similar to: SIP ATA's as "house" phones

I have a call center where we're looking at converting it from a traditional PBX w/ digital phone "agent" sets (keyless phones) that have headsets to a SIP based environment. I am having trouble finding anything on the market that resembles this in the VoIP world. For reference, we're currently using Inter-Tel Agent Sets, which are basically a digital phone with out any keypad,

This is not specifically Asterisk-related, but I think that sometimes a "pre-emptive" clue is a good thing. Viking Electronics (http://www.vikingelectronics.com/) has some neat widgets that attach to phone lines, which I'm sure many of the people on this list would find at least somewhat useful in conjunction with their Asterisk systems. Many of the widgets can be replaced with

Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then

@products["car"] is not being recognized in my tests: class ProductsControllerTest < Test::Unit::TestCase fixtures :products def test_view_product post :view_product, :id=>@products["car"].id assert_template "view_product" ... end end products.yml : car: id:1 name:test Does anyone have any idea why its not working? Im having to do

Hi Greeting to all you out there. I am new at asterisk, I have been working with PLMN platforms telecommunication for 5 years with NSN and Huawei. We have recently built an asterisk PBX with Trixbox and connected it to our MSS using Digium E1 cards(ISDN). Everything went smoothly as there are tons of information out there, except for the TON number. If you have worked in Telecommunication you

I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we

Hi, I'm trying to fit a glm with a negative binomial error distribution and a log link, using the example found in the paper Stochastic Claims Reserving In General Insurance by England and Verrall. I am attaching a pdf since it is more difficult to write equations in here. So to hopefully explain myself better I wrote a pdf, where I'm explaining what I need to do and where I got stuck.

On Tue, 13 May 2025, Graziano Stefani (Nokia) via openssh-unix-dev wrote: > Hi, > > With reference to the latest version of the portable OpenSSH, in > file sftp-client.c, it looks to me there may be a bug in function > sftp_upload. > > My understanding is that, when variable "len" is equal to 0, no more > SSH_FXP_WRITE messages are sent out and you start draining

I am looking for an ATA like device but instead of VOIP to analog phone I want VOIP to low level audio out. Something that looks like a sound card output. I know I can use cheap PC's but that then you have HD's to setup etc... HD failures etc... Anyone know of something like that? Jerry

Hey all, I know you all saw the topic and let out a groan. However, I understand how to get an overhead paging system to work with respect *, however I am now looking for a small(?) paging amp, that I could hook 3 or 4 horns to. I would like to just have the * extention be routed to a soundcard and out an output, so I would like an amp that is voice signal activated. Has anyone found anything

Hi, With reference to the latest version of the portable OpenSSH, in file sftp-client.c, it looks to me there may be a bug in function sftp_upload. My understanding is that, when variable "len" is equal to 0, no more SSH_FXP_WRITE messages are sent out and you start draining the queue of pending responses. Variable "len" is set to 0 either when the upload is interrupted, or

Very friggen cool, that you very much for the information it looks like it will do the job nicely. What did you use in your extensions list to activate the relay? James > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jay Milk > Sent: Sunday, 20 February 2005 6:24 PM > To: 'Asterisk

Phil I have very good experience with the vegasteam ATA's devices.(you might also want to look @ sipura ATA's, since vegastream is doing an oem on there boxes) They support modem until v.90 speeds and faxes for g3. They are expensive, and again, work great and configure very easy joash ________________________________ From: asterisk-users-bounces@lists.digium.com

Hi, Has anyone had any luck using modems on ata's other than with Cisco ATA-188's? I really don't have the money pay for the 188's as this is for my personal use. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041003/5ffde3f4/attachment.htm

Hello. I have a linux and two sip-ata's, a sipura 2002 and a GS ht-386. I also have three sipphone numbers. I can connect the atas to the sipphone accounts and I get a dial tone and I can call my house and it says, "Thank you for using SipPhone..." Using asterisk, I have the ata's registering to my computer and I register two sipphone numbers with my computer. When I

We are testing the innomedia ATA's to possibly replace our current line up of ATA's that we are using. Has anyone used their product? What is their track record on stability, voice quality, DTMF talkoff, T.38 Thanks Bryant ---------------------------------------- From: "Zeeshan Zakaria" <zishanov at gmail.com> Sent: Wednesday, October 13, 2010 10:41 AM To:

Does anyone know how the Linksys PAP2T ATA's can be mass provisioned? Documentation seems to be sketchy, even on the Linksys web site. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070607/3f90695c/attachment.htm

We have the following available for sale. All equipment tested with a 90-day warranty. (150) CP-7960G refurbished $270/ea (20) CP-7960G new $305/ea (20) CP-7940G new $260/ea (50) CP-7905G new $140/ea (200) CP-PWR-CUBE new $25/ea (50) ATA186-I1 refurbished $130/ea (arriving next Monday) Let me know if we

Are their any good chooses for IAX Adapters? -Thanks

I'm trying to figure out how Asterisk decides how often it will send SIP NOTIFY's to an ATA when a voicemail message is waiting for the user on the server. >From watching, it seems to be completely random. Sometimes 10 seconds apart, then 33 seconds, then 13 seconds, etc. Each time causes a "ring" on the ATA. I can change the ATA's behavior, but what I'd really