<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2800.1106" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>Thanks for the encouraging advice. I actually spent
many hours searching for and reading through documentation about this (on the
wiki and in the handbook) and I couldn't figure out how Asterisk was supposed to
work as an SIP server.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Since I posted my original message I've made a lot
more progress (and spent considerably more than 15 minutes) but I still have not
managed to get it to work. </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have specified an SIP extension (many, actually)
in the sip.conf file but I cannot get DIAX to register with Asterisk. I've tried
changing just about every variable I can while troubleshooting. One thing that
is kind of suspect is what comes up after I have it re-read the config
files:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>------</FONT></DIV>
<DIV><FONT face=Arial size=2>Messages-Waiting: no</FONT></DIV>
<DIV><FONT face=Arial size=2>Voice-Message: 0/0</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> to 192.168.9.102:5060</FONT></DIV>
<DIV><FONT face=Arial size=2>Retransmitting #5 (no NAT):</FONT></DIV>
<DIV><FONT face=Arial size=2>NOTIFY sip:200@192.168.9.102 SIP/2.0</FONT></DIV>
<DIV><FONT face=Arial size=2>Via: SIP/2.0/UDP
192.168.9.101:5060;branch=z9hG4bK7cc5dc1e</FONT></DIV>
<DIV><FONT face=Arial size=2>From: "Unknown"
<sip:Unknown@192.168.9.101>;tag=as63d4a421</FONT></DIV>
<DIV><FONT face=Arial size=2>To: <sip:200@192.168.9.102></FONT></DIV>
<DIV><FONT face=Arial size=2>Contact:
<sip:Unknown@192.168.9.101></FONT></DIV>
<DIV><FONT face=Arial size=2>Call-ID: <A
href="mailto:2c9110f8126bc7c12ea475460afb633c@192.168.9.101">2c9110f8126bc7c12ea475460afb633c@192.168.9.101</A></FONT></DIV>
<DIV><FONT face=Arial size=2>CSeq: 102 NOTIFY</FONT></DIV>
<DIV><FONT face=Arial size=2>User-Agent: Asterisk PBX</FONT></DIV>
<DIV><FONT face=Arial size=2>Event: message=summary</FONT></DIV>
<DIV><FONT face=Arial size=2>Content-Type:
application/simple-message-summary</FONT></DIV>
<DIV><FONT face=Arial size=2>Content-Length: 42</FONT></DIV>
<DIV><FONT face=Arial size=2>------</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>192.168.9.101 is the Asterisk server and
192.168.9.102 is the machine I've been trying to get DIAX registered on. In the
past, I've specified the .102 address in the SIP config file for an extension
but at this point I can't think of anywhere where that IP address is specified
so this is a big mystery to me. Can anyone make sense of it?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have the following users in my sip_additionals
file (as generated by AMP):</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[200]<BR>username=200<BR>type=friend<BR>secret=test<BR>qualify=no<BR>port=5060<BR>nat=never</FONT></DIV>
<DIV><FONT face=Arial
size=2>mailbox=200<BR>host=dynamic<BR>dtmfmode=info<BR>context=from-internal<BR>canreinvite=no<BR>callerid="test"
<200></FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>[222]<BR>username=222</FONT></DIV>
<DIV><FONT face=Arial
size=2>type=friend<BR>secret=222<BR>qualify=no<BR>port=5060<BR>nat=never<BR>mailbox=556<BR>host=dynamic<BR>dtmfmode=rfc2833<BR>context=from-internal<BR>canreinvite=no<BR>callerid="Jack"
<222></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>And I've tried making simpler a simpler one with
the bare minimum:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[111]</FONT></DIV>
<DIV><FONT face=Arial size=2>username=111</FONT></DIV>
<DIV><FONT face=Arial size=2>type=friend</FONT></DIV>
<DIV><FONT face=Arial size=2>secret=111</FONT></DIV>
<DIV><FONT face=Arial size=2>port=5060</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I haven't been able to register with any of these.
I'm probably missing something really simple, I'm sure, but I haven't been able
to find it in all of the time I've spent and I imagine it would take someone
less time to point it out to me than it would to write a message telling me how
I shouldn't have posted.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Matt</FONT></DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=guillaume@ea.com href="mailto:guillaume@ea.com">Chamberland-Larose,
Guillaume</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">Asterisk Users Mailing List -
Non-Commercial Discussion</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, February 03, 2005 2:28
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> RE: [Asterisk-Users] Calling
Asterisk Autoattendant With SIP Phone</DIV>
<DIV><FONT face=Arial size=2></FONT><BR></DIV>
<DIV dir=ltr align=left><SPAN class=442422219-03022005><FONT face=Arial
color=#0000ff size=2>I believe the web page should be modified to include a
huge, red, bold, blinking "please read the asterisk handbook available here
and search the wiki and mail archives before you post a message to the list".
That would prevent so many questions on how and where to start when first
installing asterisk. :s</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=442422219-03022005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=442422219-03022005><FONT face=Arial
color=#0000ff size=2>So, I would suggest you check out the asterisk handbook
here: <A
href="http://www.digium.com/handbook-draft.pdf">http://www.digium.com/handbook-draft.pdf</A></FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=442422219-03022005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=442422219-03022005><FONT face=Arial
color=#0000ff size=2>Page 56 to 61 explain in lots of detail and give a
working example of sip.conf with 1 phone and 1 voip provider. The whole thing
is good to read though so you might as well read the whole thing (quickly)
hehe.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=442422219-03022005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=442422219-03022005><FONT face=Arial
color=#0000ff size=2>The handbook assumes you know nothing about asterisk and
pretty much everything else. You shouldn't have to spend more than 15 minutes
configuring this. </FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=442422219-03022005></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=442422219-03022005><FONT face=Arial
color=#0000ff size=2>Guills</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=ltr
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> Matt Waterman
[mailto:waterman@dehp.net] <BR><B>Sent:</B> Wednesday, February 02, 2005
7:08 PM<BR><B>To:</B> asterisk-users@lists.digium.com<BR><B>Subject:</B>
[Asterisk-Users] Calling Asterisk Autoattendant With SIP
Phone<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><FONT face=Arial size=2>
<DIV><FONT face=Arial size=2>I'm trying to get into the world of
Asterisk in order to use the voicemail and autoattendat features (and more
stuff later) with a Redcom switch. But, I've only started and haven't gotten
to that yet. At this point my solitary goal is to talk to the autoattendant
via an SIP phone (SJPhone). I've spent countless hours trying to find the
documentation I need to accomplish my goals but everything I find always
assumes so much and I'm left lost. Plus I haven't found a thing about
setting up Asterisk as an SIP server.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I installed the <A
href="mailto:Asterisk@Home">Asterisk@Home</A> package, so I can edit all the
config files through HTTP and I can use AMP. </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I've tried 'dialing' to the IP address of
the Asterisk machine with SJPhone but the call is rejected ("number not
available"). Now, how do I specify an extension number when I
'dial'?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks for any help :/</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Matt</FONT></DIV></FONT></DIV></BLOCKQUOTE>
<P>
<HR>
<P></P>_______________________________________________<BR>Asterisk-Users
mailing
list<BR>Asterisk-Users@lists.digium.com<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>To
UNSUBSCRIBE or update options visit:<BR>
http://lists.digium.com/mailman/listinfo/asterisk-users</BLOCKQUOTE></BODY></HTML>