[Asterisk-Users] Calling SIP
Tim Sailer
tps at buoy.com
Mon Feb 9 12:51:57 MST 2004
On Mon, Feb 09, 2004 at 01:37:55PM -0600, Eric Wieling wrote:
> That's just the way Asterisk's dial command works.
Hmm. I see. If it can't create the channel for either reason
(busy or not registered), it's handled the same. I think I'll
kludge up a perl script to watch the SIP channels register and
unregister, and update a database table, which will be displayed
on a web page to show who is actually active.
Tim
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>> Tim Sailer >< Coastal Internet, Inc. <<
>> Network and Systems Operations >< PO Box 726 <<
>> http://www.buoy.com >< Moriches, NY 11955 <<
>> tps at buoy.com >< (631) 399-2910 (888) 924-3728 <<
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