similar to: SIP Peering--call terminated prematurely

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

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.

I've tried timing faxes two ways:

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

ok guys I have a PHP GUI that will be great for both of you. direct editor to the whole file intact OR click to go to an extension. I will post a link to it tomorrow morning... as soon as I can get it off my production server HEHE.... it can do CRC checks on the *.cnf files and it will allow you to edit and parse out for you all your config entries with complex cnf files and default sample

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:

I'm attempting to do a first-time Asterisk install at home, firstly for use by my self and my family, 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

Hello, I have been retained by a Building Management Company to install a combined Voice/Data solution for a Tennated Office Space. This space will rent offices, with telephone and internet service to inviduals or small groups of individuals. As fate would have it, the service will be provided in a building where we have a major Pop, with a DS-3 worth of ISDN PRI circuits, 345 megs of

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

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: --

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

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

Hi, I have successfully configured two OpenBSD ( 4.2 & 4.0 ) Servers to do IXA2 peering on two remote Sites. Now asterisk users on Site1 can talk to users on Site2. I just would like to know the following details. 1) Currently I have allowed all in coming traffic from "Site1 Public IP Address" on Site2 Server and vice versa. Is that really required? Is it possible to narrow down

Hi, Anyone know any decent trunk provider in Canada that can offer multiple channel trunks (16channels) via IAX or Sip trunking? Having some pleasant experience with IAXTEL out of Denver, though they don't offer services into Canada. Please let me know S.

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

There is now a peering arrangement between e164.org and FreeWorldDialup which means any and all subscribers on FWD are now easily able to make enum calls by prefixing their call with **164, like wise it's almost as simple to make a call to FWD by hitting 8829990<fwd number> This means that for those of you wanting to send/receive calls to/from FWD subscribers you can now do so, easily

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

Has anyone had good results with an on-line database that returns a LATA based on NPA NXX? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100518/d77d09b3/attachment.htm

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

Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD