asterisk users - Mar 2005 - All Circuits are Busy Now

I have downloaded and installed Asterisk@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@sip.broadvoice.com:pppppppppp:xxxxxxxxxx@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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One thing you have to do is take the 9 out of the extension before you send it
on to broadvoice.  exten:1 on your dial cmd there.

Pulu



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Quoting Scott Wolfe <scottwolfe@orbus.net>:
> I have downloaded and installed Asterisk@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@sip.broadvoice.com:pppppppppp:xxxxxxxxxx@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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