similar to: Any good dialplan code out there to implement vertical service codes?

I haven't dealt with Centrex for a long time, and one of my customers is being courted heavily by a Sprint salesperson. Am I not correct in assuming that each "line" of Centrex corresponds to an "extension" in the PBX world? This site has 2 POTS lines and 5 extensions, and they told me that for the same thing they're paying right now (~$40/POTS line) they will be

Hello, I had a partner of mine present a Centrex 21 brochure and ask how many of those features can I fulfill. There is nothing out of the ordinary, it's stuff like call hold, call forward, 3-way calling, etc. Has anyone assembled a how-to that shows how to configure PBX or Centrex type functionality? I found one in the voip-info wiki but only a couple of topics were filled out. Regards,

Hello all, I am working on setting up a basic Asterisk system (using Digium FXS and FXO cards, Polycom IP phones). I've got a few questions in regards to voicemail and was hoping that someone could give me some ideas... Right now we have three incoming POTS lines. There are times when all three lines are used. When the lines busy out, the incoming call is sent to a voicemail

anyone know of a solution where mobile handsets out roaming the pstn cellular network can be used and treated as full fleged centrex extentions, i.e. I can transfer a call that comes in on a wired centrex copper pair out to a cell phone and the cell phone can transfer the call back or vice versa where the cell phone recieves the call directly and can transfer to the office all without hairpinning

The benefits of * are obvious so that part of the marketing an * solution is easy. Anybody care to share ideas on how to target companies who would benefit most from */Digium? It seems to me that it would be an easy sell to small/medium companies who need advanced features such as ip trunking, IVR, Conference bridging, etc., etc. I would like to find a way to identify multi-location companies

Greetings, List. I have my Asterisk box setup with 8 Centrex lines that were "left over" from our old PBX system. My boss is asking me to set up Asterisk so that he can flash hook and make an outgoing call on the same line to have a 3 way call. This is what he wants to do: 1) Incoming call on his Centrex line 2) Flash hook and dial a new number (goes out the same line) 3) Flash

Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would

Does anyone know what would be involved in making Asterisk work as a voicemail system in a Centrex environment? We have a Centrigram voicemail system that belongs in the Smithsonian. There are analog lines coming into the box and a 56KB data feed from the phone company's switch. Peter

For whatever reason I decided to implement the validator class tonight. It''s my first go at implementing the class. My first run at it looked pretty good. I just added a generic .i for it and it all compiled. Sadly, the samples didn''t work because the base Validator class can''t actually be used as a validator! You have to derive from GenericValidator. I

I can't seem to get *80 to do its thing on a Zap channel. Looks like *8 is being seen by asterisk first, and *80 is basically inaccessible. What *80 is intended to do, by the documentation on the wiki and by inspection of the source code, is add the last callerid to the blacklist. Looking at the source, I see the same behavior coded in chan_zap, chan_mgcp, and chan_skinny. While *8

Hi, I'm looking at setting up an Asterisk PBX in our office, which gets its phone lines (digital signaling, analog voice) from the main campus, which uses Centrex. Does anyone know if this falls under analog or digital for hardware buying? I was looking at getting a Digium TDM-series, but apparently our lines aren't pots (due to the digital signaling). Could someone enlighten me a bit?

The file swig/classes/includes/wxLayoutConstraints.h may need to be updated. It''s missing centreX and centreY from the list of wxIndividualLayoutConstraints. I''d send a patch in but I''m wondering if this file just needs to be regenerated from scratch. Roy

Hello, I've been planning to replace my aging CENTREX switch with a new PBX and am seriously considering Asterisk as my solution. I work at a college, and we currently support just under 300 regular analog lines to the offices and whatnot. I was wondering.. Is asterisk ready for such a job? Would I be making a mistake to deploy this across the campus at this time? It seems that

Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1

Message: 5 To: asterisk-users@lists.digium.com From: Doug Meredith <doug.meredith@skyridge.com> Date: Wed, 21 Jan 2004 20:05:19 -0400 Organization: Skyridge Systems Inc. Subject: [Asterisk-Users] Re: What technology could my phone company be using? Reply-To: asterisk-users@lists.digium.com >>Mark Hazlewood <lists@idontknow.com> wrote: >>Sounds like Centrex services, we

Finally got around to deleting those lines in renamer.rb to un-camel-case centreX properly. Here it is. _______________________________________________ wxruby-users mailing list wxruby-users@rubyforge.org http://rubyforge.org/mailman/listinfo/wxruby-users

We are trying to implement "area-code dialing" in our asterisk PBX. Basically: we will have a number of customers, who may be in different area codes, that want to direct-dial each other's extensions. We want this to work like a "real" centrex, in that seven-digit numbers should try (1) "local" VoIP extensions, and then (2) "local" PSTN numbers.

Hi All, I hope this is the right list for this sort of request. I'm wondering if you all could recommend (or are) an asterisk integrator. I've been following the lists, etc, and have played with the software, but just don't have the time to really figure it out, nor to deliver a solution in a fixed time. I need someone who can help me spec the hardware and configure * for a small

This unit looks nice: http://www.xorcom.com/ts-1/features.html It's not very powerful but I'm sure it's more than enough as a clever SOHO end point, say to manage SIP phones properly. You probably still have to do voicemail IP-centrex style though. Any ideas about pricing? Cheers, Jean-Michel.

I need to accept an incoming call, make a series of outgoing calls, and once I find someone willing to accept the call, bridge the original incoming call to the outgoing call. Using Dial from an AGI script isn't enough because once the Dial'ed number connects, the call is immediately bridged and I need to ask the called party if they will accept the call. I can see a couple of