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aaron.fransen at gmail...
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 Posted: Thu Mar 20, 2008 9:59 am Post subject: [asterisk-users] Multiple sites, same extension |
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Holy Mackeral. Ignore that last message. I still do NOT know how to route
calls with the same extension being used in two locations, however the issue
I've resolved is getting Cisco CallManager and Asterisk talking together
properly.
Sorry folks AGAIN.
So if anybody has ideas on how to have extension 2000 in two places I'd
appreciate the advice! Thanks!
On Thu, Mar 20, 2008 at 7:14 AM, Aaron Fransen <aaron.fransen at gmail.com>
wrote:
| Quote: | Must be having a "DOH!" week.
Problem turned out to be the Fedora core firewall that was turned on.
Sorry folks.
On Wed, Mar 19, 2008 at 3:01 PM, Aaron Fransen <aaron.fransen at gmail.com>
wrote:
| Quote: |
Finally got my Cisco Call Manager link going; what it turned out to be
was having the same extension on the Asterisk system and on the Call Manager
side of things. Changing the extension on one side fixed it. Which brings me
to...
I need to have the same extensions on two sites. So if I use an 8bbbeeee
dialing plan (8, then bbb location, then eeee extension), site 1 might dial:
8-099-2000
Site 2 might dial: 8-101-2000
A 2000 extension exists on both sides, however Asterisk doesn't seem to
like it and drops the call before it even hits the logging facility and send
a busy signal back to the caller.
I've tried a dialing plan like:
exten => _8101XXXX,1,Dial(SIP/${EXTEN:4},,r)
to no avail.
Thoughts everyone?
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anselm at hoffmeister-...
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| Posted: Thu Mar 20, 2008 12:21 pm Post subject: [asterisk-users] Multiple sites, same extension |
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Am Donnerstag, den 20.03.2008, 08:59 -0600 schrieb Aaron Fransen:
| Quote: | Holy Mackeral. Ignore that last message. I still do NOT know how to
route calls with the same extension being used in two locations,
however the issue I've resolved is getting Cisco CallManager and
Asterisk talking together properly.
I've tried a dialing plan like:
exten => _8101XXXX,1,Dial(SIP/${EXTEN:4},,r)
to no avail.
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Hi Aaron,
for my personal taste your Dial() command is lacking a SIP domain (or IP
address). Consider location A (Asterisk 10.1.1.1, prefix 8101) and
location B (Asterisk 10.2.2.2, prefix 8202), where users at B want to
dial 81012000 for extension 2000 at location A.
In that case, your Dial command looks like
Dial(SIP/2000,,r), which looks pretty much useless, unless one of B's
local (see, local to B, not A!) SIP peers has a [2000] stanza in
sip.conf, and even then you would not call peer 2000 at A, but at local
(B).
If you replace your command with
Dial(SIP/${EXTEN:4}@10.1.1.1,,r)
the world looks completely different.
At least I hope so...
BR & HTH
Anselm |
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