similar to: Snap for Asterisk

it work with just the main ringing phone and not have it pickup the rest of the phones in your group. That's more what you are looking for. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of mitcheloc Sent: Saturday, June 23, 2007 10:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh

http://www.crn.com.au/story.aspx?CIID=76033&eid=4&edate=20070320 The company developed Response Point to work alongside traditional phone systems or voice-over-IP systems. Continuing its recent foray into the market for digital communications products, Microsoft on Monday introduced its first packaged digital phone system for small business. Anyone know anything about it?

We are two programmers who are passionate for Asterisk and we will be dedicating the next three months towards programming for Asterisk and would like to get some input from everyone on what they feel Asterisk is lacking or needs based on what is not currently a part of it or available through third parties. Hopefully, by asking up front we won't be wasting our time on something nobody wants

Where's it gone? Going to http://www.snapanumber.com/ goes directly to the digium site with no indication of where it is ... Has it gone forever? Gordon

Hey everyone, I've been doing a lot of research into a decent server for Asterisk but I seem to be running and circles and now I am turning to you. The issue I have is it needs to be rack mountable (so a Dell SC430 isn't going to work) and preferably have 3 pci ports. The problem that I seem to be running into is that when I look at servers from Dell or IBM or the like they only seem to

Hello, I'm trying to receive faxes with asterisk. My configuration is like this: PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf:

I have a customer with a more or less unique need. Right now we are using Wombat as a dialer software so they can contact clients for QA purposes. Everything is working very well and their contact center productivity is way up from the old manual dialing method. The only thing we are having a problem with is that they have up to 5 phone numbers to contact a single customer. Obviously

Hi Tilghman!   > Clearly, you missed the point.  Since there is a FREE predictive dialer out > there, and your product costs something, you are not the world's cheapest > predictive dialer.  I respect your wording and the way you or other people think on the list about difference between cheapest and free predictive dialer.   Surely

Does anyone have any experience using asterisk with an external predicitve dialer, like MediaTel? Specifically: The predictive dialer dials out over T1 circuits. It connects to asterisk via amphenol cable from an fxs card in the dialer to asterisk with a tdm2406 fxo card. In the analog world, the dialer dials out through the t1 circuit, and the fxs card is plugged into a 66 block so the

Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with "no, this is absolutely not for doing call marketing". I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students

Hi BJ, Further explanation about the 3rd party ecosystem question this morning Cory Andrews from VoIP supply was on the Voip-Users conference call last week. I asked the question - how much of VoIP Supply revenue is product versus applications - he said we don't sell any services such as ITSP hosted Asterisk so I replied that wasn't what I was thinking of and gave the example of Snap

One of my user is using asterisk 1.4 based Dialer i.e Vici 2.0. Today, when they downloaded , the CDR from the carrier site for 26th June 2010 , they see 50% calls are NEVER dialed by Dialer but it appears in CDR. Amazingly, all the call durations are of 29-30 secs. When we checked the status of the same in Dialer, lead is present there but its marked as NEW which means Dialer has ever dialed

Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ?

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone

community there is a real possibility this may come off so if you have an interest in this space and want to contribute to the discussion then this is your opportunity to do so. =20 I look forward to all opnions on this topic. =20 The slide deck for the agenda of this call is located here http://voipusersconference.org/2008-05-09-Slides=20 Cheers, Dean=20 ________________________________

Hello, Does anyone know why I am unable to retrieve the "Redirecting Number"? I've done a "pri debug span 1/1" and can see the number being passed correctly to Asterisk: < Redirecting Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) < Ext: 0 Presentation: Presentation allowed of

I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent On 8/20/07, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hello, I am using an ARI dialer for my applications and since my last upgrade to Ver. 13.29.2 from 13.23.1 I am getting this strange bug from the ARI debugger: Debugging on all applications enabled <--- ARI request received from: x.x.x.x:63036 ---> HOST: x.x.x.x:8088 content-type: application/json authorization: Basic xxxx content-length: 265 body: { "context":