesday, June 26, 2007 11:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco
> SIP Proxy
>
> This is usually a Cisco issue.
>
> You need to set the Cisco to use RFC2833 DTMF.  Check the Cisco docs.
>
> tracinet wrote:
> > Jason,
> > I am at least having similar issues with rfc2833 DTMF:
> >
> > http://bugs.digium.com/view.php?id=10058
> >
> >
> > On 6/20/07, Jason Ma <realmj at gmail.com> wrote:
> >>
> >> Hi buddies,
> >> I encountered DTMF issue when I tried to place call from x-lite to a
> >> sip conference serice,here is the diagram.
> >> X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service
> >>
> >> The Call can be established,and I can hear from x-lite the prompt of
> >> the conference,but when I input any digits,nothing happened,the
> >> conference service did not recognize my input.At the same time,in the
> >> log of asterisk,I can find that asterisk recognized all the
> >> digits....I tried "rfc2833","inband","info" in the "dtmfmode"
> >> parameter,but did not work ,I'm not sure whether asterisk send the
> >> right dtmf to cisco proxy,how can I track that?
> >>
> >> I made another test,dialing from x-lite registered with Cisco proxy to
> >> voicemail service of Asterisk.
> >> x-lite---->Cisco SIP proxy---->Asterisk--->Voicemail service
> >>
> >> Both the call and dtmf worked fine,I can input my mailbox number and
> >> password and listen my  voicemail.both "rfc2933" and "inband" worked
> >> in this situation,but not "info".
> >>
> >> My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in
> >> the section of  xlite and the trunk to cisco proxy,just configure the
> >> dtmfmode in sip.conf.
> >>
> >> When I used "rfc2833",I can see the log in asterisk as :
> >>
> >> [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on
> >> SIP/9999-08269470
> >> [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on
> >> SIP/9999-08269470, duration 160 ms
> >> [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on
> >> SIP/9999-08269470
> >> [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on
> >> SIP/9999-08269470, duration 140 ms
> >>
> >> and when I used "inband",I can see :
> >>
> >> [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on
> >> SIP/9999-09d916c0, duration 0 ms
> >> [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on
> >> SIP/9999-09d916c0, duration 0 ms
> >>
> >> Is that right?Can I check what digits that asterisk sent out ?
> >>
> >> How can I track where is wrong with the dtmf?Did asterisk send dtmf to
> >> Cisco proxy correctly?
> >> I really have no idea about that.Please advise.Thank you very much!!!!
> >>
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> >
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