[asterisk-users] newbbie question

[Pablo L. Arturi]

Hello, I am playing with my new * install, and there are a couple of things
that I don't understand, if someone could point me in the right direction it
will be appreciated.

I am trying to configure a voipstunt.com account to place outgoing calls,
and this is my config.

sip.conf:

[voipstunt]
type=friend ; (or "peer" if we don't need incoming calls, or if there is a
separate section with "type=user")
host=sip.voipstunt.com
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
username=voipstuntuser_replacement
fromuser=voipstuntuser_replacement
secret=hiddenpassword
qualify=1000 ; optional
canreinvite=no ; new SIP servers don't like reINVITEs
dtmfmode=inband ; only inband currently works, and not that well

extensions.conf:

[internal]
exten => 787793,1,Dial(SIP/john)
exten => 700099,1,Dial(SIP/maribel)
exten => 100000,1,Dial(SIP/dieguez)
exten => 100001,1,Dial(SIP/chparson)
exten => _NXXXXXXXXXXX,1,Dial(SIP/+{EXTEN}@voipstunt)


As stated in asteriskTFOT _NXXXXXXXXXXX will match 541152184829 which is the
phone number of my place, which I am trying to place a call.

I am asuming that the sign + in (SIP/+{EXTEN}@voipstunt) will be appended to
what I press in my softphone.

All I get when I call to 541152184829 is:

    -- Executing Dial("SIP/john-0819a010", "SIP/+{EXTEN}@voipstunt") in new
stack
    -- Called +{EXTEN}@voipstunt
Jul 22 20:15:45 NOTICE[1396]: chan_sip.c:1997 auto_congest: Auto-congesting
SIP/voipstunt-0819f520
    -- SIP/voipstunt-0819f520 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/john-0819a010' status is 'CONGESTION'


Any idea? suggestion?

Thanks in advance for any comment/help.

Pablo