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almarzuki2011 at hotma... Guest
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Posted: Mon Nov 09, 2015 3:02 pm Post subject: [asterisk-users] Asterisk unable to receive DTMF tone. |
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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
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ikka.tirta at gmail.com Guest
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Posted: Thu Nov 19, 2015 9:48 am Post subject: [asterisk-users] Asterisk unable to receive DTMF tone. |
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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@hotmail.com (almarzuki2011@hotmail.com)> wrote:
Quote: |
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
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