Give this a whirl:
www.asteriskdocs.org
Just click on the image of the book cover and you'll get the PDF.
Jim
Darryl Moore wrote:
Hi list,
BTW, great to see your group at LinuxFest. It was my first time out
there, but it won't be my last.
I've managed to hack a few Panasonic JOIP/Globarange phones so that they
are freed from there original service provider and now work for me on a
standard SIP network with Asterisk.
I did this before on another computer and had it working well with
outbound lines and everything. Now I can't even get the extensions to
talk to each other. When I try I only get "That number is not valid" or
"The number you have dialed is not in service."
All the extensions appear on the flash panel, and I was able to have
time and echo test working, though I think I recently broke those too.
Do I need to set up a trunk or incoming or outgoing routes in order to
get the various extensions to talk to each other? If there are any
tricks that I am missing I'd love to hear them.
I don't want to post all the config files and ask you guys to debug it
for me. I just want to know if there are some basic principles i need to
apply to get a simple set of extensions working. I've so far not found
any faqs to this end.
Thanks
darryl
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Jim Van Meggelen
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http://www.oreillynet.com/pub/au/2177
"A child is the ultimate startup, and I have three.
This makes me rich."
Guy Kawasaki
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