[asterisk-users] Problem with SendDTMF

[Shahid H]

Hello,

I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.

I use software phone to test it... when I dialed 501, I cant hear anything
for about 10 seconds (this is because of SendDTMF)  and then I can hear
the operator saying to enter the numbers but SendDTMF already did it?!

Asterisk server are connected to voip.ms provider.

I have spent many hours trying to get to work, how to fix this issue?

See the configuration and debug log below:

extensions.conf
================
[test]
exten => 501,1,Set(CALLERID(num)=004471XXXXXXX)
exten => 501,n,Dial(SIP/+44797XXXXXX at voipms,30,M(sendnumber)t)
exten => 501,n,Hangup()

[macro-sendnumber]
exten => s,1,Wait(3)
exten => s,n,SendDTMF(www0w7w8w8wXwXwXwXwXwX)

sip.conf
==========
[general]
context=default
tcpbindaddr=0.0.0.0
dtmfmode = rfc2833
register => xxxxx:vxxxxx at london.voip.ms:5060

[test]
type=peer
secret=2xxx
host=dynamic
context=test

[voipms]
canreinvite=no
host=london.voip.ms
secret=xxxxxx
type=peer
username=135xxx ;your account
disallow=all
allow=gsm
; allow=g729 ; Uncomment if you support G729
fromuser=135xxx
insecure=invite
trustrpid=yes
sendrpid=yes
nat=yes
dtmfmode=rfc2833




debug:
=====
  == Using SIP RTP CoS mark 5
    -- Executing [501 at test:1] Set("SIP/test-00000026",
"CALLERID(num)=004471XXXXXX") in new stack
    -- Executing [501 at test:2] Dial("SIP/test-00000026",
"SIP/+4479XXXXXX at voipms,30,M(sendnumber)t") in new stack
  == Using SIP RTP CoS mark 5
    -- Called +44797XXXXXX at voipms
    -- SIP/voipms-00000027 is making progress passing it to
SIP/test-00000026
    -- SIP/voipms-00000027 answered SIP/test-00000026
    -- Executing [s at macro-sendnumber:1] Wait("SIP/voipms-00000027", "3") in
new stack
    -- Executing [s at macro-sendnumber:2] SendDTMF("SIP/voipms-00000027",
"www0w7w8wXwXwXwXw4wXwXwX") in new stack




Thanks!
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