similar to: AW: Large Enterprises using asterisk

Hi everyone, our setup includes a NEC PBX connected to our asterisk via bri lines. The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door. So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so. Setup: asterisk

> Hi all, Hi Marco, > I was thinking about integrating an old PBX with Asterisk and I was wondering > some possible configurations. You didn't mention the number of lines your PBX uses, but think of a third scenario: Install an asterisk with twice the number of BRI/PRI-Ports your current PBX has. Connect half of them to your carrier, the other ones to your old PBX (Some sort of

I'm trying to find a way to generate an 'internal extensions' tonelist but I can't seem to find anything on how to do this. My idea was to start a Playtones(intercom) tonelist and not indicate ringing to the line (dead air). But then, somehow StopPlayTones needs to be run once the ringing telephone picks up. This seems like a dirty way to do this. I envision an option to the

Is it possible to set up Asterisk without any of the cards? I'm interested in setting it up for the company I work for, but I would like to set it up and see how difficult it will be before I start having the company spend a chunk on equipment. Additionally, what phones can be used with Asterisk? we currently use a NEC Nitsuko phone system with phones, but I have been confused as how to set

For all the peoples that wanted to test my windows IAX2 phone, I've put it up on a server where it can be downloaded. For the ones that wanted the DLL, it's available on the same page. For the DLL, I will post a list of the functions in it and the parameters it expect as soon as I have some free time. All comments (good or bad) are welcome The phone can be used mostly with the keyboard

We're getting ready to ditch our hosted virtual PBX for an Asterisk solution and I've got a couple questions that probably come from experience; we're looking to host two PBXs on the same box if the PBXs are identified by different DID numbers, and I was curious if: 1. Is it possible to have duplicate extensions between the two PBXs? Eg: 555-1111x100 and 555-2222x100 on same

Hi all, I have just been reading an article on the asterisk-doc site about ISDN X-Over cables. The article mentioned the converting of an NT1 to make this possible, has anybody got the information required to modify a BT NT1? Or any information on the BT NT1. Thanks in advance. Regards Dave

Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both boxes. I need to connect both PBXs with E1/R2 and UTP cable. What are the requirements to deploy the UTP cable ??? Straight-through or crossover ??? What are the pinouts in both peers ??? Thanks a lot, Alejandro

Hi, I'm buying an interface for my Toshiba DKT424 to connect to *. Initially it is a trial to route internal calls over our wan. Think of it as a tie trunk between 2 pbxs. I expect to get PRI from the telco our layout will be like this: telco_T1_PRI to * to various PBX or TDM or IAX to * to other pbxs My question is, is there benefits to connecting with PRI over a standard T1 within my

Hello. I am going through the documentation and trying to find if asterisk can help me in my case. It is quite difficult to find answer because I do not know the exact question. I have two location. Each in different country. Both locations have Siemens HiPath - different type and software. I can not use card that would allow me to connect those PBXs using SIP. But I have some free ISDN and

Hi to all, i already searched the archive without finding a solution to my problem. I have asterisk installation 1.2.18 to support multiple virtiual PBXs. I use SIP peer in the format <ID>-<EXT> to let every virtual PBX to share the same numbers of EXT. Ex. (PBX ID 10 Extensions) 10-101 10-102 10-103 (PBX ID 20 Extensions) 20-101 20-102 20-103 I use some rules in the dialplan to

Hi Stefan, > At the moment I'm into the Bleuz mailing list for getting the SCO > feature to work, starting with my laptops BT and my dads Nokia. > > I have now a basic understanding from the RFCOMM layer which delivers > the AT commands, AT works. If I get the basic SCO functions > working with > the test tools, to get the headsetfunction itself into the > pc.

I've read quite a bit in the older mailing list posts and the wiki but I'm missing some simple point. 1) What is required to send an SMS to a mobile outside the office given: Channel: ZAP/1 send it to $SMS_RECIPIENT (which includes the final "extra" digit) via $SMS_CENTER=the national message center server for sending messages $MESSAGE= the message text How is the .call file

When CDR reporting was raised a few days ago, it prompted me to add a section on how ADTransform could be used to address the problem raised about consolidating CDR information from various divisional PBXs and producing consolidated reports. I wrote a short Use Case article. http://www.artifact-software.com/?page_id=1666 I also thought about the problem of getting the configuration files into

Hi, I'm wondering how I can support dial by name across 2 different asterisk systems. If I create a voicemail account (add an entry in voicemail.conf) for both sets of users on both PBXs will this allow callers that use dial by name on each of the IP-PBXs to dial any user on either IP-PBX? I assume that the dial by name application uses the context that it is called from, so if dial by

Several people have asked about integrating asterisk in a company that currently has a PBX installed. Considering the capital outlay for most PBXs, depending on the model and age of the system, most companies do not want to look at replacement. So, what should one look at in a PBX in order to integrate asterisk into a pre-existing PBX? I have read of some systems with T1 interfaces, but what

I must be doing something stupid, but I can't figure it out. I have three PRI lines connected to Asterisk, one from the phone company, and two more connected to PBXs. Asterisk uses the incoming DID information to decide which PBX to route the call to. Should be simple. Asterisk is clearly getting the caller id info from the phone company: -- Accepting call from '512345xxxx'

Hey guys, Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Ive read the Success stories form voip-info.org but

I have several Linux machines some running on really old hardware and some on brand new, some run old distros (RedHat 6) and some new (FC3 or CentOS). All of them experienced power failure more then once, none of them has failed to load after a reboot. BUT, Asterisk is running your PBX. Your PBX isn't your proxy server, it isn't your web server, mail server, firewall, or whatever

hi all, i need an asterisk board that can support up to 30 FXS ports. i found the following on digium: TDM400P. It only has 4 port, does it mean that i have to buy 8 of those to have my 30 FXS line? Or is there any channel bank solution that digium or other providers can provide us with in order to do that. Experienced asterisk user, how do you manage to make PBXs for 30 to 100 regular phones?