similar to: Finding the busypattern

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

I have an adit 600 with one fxo card connected to a Digium single span T1 card. CallerID, disconnect supervision work perfect, however the users complain that they have some sound quality issues, after testing it I realized that whenever one is in a phone call they get like silence between the sounds coming from the other party, almost like a cell phone, in other words if there is no sound coming

Thanks for your suggestion Steve. I have done as you advised and set busypattern=300,200 to match the sample I recorded. This hasn't worked though, asterisk doesn't seem to detect the busy signal. Does asterisk require a the signal to be in a certain power range? The signal I get is very quiet. Thanks for your help Regards Jonathan On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s

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

We have an Adit 600 with some fxo lines. Callers calling in on the fxo ports do not hear echo, but the party answering hears bad echo. Here is the gain settings on the Adit: Corporate Adit 600> show 1 SLOT 1: Settings for FXO: Channel 1: Signaling: LSCPD RxGain: -2 dB TxGain: -3 dB And here are the zap settings: busydetect=no callprogress=no

Hey guys, I've been struggling with hangup detection on a Centrex system for a bit now, I was on site Saturday and I took my DMM with me. It would appear that this Centrex service provider uses polarity reversal at the beginning of a call and several times after hangup. I've been reading the archives for the last year and it looks like there was some work done that matches my

I just noticed that quality of .gsm files for using with asterisk is not that good.. is there any way to make then sound better? asterisks sample voices sound way better than theones recorded using applications like wavepad or with asterisk like unavailable messages... any tips? Do you know the command line for sox to adjust the volumen levels or gsm files (make the louder)? Also, do they have

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

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

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

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

Hello all, I have a new Carrier Access Adit 600 and am having a problem dialing out and receiving incoming calls. After the initial ring, the calling party will here nothing. There is a silence, but the SIP phones that is supposed to ring will ring. The calling party will not hear the sound of the ring that * sends out on the line. Additionally, if someone picks up the receiver there is no

stuff in the dom0s, so the domUs only need to see actual block devices. > That''s one of the difficult issues with managing storage with puppet: > initial provisioning of the storage has to happen before puppet has a > chance to run, so you''d always have to resort to another tool (aka the > installer) Yeah, but only for the initial installation. I don''t mind

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 people, this is my first mail to this list. I'm new to asterisk and trying to set up an IVR. So far my dialplan works nice connecting with softphones, but I'm having problems to detect hangups on the analog line. Here are the details: -Clone pc with Ubuntu server 10.04 64 bits -Asterisk 1.6 -Openvox A400P, lspci -v says: 01:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX

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

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?

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Hi: Does anybody know of a TDM interface card for *digital Centrex* that will work in Asterisk? We're not talking about BRI, here -- the lines have Nortel digital sets on them, and we want to run them into an Asterisk PBX. Centrex is more widely used in NAm. -Stephen-