similar to: off topic - SSH configuration for Digium Support

Has anyone here ever tried using a wifi bridge to place a deskset in someplace where there was no LAN drop? If so what hardware did you use and was it succesful? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005

Hi All, The past day or so I've setup a new * server based upon the Xorcom Rapid ISO. It did as promised; wiped the base system, installed Debian OS, installed Asterisk with a dummy configuration. So far so good. If I could get the config from my existing * server migrated to the Xorcom box the I'd be ready to roll. Essentially, it would be the same as I have now on Fedora Core 1, but

OK. Now I show my ignorance. What's the difference between a T-1 and a channelised T-1? I see that Covad's voip service (formerly GoBeam) requires a channelised T-1. Then I read recently on the list that many T-1s being installed are actually HDSL....which would be not a T-1 at all...right? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

I am currently having our new office wired up with 8 PSTN lines. The guy asked me if he could wire it up such that one line had two phone numbers. I bought a Sangoma A200 with 8 FXO ports, but now I'm wondering if all I needed were 4 FXO ports. Is it possible to set up Asterisk with 2 numbers per FXO? Thanks for any help, Andrew -------------- next part -------------- An HTML attachment

My router (1605R) currently does not support QoS. Is there any open source software available so that I can set one up before the router? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/38b6e3f2/attachment.htm

Hello, I bought a copy of this software in the hope of bridging Skype into my * box. It sort of works but the audio is all distorted and there's huge latency. Does someone have this working well with their * server? According to the wiki someone does, but I don't know who. I'm needing advise on what to fiddle with to get it working correctly. Thanks Michael -- Michael Graves

Hi All, I'm using DISA on my * server to avoid overseas toll charges when making calls to Western Europe from my cell phone. I have DISA working with a DID from a VoicePulse Connect account. The outgoing call to Europe is also made via Voicepulse Connect. I see that the IAX media path is bridging the inbound call to the outbound call so that the media stream entirely bypasses my server once

I see that ABPTech are now offering this SIP Wifi phone. Does anyone have an experience with it? I've considered the Pulver/Zyxel wifi phone but would really rather have something that can handle WEP or WPA. I run an 802.11g wlan with 256 bit WPA encryption. This Hitachi phone looks very much like a cell phone, which could be ideal. Michael -- Michael Graves

Anyone have one of these yet? http://www.voxilla.com/shop/index.php?action=item&id=38 -Dan

Hello: I am new to the list. I am trying to set up asterisk with voicepulse. I have a voicepulse username + password, and SIP DID. When I login to voicepulse, I have this under my devices tab: Devices *Login:* Sysxxxxxxx *Password:* xxxxxxxxxx *Context:* VPWS *Connects to:* gw5.voicepulse.com My question is: Do I need a 2.4.x kernel? Currently I am running Debian/stable stock 2.2.x ? Has

Hello, I just started using asterisk, and have a question. I have setup two asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1 FSX modules) and is connected to the PSTN. B has same, but is NOT connected to PSTN. I want to configure B to call A via iaxtel, and connect to the PSTN using A's line. How can I configure iaxtel dial plan for B in extensions.conf? I want to be

Hello All, I have a small * server in my home office with several IP phones. The system is not fully in service yet as I'm still hunting for a cost effective FXO adapter that I can rely upon for my two primary PSTNs. That said, I'd like to move it into service for another application...which brings up a question. I'd really like to stop making international calls from my cell phone

How about it. Anyone use these guys? Their rates look ok. I'm looking for alternatives to VoicePulse Connect that provide DIDs in Houston TX. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262

Is there some concensus on where to set callerid when making outgoing calls via an ITSP over IAX? Is this best accomplished in IAX.CONF or EXTENSIONS.CONF? Also, tech support at one ITSP told me that the SetCIDName command doesn't do anything. Is this something that might be unique to their server? Or a general statement of fact? Thanks, Michael -- Michael Graves

Is anyone here able to make calls to FWD via IAX? I haven't beenable to for some while. I'd like to get to the bottom of the problem. There's been little response in the FWD support forum thus far. I can call my own number and it rings my server, but I cannot call any other number. It generates and error reporting "everyone is busy at this time." Michael -- Michael Graves

Does anyone on-list have any experience with this device? It's a 6 port FXO<>SIP gateway. Michael Graves -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245

Does anyone have any experience using this DLink quad FXO <> SIP gateway with Asterisk? I'm still looking for an analog interface that I can live with, having tried X101p, SPA-3000 and TDM400...all with less than desirable results. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power

Does anyone on-list have direct experience with the new analog cards from Sangoma? I'm thinking about FXOs with echo cans. Need 2-4 ports but don't want to go through another TDM400 style experience. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

Hi everyone, I just got a Sipura SPA-1001 and have connected my Tivo Series 1 (yes its old). When I do a test call with Tivo, the call always fails (it seems to dial the number but never connects). I can pick up the phone line and hear the Tivo "talking". I've tried looking around for anything special I need to do but its still not working. I can connect a phone to the SPA-1001

Hi All, I've read J.R. Richardson's paper "Create an Embedded Asterisk Server" which outlines making a Debian server that boots from a compressed disc image on a CF card. I'm really interested in this as I want my * server to be more like an appliance than a PC. However, the paper is only an outline and some of the processes of pruning the installation down to a minimum are