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 <[email protected]> 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 >
-- _____________________________________________________________________ -- 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
