0
votes

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

2

2 Answers

0
votes

In logger.conf add to console line

 console=>dtmf,verbose,debug

After that see debug output.

You may want also change dtmfmode param in your trunk config.

http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode

0
votes

If your peers are in the same network, set nat to "no", that may help.