[clue-tech] Asterisk @ Home 2.8 with FreePBX Upgrade
Mike Staver
staver at fimble.com
Tue Aug 15 12:03:12 MDT 2006
I set up an Asterisk based PBX in my office a while back using Asterisk
@ Home 2.8. Apparently the people who started that project have
completely abandoned all support for it now that they moved to
TrixBox.org. I'm having what I feel should be a relatively simple
problem, and my forum posts have been ignored when I ask about it on
their site since I'm not using the new trixbox setup. There is no clear
upgrade path to using the new system, and since I have a fairly large
set up now I have no intention of moving a few months after I just got
the other system working mostly. The one issue I have left to figure
out is puzzling to me - what's happening is that when a phone extension
registers with the server, it is somehow linked up to an outgoing SIP
trunk. My VoIP provider requires a seperate SIP trunk for each DID I
have. I wish they didn't, but they do. So, what happens is that if I
have 5 extensions, each one that registers "locks onto" a SIP trunk, so
for outgoing calls, only that extension can use it. Incoming still work
fine and will take any incoming trunk open. Currently, the solution is
to have a trunk for each extension - but then whey you try to make a
conference call, you get a message about all circuits being busy for the
second number being called, even though all other trunks are not being
used at the moment. This also means I can't have 10 internal extensions
and only 3 trunks for example, which I should be able to do if I wish.
I'm exhausted with trying to get people at the Asterisk at Home project to
help me and at this point, I would like to pay somebody locally to help
me figure out the issue. If you can help, I'm in Denver at 18th & Grant
St. It obviously doesn't have to be today - but my employees would like
to be able to make a conference call at some point here.
--
-Mike Staver
staver at fimble.com
mstaver at globaltaxnetwork.com
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