Im trying to configure voicemail, but asterisk doesnt respond to dtmf codes. I uses kphone with g711u codec (I've tryed the others one) and in sip.conf I configure dtmfmode=rfc2833 (I've tryied inband and info). Asterisk seems not to "see" the tones. Could somebody help me? Thanks
Hi all, I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no longer works with external phone systems. I have a Wildcard TDM400P with 4 FXO's? (it connects to analog lines). No changes were made to the config files. Here's my config: /etc/zaptel.conf fxsks=1-4 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes rxgain=2.0 txgain=2.0 callgroup=1 pickupgroup=1 musiconhold=default context=incoming group=1 signalling=fxs_ks echocancel=64 echocancelwhenbridged=yes relaxdtmf=yes channel => 1-3 [pete_desk] ;Pete's Desk phone (Polycom IP 300) type=friend username=pete_desk secret=pass context=longdistance callerid=Pete <601> host=dynamic mailbox=601 dtmfmode=inband disallow=all allow=ulaw allow=alaw Thanks, Pete
Everywhere it is RFC2833 including in SIP phone, Asterisk's sip.conf. DTMF work only from the phone that is hooked with asterisk box. Thanks,> -----Original Message----- > From: jnovack@stromberg-carlson.org > Sent: Wed, 24 Aug 2005 12:04:04 -0400 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] DTMF not working > > > > Innocent Evil wrote: > > >I am having same problem .. DTMF is not working from a SIP phone while > >sending to Asterisk cmd VoiceMailMain. > > > > > > > Have you set DTMF to out of band RFC2833? > > In band won't work. At least in my version of HEAD > > John Novack > > >Would you please explain this line > >"!941+1336/100,!0/100", /* 0 */ > > > >what value is what and how it affect on DTMF tone generation. > > > >Thanks, > > > > > > > > > > > >>I had a similar problem that seems to be caused by the DTMF tone > lengths > >>being to short. Try this: > >> > >>Asterisk generates DTMF tones in do_senddigit() in the file channel.c. > >>The tones are defined in a const char array called dtmf_tones[]. Each > >>DTMF tone is a string that looks something like: > >> > >>"!941+1336/100,!0/100", /* 0 */ > >> > >>The part that reads !941+1336/100 is the part that you want. Change > the > >>"100" to something bigger and recompile. You will have to do that for > >>every tone. I'm using 400 right now, and it seems to be working. > >> > >>I hope that helps. > >> > >>Rob > >> > >>Peter Osborne wrote: > >> > >> > >> > >>>Hi all, > >>> > >>>I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no > >>> > >>> > >>longer > >> > >> > >>>works with external phone systems. I have a Wildcard TDM400P with 4 > >>> > >>> > >>FXO's? > >> > >> > >>>(it connects to analog lines). No changes were made to the config > files. > >>> > >>>Here's my config: > >>> > >>>/etc/zaptel.conf > >>>fxsks=1-4 > >>>loadzone = us > >>>defaultzone=us > >>> > >>>/etc/asterisk/zapata.conf > >>>[channels] > >>>usecallerid=yes > >>>hidecallerid=no > >>>callwaiting=yes > >>>usecallingpres=yes > >>>threewaycalling=yes > >>>transfer=yes > >>>cancallforward=yes > >>>callreturn=yes > >>>echocancel=yes > >>>echotraining=yes > >>>rxgain=2.0 > >>>txgain=2.0 > >>>callgroup=1 > >>>pickupgroup=1 > >>>musiconhold=default > >>>context=incoming > >>>group=1 > >>>signalling=fxs_ks > >>>echocancel=64 > >>>echocancelwhenbridged=yes > >>>relaxdtmf=yes > >>>channel => 1-3 > >>> > >>>[pete_desk] > >>>;Pete's Desk phone (Polycom IP 300) > >>>type=friend > >>>username=pete_desk > >>>secret=pass > >>>context=longdistance > >>>callerid=Pete <601> > >>>host=dynamic > >>>mailbox=601 > >>>dtmfmode=inband > >>>disallow=all > >>>allow=ulaw > >>>allow=alaw > >>> > >>>Thanks, > >>>Pete > >>>_______________________________________________ > >>>Asterisk-Users mailing list > >>>Asterisk-Users@lists.digium.com > >>>http://lists.digium.com/mailman/listinfo/asterisk-users > >>>To UNSUBSCRIBE or update options visit: > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >>> > >>> > >>> > >>-- > >>Robert Tarte > >>Pacific CodeWorks > >>P.O. Box 29050 > >>San Francisco, CA 94129 > >> > >>(p) 831-426-7582 > >>(f) 831-426-7584 > >> > >>_______________________________________________ > >>Asterisk-Users mailing list > >>Asterisk-Users@lists.digium.com > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >>To UNSUBSCRIBE or update options visit: > >> >http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________> >> > >> > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Rob, I am using RFC2833 everywhere including SIP phone, asterisk's sip.conf Do you think, to raise the value from 100 to 400, would solve my issue? Thanks,> -----Original Message----- > From: rtarte@pacificcodeworks.com > Sent: Wed, 24 Aug 2005 08:46:43 -0700 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] DTMF not working > > Hi Mr. Evil, > > I'm not sure if the problem that I am describing relates to the problem > that you are having. It seems that when you press a key on a SIP phone > that is set for inband DTMF, asterisk absorbs the tones until you > release the key. This way if you are using DTMF to do things like > transfer calls, the user won't get tone blasts in their ear until > asterisk has had a chance to interpret the tones. After asterisk has > figured out what to do with the tone, it generates and transmits it's > own tones in the routine do_senddigit() (assuming that the DTMF tone > should be passed on). The duration of the DTMF tones that asterisk > generates is fixed and independent of how long you pressed the key on > your phone. > > In the line "!941+1336/100,!0/100", the 941 is one tone of the DTMF > (dual tone multi-frequency), and 1336 is the other tone. The 100 is the > duration of those tones. The tones are in Hz. I'm not sure what units > the duration is in, but I bumped mine from 100 to 400 and that seems to > do the trick. The part of the string that reads "!0/100" just shuts the > tone generator off. > > Rob > > Innocent Evil wrote: > > >I am having same problem .. DTMF is not working from a SIP phone while > >sending to Asterisk cmd VoiceMailMain. > > > >Would you please explain this line > >"!941+1336/100,!0/100", /* 0 */ > > > >what value is what and how it affect on DTMF tone generation. > > > >Thanks, > > > > > > > >>I had a similar problem that seems to be caused by the DTMF tone > lengths > >>being to short. Try this: > >> > >>Asterisk generates DTMF tones in do_senddigit() in the file channel.c. > >>The tones are defined in a const char array called dtmf_tones[]. Each > >>DTMF tone is a string that looks something like: > >> > >>"!941+1336/100,!0/100", /* 0 */ > >> > >>The part that reads !941+1336/100 is the part that you want. Change > the > >>"100" to something bigger and recompile. You will have to do that for > >>every tone. I'm using 400 right now, and it seems to be working. > >> > >>I hope that helps. > >> > >>Rob > >> > >>Peter Osborne wrote: > >> > >> > >> > >>>Hi all, > >>> > >>>I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no > >>> > >>> > >>longer > >> > >> > >>>works with external phone systems. I have a Wildcard TDM400P with 4 > >>> > >>> > >>FXO's? > >> > >> > >>>(it connects to analog lines). No changes were made to the config > files. > >>> > >>>Here's my config: > >>> > >>>/etc/zaptel.conf > >>>fxsks=1-4 > >>>loadzone = us > >>>defaultzone=us > >>> > >>>/etc/asterisk/zapata.conf > >>>[channels] > >>>usecallerid=yes > >>>hidecallerid=no > >>>callwaiting=yes > >>>usecallingpres=yes > >>>threewaycalling=yes > >>>transfer=yes > >>>cancallforward=yes > >>>callreturn=yes > >>>echocancel=yes > >>>echotraining=yes > >>>rxgain=2.0 > >>>txgain=2.0 > >>>callgroup=1 > >>>pickupgroup=1 > >>>musiconhold=default > >>>context=incoming > >>>group=1 > >>>signalling=fxs_ks > >>>echocancel=64 > >>>echocancelwhenbridged=yes > >>>relaxdtmf=yes > >>>channel => 1-3 > >>> > >>>[pete_desk] > >>>;Pete's Desk phone (Polycom IP 300) > >>>type=friend > >>>username=pete_desk > >>>secret=pass > >>>context=longdistance > >>>callerid=Pete <601> > >>>host=dynamic > >>>mailbox=601 > >>>dtmfmode=inband > >>>disallow=all > >>>allow=ulaw > >>>allow=alaw > >>> > >>>Thanks, > >>>Pete > >>> > >>> > > -- > Robert Tarte > Pacific CodeWorks > P.O. Box 29050 > San Francisco, CA 94129 > > (p) 831-426-7582 > (f) 831-426-7584 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users