similar to: Asterisk@Home 0.7 released

I'm new to this and have tried to find the answer in the discussions and docs but to no avail. I even read the posting saying the password issue has already been discussed. So, at he risk of being exiled, here goes. Question 1: I've installed 0.7 and can log into the asterisk server from windows by typing http://192.168.1.11 I can log in with wwwadmin and the password I set myself

We have replaced the simple contact management system in Asterisk@Home with SugarCRM a full CRM system. This might seem like over kill for a home PBX but Sugar has the best contact management we have seen. With click to dial functionality and the ability to import data from other contact managers it’s a great fit for Asterisk@Home. We have also added new version of the usual Asterisk software AMP

This script was developed by Mark Johnson. All I did (Dean Collins) was type up the instructions and make it easy to understand. This Script will allow you to dial and extension number on your Asterisk@home V0.6 or later pabx and have it read you the weather for your city How it does this is; 1/ When you dial your extension it causes the weather.agi script to run 2/ This makes the program

Hi, I have been playing about with meetme as a conference bridge, and find it lacking in some features which I believe are out their somewhere. Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design it looks like a plan happened, but where is meetme2 at now? Things like recording a conference, allowing callers to adjust volume, allowing the conference to be locked, having

Hello I saw Ver1.2 is out. Whats new? Thanks for the hard work, David

I am looking at some directions on how to install and it is asking me to edit defines.php, it states that the file should be located in the source directory, but I can't seem to find it anywhere on my machine. Anyone been thru this? Jordan Novak Communications Technician -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm afraid I am at a loss. If the three files, app_cbmysql.c, app_meetme2.c and Makefile all exist in ../apps then a patch -p1 from the ../asterisk directory should work. The -p1 tells patch to ignore the first directory in the path to the file in the patch, -p2 ignores two directories. Another option is to just edit the apps-meetme-cbmysql.txt and split it into three patchs and apply

Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In

I am trying to compile the meetme2 application with the latest CVS head and it fails. Here is the error message that I get. Can someone point me in the right direction? gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS

Lots of interest here for conferencing. I've probably convinced more people to start using asterisk@home for this feature than anything else. Can I input some suggestions; Need to change the rinky dink call icons (let me know if you need some better samples) Need to change the ability for a muted caller to flag they have a question and a desire to talk (I do a lot of mute all except 1 or 2

Hi, Anyone built Asterisk in CentOS? Any luck on Zaptel module like wcfxo? I just could not build it. Thanks in advance. Bill

While researching Areski's new Web-MeetMe management gui, I found some odd (from what I expected) behaviour). Using the CLI to set un/mute status works but does not update the flags, or so it appears. Starting with a fresh conference (1 user) *CLI> meetme list 3456 User #: 1 Channel: OH323/R61 Using the CLI to mute the caller (no change in the user status0 *CLI> meetme mute 3456 1

Hello Team, I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i

I am running Feodra, I have downloaded the WebMeetMe Program, untar it to /var/www/html/WebMeetMe. I can access teh web page as of now. I cannot for the life of me figure out where defines.conf is. The install tells me it is in /var/www/html/WebMeetMe/lib/ however a complete search of the computer cannot find it anywhere. The /lib/ subdirectory does not exist in the untar'ed folder either. I

Feel free to hit me with a cluestick; maybe it will jar something lose... I'm running a stock *@home 0.6 install and after four days of excrutiatingly annoying debugging, I finally came across the obvious problem. The zaptel driver is reading /etc/zaptel.conf; which by default configures one FXS Kewlstart channel. Asterisk, according to "full" log, will see the first channel, then

I have started to receive a lot of positive response for the Asterisk@Home project. For those of you unfamiliar with this project the goal of Asterisk@Home is to make a full featured version of Asterisk very easy to install. We have created a 1 step .iso that installs RHEL (RedHat Enterprise Linux) and Asterisk. It includes a web GUI that allows easy editing of the Asterisk Config files.

Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I am trying to compile meetme2 in my asterisk box and getting the following compilaton error. Please help me to sort it out. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function)

http://www.srh.noaa.gov/fwd/productviewnation.php?pil=OKXZFPOKX&version= 0 can anyone suggest how I could set up asterisk@home to read out allowed the following text when I dial extension 850? 815 PM EST WED FEB 16 2005 .OVERNIGHT...MOSTLY CLEAR. LOWS 30 TO 35. NORTHWEST WINDS 15 TO 20 MPH WITH GUSTS UP TO 30 MPH...DIMINISHING TO 10 TO 15 MPH LATE. .THURSDAY...PARTLY CLOUDY. COOLER

Hello! Do somebody knows how to compile meetme2 with 1.0.6. I readed wiki, applied patches, but no luck ;-( Me be someone can give me working meetme2.c ? :-)