<!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>Oh wow, look at that. Don't I feel stupid, haha.
All right, it works now, thanks! :/</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=asterisk@ropeguru.com href="mailto:asterisk@ropeguru.com">Robert
Webb</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> Saturday, February 05, 2005 12:05
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> RE: [Asterisk-Users] Calling
Asterisk Autoattendant With SIP Phone</DIV>
<DIV><BR></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=650400317-05022005>Matt,</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=650400317-05022005></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=650400317-05022005> I thought that DIAX was an IAX based phone not
SIP based. If this is the case then you need to be putting your configs in the
iax.conf not sip.conf file. I have several iax soft phones I have been testing
and have them registering with asterisk. If you want, I can email you the
config I have for them off-list.</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=650400317-05022005></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=650400317-05022005>Robert</SPAN></FONT></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> <A
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</A>
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Matt
Waterman<BR><B>Sent:</B> Saturday, February 05, 2005 11:38 AM<BR><B>To:</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[Asterisk-Users] Calling Asterisk Autoattendant With SIP
Phone<BR></FONT><BR></DIV>
<DIV></DIV>
<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"><U>2c9110f8126bc7c12ea475460afb633c@192.168.9.101</U></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"><U>Chamberland-Larose,</U></A><U></U><U>
Guillaume</U></A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com"><U>Asterisk Users Mailing List
-</U></A><U></U><U> Non-Commercial Discussion</U></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"><U>http://www.digium.com/handbook-draft.pdf</U></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"><U>Asterisk@Home</U></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>
<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>