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<DIV><FONT face=Arial size=2>I have downloaded and installed <A
href="mailto:Asterisk@home">Asterisk@home</A> 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
<A
href="http://voip-info.org/wiki-Asterisk+settings+Broadvoice">http://voip-info.org/wiki-Asterisk+settings+Broadvoice</A> and
<A
href="http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup">http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup</A> but
when ever I try and make a call from Xlite I get the all circuits are Busy now
recording. </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Do I need to create a Trunk or get rid of the one
that's there? Currently listed is the </FONT> </DIV>
<DIV><FONT face=Arial size=2>ZAP/g0 wich I think is for a hard line. Here is my
current sip.conf and extensions.conf</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks for any tips. </FONT></DIV>
<DIV><FONT face=Arial size=2> -Scott</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>========== sip.conf
==============</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>; Note: If your SIP devices are behind a NAT and
your Asterisk<BR>; server isn't, try adding "nat=1" to each peer
definition to<BR>; solve translation problems.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[general]</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>port =
5060 ; Port to bind
to (SIP is 5060)<BR>bindaddr = 0.0.0.0 ; Address to bind to
(all addresses on
machine)<BR>disallow=all<BR>allow=ulaw<BR>allow=alaw<BR>context =
from-sip-external ; Send unknown SIP callers to this context<BR>callerid =
Unknown</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>#include sip_nat.conf<BR>#include
sip_additional.conf</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>register => xxxxxxxxxx<A
href="mailto:xxxxxxxxxx@sip.broadvoice.com:pppppppppp:xxxxxxxxxx@sip.broadvoice.com/2197">@sip.broadvoice.com:pppppppppp:xxxxxxxxxx@sip.broadvoice.com/2197</A></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[sip.broadvoice.com]<BR>type=peer<BR>user=phone<BR>host=sip.broadvoice.com<BR>fromdomain=sip.broadvoice.com<BR>fromuser=xxxxxxxxxx<BR>secret=pppppppppp<BR>username=xxxxxxxxxx<BR>insecure=very<BR>context=from-broadvoice<BR>authname=xxxxxxxxxx<BR>dtmfmode=inband<BR>dtmf=inband<BR>authuser=xxxxxxxxxx<BR>;Disable
canreinvite if you are behind a
NAT<BR>canreinvite=no<BR>quality=yes<BR></FONT></DIV>
<DIV><FONT face=Arial size=2>=== Extensions.conf ===========</FONT></DIV>
<DIV><FONT face=Arial size=2>; I only addedd:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[VOIP-OUT]<BR>exten => _9NXXNXXXXXX, 1, dial(<A
href="mailto:SIP/${EXTEN}@sip.broadvoice.com,30">SIP/${EXTEN}@sip.broadvoice.com,30</A>)
<BR>exten => _9NXXNXXXXXX, 2, congestion() <BR>exten => _9NXXNXXXXXX, 102,
busy()</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> </DIV></FONT></BODY></HTML>