[Asterisk-Users] All Circuits are Busy Now

[Scott Wolfe]

I have downloaded and installed Asterisk at home and I have installed X-Lite on my Windows machine and I am able to connect it to the Asterisk server. I went ahead an created an account on Broadvoice today and followed the directions on http://voip-info.org/wiki-Asterisk+settings+Broadvoice and http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but when ever I try and make a call from Xlite I get the all circuits are Busy now recording. 

Do I need to create a Trunk or get rid of the one that's there? Currently listed is  the  
ZAP/g0 wich I think is for a hard line. Here is my current sip.conf and extensions.conf

Thanks for any tips. 
  -Scott



========== sip.conf  ==============

; Note: If your SIP devices are behind a NAT and your Asterisk
;  server isn't, try adding "nat=1" to each peer definition to
;  solve translation problems.

[general]

port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown

#include sip_nat.conf
#include sip_additional.conf

register => xxxxxxxxxx at sip.broadvoice.com:pppppppppp:xxxxxxxxxx at sip.broadvoice.com/2197

[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=xxxxxxxxxx
secret=pppppppppp
username=xxxxxxxxxx
insecure=very
context=from-broadvoice
authname=xxxxxxxxxx
dtmfmode=inband
dtmf=inband
authuser=xxxxxxxxxx
;Disable canreinvite if you are behind a NAT
canreinvite=no
quality=yes

=== Extensions.conf ===========
; I only addedd:

[VOIP-OUT]
exten => _9NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) 
exten => _9NXXNXXXXXX, 2, congestion() 
exten => _9NXXNXXXXXX, 102, busy()

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