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<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>
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<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></BODY></HTML>