asterisk extensions.conf & sip.conf

Posted by Josh on Server Fault See other posts from Server Fault or by Josh
Published on 2009-10-15T18:47:14Z Indexed on 2012/04/08 17:33 UTC
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I'm trying to get my Dialplan to work. When I call, the only thing I get is a dial tone to enter extension "no Background(thanks-calling) is played". When extension 123 is dialed, busy signal is triggered and asterisk CLI get frozen. Any help will be appreciate it. Conf files below.

; PSTN on sip.conf
[pstn]
type=friend
host=dynamic
context=pstn
username=pstn
secret=password
nat=yes
canreinvite=no
dtmfmode=rfc2833
qualify=yes
insecure=port,invite
disallow=all
allow=ulaw


; PSTN on extensions.conf
[pstn] 
exten => s,1,Answer
exten => s,2,Wait,2
exten => s,4,DigitTimeout,5
exten => s,5,ResponseTimeout,10
exten => s,6,Background(thanks-calling)
exten => 0,1,Goto(incoming,123,1) ; (Member Services)

[incoming]
exten => 123,1,NoOP(${CALLERID}) ; show the caller ID info in the console
exten => 123,n,Ringing()
exten => 123,n,Answer()
exten => 123,n,Playback(silence/1)
exten => 123,n,Playback(connecting1)
exten => 123,n,Wait(3)
exten => 123,n,Dial(SIP/line1,60)
exten => 123,n,Congestion

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