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I run the phone with sip firmware so I can confirm it does. ;)
Actually the "G" means global and replaces the actual text on the
buttons with icons instead. The gigabit interfaces come on the later
- -GE models. My question was more directed to if anyone has gotten
SIP hints to work on the older 7960s at all. Looks like I might just
have to give the new snom 370 a try...
- --J.
On Apr 25, 2007, at 7:59 PM, Brad Sumrall wrote:
I am very confident the 7960G has a sip load. I know for sure the
regular
7960 does and the G just means gigabit interface. The 7970 was the
only one
that didn't because of all the color interface/touch screen, and
Cisco was
still pushing call manager big time, so skinny was the only load
available.
If you log into cisco.com, they have it under software.
Sometimes people post it on the internet.
Asterisk is supposed to be more skinny friendly these days.
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Howk
Sent: Wednesday, April 25, 2007 7:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SLA Appearance between 2 Cisco 7960's
(SIP)
From reading the SLA docs, SIP hints are use to get the lights on the
phone to show the "correct state". I was under the impression that
the
SIP firmware on the 7960's didn't support the SIP hints properly
(or at
all), which means that SLA won't work properly on a 7960.
If anyone has gotten this to work, I'd like to hear about it.
--Jason.
John C. Wolosuk Jr. wrote:
Has anyone had any success with getting SLA going between 2 SIP
phones?
(Particularly a set of Cisco 79xx's) The SLA document that comes with
the asterisk source is about as clear as mud.
Does anyone have a working sip.conf, sla.conf, and extensions.conf
that
I can use for reference?
The part I'm most confused about is how to build the lines in
sip.conf
and how the phones should behave. It seems apparent that the phones
should not register with asterisk, otherwise all the phones will
try to
register to be THE phone for a given extension. should these lines be
built like a trunk/peer? if I could be an example of how lines for
SLA
should look in sip.conf, that would be helpful.
Also I'm somewhat annoyed that I have to compile zaptel drivers
that I
don't use in order to compile the app_meetme.so module so I can
have the
SLA functions available to the dialplan...
Any feedback is greatly appreciated!
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