similar to: Asterisk support for NEON or Centrex/CLASS/VMWI MW formats ?

Off topic but I am after a DECT phone to connect to my sipura 3000 that has a FSK VMWI light or flashing envelope on the LCD screen. Any ideas Chris

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

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

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

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?

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

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/eae682d3/attachment.htm -------------- next part -------------- Hello Whenever I receive voicemail on CISCO or SNOM or Analog Phones (Scitec), I would like to have some kind of indication in terms of beep sound or blinking voicemail indicator....Could you please tell me the way to enable

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

The TDM410P doesn't support 'hvac', only the obsolete TDM400P supports that option was for the old phones that have a neon light (or equivalent LED+ZENER ciruit). Are other phones off the TDM410P (other than the VTECH) working, or is the Vtech the only model with VMWI available to you. I'm not able to check at the moment, I have copied the asterisk-users list, someone else may

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

see below I tried to call 98013356 from the following phone (from mgcp.conf) [iptlf03] host = 192.168.33.3 context = default inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4

Basically, pretty much all the 2 line cordless systems I've seen come with a built in digital answering system that I'll never use, the main problem with this is that these units don't support VMWI (visual message waiting indicator) with telco supplied voicemail. This is a problem because I'll be setup an Asterisk system in the next month or so to handle my 2 analog and 1

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

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

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,

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry

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

asterisk-users-request@lists.digium.com is believed to have said: >> Haha....well the MWI is the blinking blue LCD. The message button >> is "reserved for future use" Hang in there. There will soon to be some >No the message button call the number you configure in the web >interface. Presumably voicemail, but could be your mistress. Then it's easy to get

Hello all, I'm looking for some help to try to understand why my CPE doesn't work good with Asterisk in MGCP. Here is what I want to do : - Register a TECOM AH4021 on Asterisk in MGCP with the following profile in mgcp.Conf : [general] port = 2727 bindaddr = 10.95.20.1 disallow=all allow=g729 allow=alaw 020202020202] context=mgcp host=dynamic canreinvite=no dtmfmode=rfc2833 nat=yes

Hey everybody, I've had issues with detecting call disconnects on the Centrex system and an Adit 600. I've discovered that if I set the channel bank's FXO to lscpd, even though the CO says they don't support, that Asterisk can see the hangups. The catch? Asterisk can no longer seize the line. I've used this to my advantage, since I have 13 lines to work with. (5) +1