[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

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users





More information about the asterisk-users mailing list