Hi, Asterisk unable to receive DTMF tone from sip client. Im using the (d) flag in dial application to perfume one digit exit during ringing state. But unfortunately doesn't work. Here is my sip configuration :- [100] type=friend username=100 host=dynamic nat=yes canreinvite=no allow=all secret=xxxxx context=sipphones relaxdtmf=yes dtmfmode=auto rfc2833compensate=yes [200] type=friend username=200 host=dynamic nat=yes canreinvite=no allow=all qualify=yes secret=xxxxx context=sipphones relaxdtmf=yes dtmfmode=auto rfc2833compensate=yes here is my extensions.conf:- exten => 100,1,Set(EXITCONTEXT=exitContext) exten => 100,n,Dial(SIP/100,30,dTt) exten => 200,1,Set(EXITCONTEXT=exitContext) exten => 200,n,Dial(SIP/200,30,dTt) [exitContext] exten =>9,1,Goto(sipphones,1,1) Regards -Hadi.Salem
Ikka Tirtawidjaja
2015-Nov-19 14:43 UTC
[asterisk-users] Asterisk unable to receive DTMF tone.
try setting your dtmfmode to INBAND or rfc2883, NOT auto... i have the same problem when using AUTO. but when i changed it to inband or rfc, the problem solved. On Tue, Nov 10, 2015 at 3:02 AM, hadi <almarzuki2011 at hotmail.com> wrote:> > Hi, > > Asterisk unable to receive DTMF tone from sip client. > Im using the (d) flag in dial application to perfume one digit exit during > ringing state. But unfortunately doesn't work. > > Here is my sip configuration :- > > [100] > type=friend > username=100 > host=dynamic > nat=yes > canreinvite=no > allow=all > secret=xxxxx > context=sipphones > relaxdtmf=yes > dtmfmode=auto > rfc2833compensate=yes > > [200] > type=friend > username=200 > host=dynamic > nat=yes > canreinvite=no > allow=all > qualify=yes > secret=xxxxx > context=sipphones > relaxdtmf=yes > dtmfmode=auto > rfc2833compensate=yes > > here is my extensions.conf:- > > exten => 100,1,Set(EXITCONTEXT=exitContext) > > exten => 100,n,Dial(SIP/100,30,dTt) > > exten => 200,1,Set(EXITCONTEXT=exitContext) > > exten => 200,n,Dial(SIP/200,30,dTt) > > [exitContext] > exten =>9,1,Goto(sipphones,1,1) > > > > Regards > > -Hadi.Salem > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151119/4fb9bb1a/attachment.html>