search for: altigen

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 t...

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...

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...

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 Aster...

I've tried timing faxes two ways:

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: <snom190{1235}>==[192.168.0.0/24]==<aster...

...amily, and secondly as a learning experience. I've got a new house, and the previous owners removed all but one (1) phone jack. So I figured I might as well build a PBX. Functional goals include station-to-station calling, rudimentary auto attendant/voice mail, and perhaps tieing into the Altigen box that I've got at work via h.323. But my first goal is to get any of my devices to talk to Asterisk, which so far I've been unable to do. Hardware in-hand consists of a K6-2 based 2.6 kernel Gentoo box, two Selsius 30 VIP phones, a couple of Selsius 12SP+'s, a loaner Altigen IP...

...lacking at present for a newbie like myself. And - yes - I've read the manual from end-to-end several times already :-) My testbed is a dual Xeon RedHat box (shows as a 4 CPU setup to top), tomorrow should provide a 4 FXS card and an FXO card and I have a fully deployed H.323 VOIP environment (Altigen) to play with. I'll snag a PRI card after I get things squared away - * will be my PBX backup to the Altigen until such a time as it proves itself superior... -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Gary...

I've been digging around with some cisco engineers for about a week & I finally got an encouraging response to the Auto-Answer issue with the SIP Phones. Here is their reply: =============== == FROM CISCO == =============== Auto-Answer feature is introduced in SIP IP Phone 6.0 version. This software version is expected to be available for customers shortly. Please let me know if 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: -- Executing Dial("SIP/2352-9056", "Zap/g1/2352")...

...upport from Digium and/or other Asterisk experts into the budget. Does Digium have paid support plans? What about other consultants out there? I'm just trying to make sure that I cover all the bases. This is got to be a bulletproof solution, and I'm departing from my comfort level with Altigen to give Asterisk a run for the money. We've got TONS of Linux experience here, and comfort with customizing code, so I am happy with what Asterisk gives me.. What else should I be worried about? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n...

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 fee...

We have a customer considering migrating from a large Nortel PBX with a third-party voicemail system to Asterisk but one of the features they really like is the automatic synchronization of voicemail between Exchange and their voicemail system -- delete a message from the voicemail system and it is deleted from their email inbox and vice versa. Searching has not revealed anything like this

Greetings, Can anyone tell me if there is any equipment or way to configure Asterisk to support station page and if there is an operator console that works with Asterisk? Also has anyone implemented a PA extension using asterisk and if so how was it done? Thanks, Gene

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 fee...

Hi- I have an upcoming order for a bunch of asterisk boxes, and I'm considering using an "assembled" package for the server, instead of building them from components as I usually do. Does anyone have experience with the Dell PowerEdge 750 server, or any other 1U rackmount server for use with asterisk? Thanks in advance Scott Stingel Scott M. Stingel President, Emerging Voice

...>configuration with analog trunk lines, groundstart signalling is the > > only > > > >>way to prevent Glare. > > > >> > > > >>I just purchased two TDM400P's for a system I'm building to replace > our > > > >>office PBX (Altigen). Since there are no statements anywhere on > > Digium's > > > >>website about lack of groundstart support (Actually, to the contrary > > they > > > >>boast about all the signalling support in their sales slick), I now > need > > > >&gt...

Hi, I was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian.