asterisk extensions.conf & sip.conf

Question

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

Answer 1

You seem to be using some ATA for PSTN calls, right? Maybe it has second-stage-dialing set up, so the call is not reaching Asterisk and the dialtone is generated by the ATA.

Set your asterisk to be more verbose (set verbose 3), and then dialin. What do you get on console?