similar to: Questions on various and sundry IP phones, and cabling

I would like to integrate * with an existing Altigen PBX. I want to spend as little money as possible to make it happen. My main goal is to inexpensively connect a branch office to the phone system. Eventually I would like to replace the Altigen system with an Asterisk server so I don't want to spend any money on Altigen hardware. Currently the Altigen has analog interfaces with a couple

I've searched high and lo and googled to I can't google no more... I knew that Cisco bought Selsius to get their VoIP solution but what I didn't know was that the 12sp+ is based upon an ITE-12 product that is apparently used at universities. I've taken two of the phones apart and started swapping parts and have looked up the IC's on the Internet... Anyway... After using

I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that

Has anyone got Asterisk talking successfully to an Altigen PBX using h323? I can successfully make calls from Asterisk to Altigen, but calls from Altigen to Asterisk get a fast busy. There seems to be a lack of h323 example files (or maybe I'm looking in the wrong places) as well as a severe lack of h323 documentation from Altigen. Any pointers would be greatly appreciated.

After much struggle I am finally beginning to get some action from wine (codeweavers). Can anyone help me with the following: 1) A small program that calls an atomic clock in America to check the time and corrects that on the computer shows its window but says "external exception 80000100" when I ping it or leave it for a while. 2) WebDwarf - a website builder - seems to work quite

Greetings All- I have an * box with the NuFone H.323 channel driver installed. I also have an Altigen VoIP system with a PRI to the PSTN. I can sucessfully make a call from a SIP extension (snom190) to an H.323 extension (altigen phone) The thing I can't seem to make work is a call from a H.323 phone to a SIP extension. Here's the layout:

Hi! I have two, not too related questions: - the probably simpler one: if anyone can help me out using a Cisco 7905G with chan_sccp? I did already managed to get it working with a SIP image, I'd just like to see it work with this one as well. It's probably something I screw up with the configuration, as the phone registers, only I don't get any lines with it, although I have it

Hi Everyone. I have a client that uses an Altigen system. I am really new to PBX systems so all this is totally foreign to me. They currently have 5 inbound trunk lines and about 20 analog phones. >From what I can gather they are using the Altigen Quantum cards that support 8 extensions and 4 trunks. >From what I can gather the solution is a TDM04B and TDM01B to bring in the lines from

I have a situation where we have an asterisk box that is extending several Mitel PBX extensions to some cordless SIP phones (Cisco WIP310). Everything works great, except when the cordless phone walks out of range of one access point and into range of another (cisco 1100 series APs). I've been able to get virtually seamless roaming between access points to work in the past with data but

After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of which do not have "phone company" compatible FSK/stutter MWI, I finally got smart and found out just which Panasonic phones have this feature. Only the following 5.8G models in their current line have FXO compatible MWI. I purchased the 5771 unit and one remote. I have confimed it does in fact work with Asterisk

Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn't cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. Thank you.

Hey Folks....I am on the 1.2 branch with the latest from Subversion. I've been having a rough go for the last several months integrating asterisk with out Altigen system. I can get calls inward just fine. I have zero missed interrupts on the digium 110p card. I have zero frame slips according to both sides. Outgoing calls sometimes work, but more often than not I get the following: --

Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some

Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some

Hello. Is anyone aware of a manufacturer of DECT cordless phone interface cards to act as the DECT base station that would be addressable channels under * ? Regards Petar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040418/c8885bbc/attachment.htm

Hello! Attached to the bottom of this e-mail is an edited version of an e-mail I originally wrote to Digium tech support regarding Ouch and Power alarm errors I have been receiving on my TDM400. It contains a great deal of detail regarding my setup. In the end, I have found that one of the 5 phones I'm trying to make work with Asterisk is contributing to the generation of these errors.

Hi! I have a Grandstream VoIP Device, at which a DECT base with 2 cordless phones are connected. If a call is placed and made through one cordless phone the other cordless phone appears as busy. What I want: 1. The Base station of the DECT cordless phones, is connected at 1 FXS Port of my Grandstream Telephone Adapter. 2. I want to place and receive as many calls at the same time through 1 SIP

In that example you could make outgoing calls only correct? (since incoming likely needs port forwards) I guess the questions becomes "how often are you going to do that to justify the extra $100 or so you going to pay for a wifi sip phone?" Paul Fielding (paul.fielding@shaw.ca) wrote: > > I think some people are missing the point. You can't throw your cordless >

This senario works great! I have a Snom phone coupled with a Linksys Wireless game adapter. When on a event trip I was able to have the snom ring where ever there was a signal. So the proof of concept passed. Unfortunatly all the wifi phones I have read about seem to suck! -----Original Message----- From: asterisk-users-bounces@lists.digium.com <asterisk-users-bounces@lists.digium.com>

Hi, I have a problem with this particular gamepad. I calibrated it under linux with jscal and it is working OK. But now I want to play FIFA with it and fifa don't detect it as Logitech WingMan Cordless Gamepad but as controler. I found wine registry tips and followed their guide for DirectInput but I have a problem because command specified there doesn't give the expected output: Code: