[Asterisk-Users] SIPPROXD for SIP thru NAT
Uriel Carrasquilla
uriel at adelphia.net
Sat May 10 12:54:53 MST 2003
I am jelous to see everybody being able to get their XTEN soft sip phone
working and I can't get pass ringing my Zap analog phone from the XTEN phone
in my own LAN. Is there a cookbook for idiots write up that I can use?
URiel
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Patrick
Sent: Saturday, May 10, 2003 9:27 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] SIPPROXD for SIP thru NAT
On Sat, 2003-05-10 at 14:07, Michael Bielicki wrote:
> Could we have a place to put all gear that works with * via NAT and which
one
> does not ?
>
Here is what I think or was told on #asterisk what works so far.
Analog phones:
all of them if they work on your analog phone line (thanks kapejod) and
do Tone. Pulse phones (the ol' mechanical rotary ones) are not supported
(thanks lele)
ADSI phones:
Aastra PowerTouch 390, 392, 480, Grandstream BudgeTone 100
SIP hardphones:
Cisco 7960, Snom 100, 200
SIP softphones
GnoPhone, Xten Lite, SJPhone, LinPhone, Gnomemeeting, MS Netmeeting
Networking kit:
Cisco ATA186, D-Link DG-104
Maybe the topic should be renamed to "supported equipment" or something
like that?
Regards,
Patrick
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