|
I have downloaded and installed [EMAIL PROTECTED] 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:[EMAIL PROTECTED]/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/[EMAIL PROTECTED],30) exten => _9NXXNXXXXXX, 2, congestion() exten => _9NXXNXXXXXX, 102, busy() |
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