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Hopefully it's okay to *announce* this here. I've been working on a project for Asterisk for some time and it is finally ready for a beta release. Any feedback is well appreciated. At the basic core it's a Dialer for Windows. I'll be adding more features quickly, but I wanted to keep everything simple and stable in this first release. Check it out at http://www.snapanumber.com

Hi all, New to the list and (mostly) new to CentOS. We have just moved from running RHEL to CentOS on our main production servers. Anyway, I am configuring a couple of new SunFire v20z servers here at the moment and trying to get the systems' management interfaces to work fully under CentOS. First issue was trying to install the openipmi RPM from the CD that came with the SunFires, this RPM

Setting up pick/callgroups really tightly is the way to go - one of our techs here does that all the time for clients. PaulH On Wed, 2007-06-20 at 20:51 +0930, Tom Lanyon wrote: > Hi list, > > Does anyone have any advice on the following: > > Incoming calls to our office come in on a SIP trunk. Since all our > offices/desks are in close proximity, we would like just a

Tom Lanyon <tom at netspot.com.au> wrote: > On 06/12/2007, at 8:20 AM, James A. Peltier wrote: >> > Tom Lanyon wrote: >> >>> >> Hi all, >>> >> Trying to install 5.0 or 5.1 (i386 or x86_64) on a new system from >>> >> an install DVD. As soon as the installer launches anaconda, the >>> >> screen shuts off

Hi list, The bug reporting guide in the wiki mentions we should notify the - devel list when reporting a new bug, so I just wanted to bring some attention to: http://bugzilla.xensource.com/bugzilla/show_bug.cgi?id=1328 I''d reported another bug myself (http://bugzilla.xensource.com/bugzilla/show_bug.cgi?id=1332 ) and marked it as a duplicate of the one above as they seem to be the

Hi list, Trying to upgrade someone's workstation here to 5.2 (was installed from a 5.0 DVD I think). The RPMs on our internal mirror are in-tact and pass a 'rpm -- checksig' test, yet when I run a 'yum upgrade' a large portion of them are corrupted and fail the GPG check. This seems to be isolated to yum, as downloading the RPM directly via FTP with wget or lftpget

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 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

Hi list, Does anyone have any advice on the following: Incoming calls to our office come in on a SIP trunk. Since all our offices/desks are in close proximity, we would like just a single phone to ring when a call comes in instead of ringing every person's phone. Currently we've got this working by having all the phones in a callgroup/pickupgroup and incoming calls ring the

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

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

Hello, Raw stats: Version:1.8.3.2 OS:Centos 5.6 Special setup: postgre database I am having a few queue issues with Asterisk specifically relating to breaking out from queues while on hold. The intent is that while someone is on hold they can press a key (lets say *) to break from the queue and go elsewhere (in this case to leave a message). However In all of my testing I am unable to get

Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: <PAGE_BEEP se.pat.ringer.13.name="Page Beep" se.pat.ringer.13.inst.1.type="chord" se.pat.ringer.13.inst.1.value="12"

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

callgroups & pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John

Working on a paging system for one of my sites and running into something I can't believe is this hard. In one of the zones, they want to have three different extensions ring over the pa system, using it as a loud ringer. Now the paging system does have a loud ringer built in and I can easily have it do a simultaneous ring, but all of the extensions will sound the same over the loud

Hi all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no

Hi all, at time i am trying to get a better idea of callgroups and pickupgroups (especially within the SIP Channel) A Pickupgroup is relative clear - everyone in the same pickupgroup may pickup a call And a callgroup does what ? - The same ? I thought that a callgroup would act like the ZAP groups - so that you then can dial SIP/g1 - and every SIP Client which is in the callgroup 1 does then