[asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP)

[Jason Howk]

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

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: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] 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!
>>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.1 (Darwin)

iD8DBQFGMD8nyocPzc/H1dsRAtlVAJ4gLjMENCyW2wDFMhxMRO6eIX76yQCdESBt
G83ykWxG1EWcxLNqZfyp5ME=
=RVLT
-----END PGP SIGNATURE-----