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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> 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!!!! >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
