similar to: Linksys BEFW1154 v4

I have added a 4th NIC to my setup, and want to set up wireless. I have stared at the configuration Tom has for the last week, and my eyes are crossing. eth0 "net" goes to my internet connected firewall with a 192.168 address eth1 "loc" goes to my switch connected to local switch also 192.168.x eth2 "work" goes to my office with a 172. address eth3 Trying to

I have added a 4th NIC to my setup, and want to set up wireless. I have started at the configuration Tom has for the last week, and my eyes are crossing. eth0 "net" goes to my internet connected firewall with a 192.168 address eth1 "loc" goes to my switch connected to local switch also 192.168.x eth2 "work" goes to my office with a 172. address eth3 Trying to

I have been studying Tom''s configuration at: http://www.shorewall.net/myfiles.htm -and- http://www.shorewall.net/NAT.htm I am using SBC as an ISP and also have 5 "real" IP addresses and because of other issues, have to re-do my set-up. If I have a block at .120/29 assigned to me, what SBC does is give you 5 usable addresses, in my case .121 is the SBC modem/router and

Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. A few quick comments: - I started a Wiki page at voip-info to post issues, firmware news, etc. I really like the wealth of info on the GXP-2000 page, so I wanted to start something similar for this phone. http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300 - My kit

Hi everyone, I'm trying to set up a VOIP mass deployment. To do so, I want to generate a configuration xml fails. I read somewhere that I had to use : http://<phone -or- device ip address>/admin/spacfg.xml but it work with an upper firmware only. My Software Version is 3.3.6(GW) The last firmware versin is 5.1.10(GW) Upgrade Enable: is set to yes on the linksys SPA3102 web interface

While we have not been able to get our hands on this model yet (we have the other Linksys models) the Linksys voice products are based on the sipura technologies ATA and will work. I was under the impression that the only shipping WRT54GP2's are provisioned and locked to ATT (for now). Also keep in mind that you must be an ITSP and get authorized by cisco/Linksys to be able to buy all Linksys

So, I'm setting up my first bridge, and I'm running into an interesting issue. I have a 4 port NIC (formerly 4 seperate /29 and /28 LANs) and an onboard NIC (to my ISP). I've setup the bridge using the script included at the bottom of this email (modified from Gentoo for Aurora). When the WET11 (the WET11 is a simple ethernet to wireless bridge) is unplugged, the bridge works

I came up with this a few days ago, mostly used the wiki examples, didn't have time to post on the wiki yet, maybe one of you guys with a few minutes can throw it up there, really, I forgot my logon. http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom The agi script didn't work for me, wouldn't call the active hint extensions, even though they were there, no

And the firewall is in between the phones and both servers or are you registering the phones on a local server and trunking to the other server through the firewall? In terms of nat and Cisco 7960s I've never had a issue registering two of them behind nat to a server on the other side, however, if you called one phone from the other, you'd end up with one way audio. -----Original

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

Hey, I'w got a problem (bug maybe?). I have recently got my Asterisk to work perfect and I'm not trying to setup some dial routes and get the system working as I wan't it to. Yesterday I was installing Festival and also did a "aptitude upgrade" on my Debian Unstable installation. After that the problem started. After some serious testing yesterday night and today I have

I have two old Sipura 3000 original (green) boxes and they are running perfectly faxes going through, no echo. Firmware 2.0.13(GWg), hardware 2.0.1(1813) Does the number in the bracket means anything? In addition I have Linksys 3000 running the same firmware and same hardware (except the number in the bracket). I can not get to work correctly. The unit has a lot of echo, can not get rid of it,

I recently got a Pulver Innovations WiSIP wireless SIP phone to determine if we want to use them in our organization. Since the WiSIP phone arrived last week, I have had nothing but headaches. I do think I now have the problem narrowed down. I have spent a bulk of my time trying to get the WiSIP to work with a couple of Linksys WAP11 Version 2.2. I have met with no success in getting the WiSIP

Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net

Hello I successfully hooked up a Linksys 3102 SIP gateway (http://www.voip-info.org/users/683/21683/images/716/SPA3102_lrg.jpg) to an Asterisk server, but since it's connected to a PSTN line, I must make sure it cannot be used by unauthorized users from the Net. Actually, even legit users with an account on the Asterisk server shouldn't be able to use it (outgoing calls should go

Will the DAHDI with OSLEC have any effect one ECHO in Linksys or Sipura adapters? I know Dahdi is a replacement for Zaptel. And Zaptel is mostly for internal cards not external units like Linksys. I have one Linksys 3K that with so echo so bad that it is pretty much useless. -- Joseph

I''m not a subscribed user, so please cc me on any replies (fier0@bigfoot.com). I know this has been asked a few times, but i have not been able to find a direct answer. I was using shorewall with 2 nics, and it worked fine, except if that linux box went down then nobody could get out to the internet (and the wife would kick my ass). I''ve now started to use my linksys

Hi All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone --> Asterisk --> Linksys SPA3102 --> PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use)

It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off is a workound, but non-ideal in may situations. Apparently the asterisk devs don't even think

Note that the newish SPA962 has 6 appearances and a color screen. I've noticed that the bright color screen does impress people when they first see it. PoE is also very nice and web provisioning was quite easy. I've yet to try a more automated provisioning method on it. I know that getting the polycom's to auto provision wasn't very straight forward. I do provision some