similar to: Announcements via IAX phones

Howdy-- I'm playing with different IAX softphones. I've got DIAX and IAXPHONE on a windows (XP) machine on my network, and I'm running asterisk on a newly loaded Fedora Core 3 machine. I set up a separate IAX account for each phone. I was EXPECTING them to each register seperately with asterisk.... But I am swiftly finding out, that ONLY one registers. The first one to start

How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080722/f9612f97/attachment.htm

Hello, everyone-- I've just posted a package on the Asterisk wiki that, when installed, will allow you to announce incoming callers over the computer speakers, based on their CID. It's pretty simple, uses the linux /usr/bin/play (which on redhat, plays gsm files just fine over the speakers), and also uses festival to play the CID name string over the speaker, if you don't already

Hi list, I have a customer which will organize a conference in a big meeting room which has a sound system. He would like to connect this sound system to a MeetMe room so participant in the MeetMe can act as if they where on site. My idea is to take a barbone or Notebook, connect it to the sound system using the soundcard and run a softphone on it. Does some of you already have success in

I am trying to determine what would need to be done/modified to enable the following: I have a SIP extension come into my asterisk box, and I then need it to call "6-10" remote Sip Stations that are set to Auto-Answer... (note, my remote sip stations are actually cisco h323 devices, I can call them fine from any softphone, or other device, and have full-duplex audio, however, i need to

Hi Guys, I am a novice of Asterisk and I need some experts help to understand what I can get out of it. I need to make multiple calls (let say 50) at once to autoanswering softphones on a LAN and send all of them the same message that they will repeat with loudspeakers in the same environment. I am a little concerned about synchronization of the phones and moreover it is not much clear to me if I

Hello-- How do I twiddle the sound drivers? I'm not that experienced with kernel twiddling and driver loading. I have Redhat 9. My previous attempts to play with the kernel and load extra drivers always ended with a new kernel that wouldn't boot. I know asterisk doesn't like to play with sound interfaces that aren't full duplex. So, when I built the system, I saw the MSI

Hi there I am using IAX2 softphones dialing into a meetme conference. In my softphone I was forcing uses to click on a button when they wanted to speak, enabling their microphone and disabling their speakers. This way when a user was speaking they did not hear their voice half a second later (because meetme mixes the voice and sends to everyone in the conference). Now because of requirements

Ladies, Gentlemen We are experiencing an unusual problem in our asterisk 1.4.34.. We are attempting to determine if channels are in use before paging to them. This works correctly, as in it pages the phone.. however, we see the error message below on the console... after googling, we discovered limited information regarding the issue... -- Executing [NPANXX7298 at from-pstn:1]

Does anyone have any recommendations for overhead paging systems for use with Asterisk? Thanks, Randy Johnson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/82ae09ec/attachment.htm

I think I've forgotten something obvious.... I've got 2 incoming calls, I want to bridge them - how can I do this ? (assume I somehow know which calls should be paired up...) I could dump them both in a meetme - but that seems wasteful as i _know_ there will only ever be 2 parties. (And I need DTMF to flow through). I may want to record the bridged call, but that isn't vital.

This is just a note that the fixes in the CDRfix4 and CDRfix6 branches are getting closer to being merged into 1.4, trunk, and 1.6.x. If CDR's are important to you, and you ignore this notice, then you deserve what you get! These branches address various long-standing bugs, most of which are regressions from 1.2. It is hoped that these fixes will solve most of the problems introduced by the

Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on "Accept" button. Greg -------------- next part -------------- An

I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Thanks for any help you could give.

I'm afraid I already know the answer because I've done a lot of searching, but does anyone know of a softphone that supports a central phone book and paging (like the sip autoanswer option of some hardphones) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

> Has anyone had any luck in changing the voices for Festival and > Asterisk? > > I have Festival installed and working, but can not get the voice > different > from the default. > > Thanks, > > Jason > Jason-- Assuming you follow the installation instructions, and install the Mbrola and other goodies for all the possible different voices, then you can,

Hello, We use MS Access 2000 (I know, we're migrating away from it) as an application to automatically dial phone numbers. The old phone system we have allowed the call representative using the application to take their phone off hook, push a button in the app, and the app would send the phone number to the phone system and dial the number. We are moving to Asterisk for our main phone

Call to Action! For those who have the courage/ability, go grab an SVN copy of the asterisk release, the HEAD version, and my latest patch, from: http://bugs.digium.com/view.php?id=6021 Right now, the latest version of the patch is 0.10. apply it to the SVN head version, and do a "make". Read the Wiki on AEL2: http://www.voip-info.org/wiki/view/Asterisk+AEL2 Look at the examples

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people