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FreePBX paging with ZAP or Dahdi

Posted: May 1st, 2012 | Author: | Filed under: Asterisk VoIP, Tech | Tags: , , , , , , , , , , , | No Comments »

This is fairly specific, but I didn’t find any posts out there to help me.
If you are using an older Freepbx with Asterisk 1.4 or 1.6 and you want to include a ZAP or DAHDI channel in a paging group, but it never rings the FXS channel, then you need to examine the dialplan.

I noticed that when it reached this point (- Executing [PAGE203@ext-paging:2] ChanIsAvail), it would hangup.
After tinkering I realized that it was checking on ZAP/4 (my channel in this case).

To fix:
-Go to the extension in Freepbx, and under the “Dial: ” options for that extension, change it from ZAP/4 to DAHDI/4.

Apply and now when you see it doing the Chanisavailable check it will pass and should call your FXS line.

Appointment Reminder Calls – With Asterisk VoIP

Posted: July 21st, 2010 | Author: | Filed under: Asterisk VoIP, Linux, Tech | Tags: , , , , , , , , | 8 Comments »

A while back I had a meeting with a friend in Orlando, FL. He came up with the idea to provide an appointment reminder service to companies that would benefit from such a service. Doctor’s offices, Dentists, Mechanics, Repairmen, etc. I drew up the logistics of the program and began working on it in the fall of 2008. Since then we have had customers use the software to easily call their customers and remind them of appointments and also use the software for call blasting. Call blasting is a feature to call an unlimited amount of numbers and playback a message. Since I have experience in the Asterisk VoIP platform I programmed the system to allow for options and the customer to enter in input with their keypad.

The purpose of this article is to detail the logistics and share it with the Asterisk community.

Basic overview:
There are a few different ways to originate calls with Asterisk.
-From the Asterisk CLI directly with the “originate” command
-From the AMI (Asterisk Manager Interface)
-From .call files placed in /var/spool/asterisk/outgoing

I went with .call files because of a few reasons:
-I can generate all of the call files ahead of time because they are read based on the timestamp of the file (less load)
-It is straightforward and easy to follow, and also leaves behind the .call file that is archived after the call is finished
-Asterisk will parse and execute based on timestamp, so to burst 100 calls would require much less overhead than through AMI/CLI

There are some negative effects to the .call files too:
-Changes made need to remove and add another .call file (If the customer changes their appointment time after the file was generated)
-The lowest interval to generate calls is every minute, so we can burst to 400 calls/minute (also more difficult to throttle calls)

Now you know how the calls are originating on the system, let’s move on to the database and interface.
I hired someone to program the interface in AJAX and PHP. That way it is secure and saved a lot of time if I were to attempt it myself. The interface is very clean and easy to use for customers. They also have the ability to import 1,000 records at a time via CSV files. Feel free to demo the interface via the link on for “Demo the Interface”.
The interface stores the call data in MySQL which is then scanned hourly to process and create all of the .call files according to their timestamp. We only care about the calls that will be made by the system in the next hour, so those are the only .call files that are generated. The script is a PHP script that connects to the database with the query of calls to be made, strips them down, then does a loop to create the .call file, set the timestamp based on the variable, then move the .call file to /var/spool/asterisk/outgoing.

The .call files include 1 custom variable that is comma-delimited so we can parse out the appointment time, AM/PM, and any other custom variables.
The other important lines in the .call file point to the context in the Asterisk dialplan that the customer’s announcement and options are defined.

Now you may ask yourself, It will be a pain to manually add customer’s options and dialplan for every sign up, right? It was at first, but I created several automated scripts to generate the dialplan for the customer and “dialplan reload” when complete.

The rest of the accounting/etc is accomplished via bash cleanup scripts that write to the database and provide reporting/etc.
The same system can be used for call-blasting to a large group of number for a survey, snow day announcement, marketing message.

Please let me know if you have any questions or advice/constructive-criticism for me regarding the program.
And feel free to try out the demo interface and hear a reminder call!


2010 – No More Ads…Resolutions

Posted: January 4th, 2010 | Author: | Filed under: Asterisk VoIP, Linux, Mac, Tech | Tags: , , , , | No Comments »

I decided to start off 2010 with a few changes:
1. Choose a new theme for the site…done

2. Remove Ads…done(With the exception of the sweet animation my buddy did)

3. I’m studying AGI scripting with Asterisk and want to share my findings along the way. I hope to have a lot of posts soon for you guys.

4. I hope to have bi-monthly updates to the site at a minimum. Topics include but are not limited to: Asterisk, AGI, PHP, Linux, and anything else I find interesting. 2010 is going to be the year of knowledge.

5. I put a link on the right column, and I’m only going to say it once. If you want to buy me a beer, or contribute to my escapades with a small donation. Then I will smile knowing that someone appreciated the info enough to help me out.

Safe1405 – Automated or Interactive Asterisk backup solution -with auto FTP upload

Posted: April 27th, 2009 | Author: | Filed under: Asterisk VoIP, Linux, Tech, Uncategorized | Tags: , , , , , , , , , , , , , , , , , , , , , , , , , | 6 Comments »


Safe1405 is a bash script that will backup the popular Asterisk directories and place them in a tar.gz file. The script can be used in 1 of 2 modes.
The first mode is interactive, with a menu to choose:

  • Backup Everything (/etc/asterisk, Voicemail, Recordings, Sounds)
  • Just VM
  • Just Recordings
  • Just Sounds

After the file is created, it will prompt to upload to a remote FTP server.

The second mode is silent, and is best suited for automated execution via cron or at.
Just enable the UNATTENDED and PUT_FTP variables by setting them to “1″.  You can directly edit the paths to each directory if yours are non-standard.

The script can be directly downloaded here: Safe1405 Download Link


# Safe1405
# Author: Gregg Hansen -
# Safely tar and gzip Asterisk files

# Version 1.0 20090427
#-Backs up all important Asterisk files - Tar/Gz
#-Choose the file name, or Date by default
#-GUI-like. Able to be silent for cron, or interactive

#EDIT the below ABSOLUTE paths to match your directory structure:

#Date Var
FILEDATE="$(date +%Y%m%d)"
#Your ServerName
#Filename is FILEDATE-SERVERNAME.tar.gz

#Enable unattended mode/Remote FTP put (useful for Cron):
# Backup of /etc/asterisk ONLY (default)
# 1 = On, 0 = Off

#FTP Credentials

###--START CODE---###

### Interactive Mode Functions ###
cat <<EOF

Safe1405 Backup

1) Everything
2) /etc/asterisk
3) Voicemail
4) Recordings
5) Sounds
Q) Quit


echo -n "Prompt> "
read INPUT
case $INPUT in
1) everything;;
2) etc-ast;;
3) vm;;
4) recordings;;
5) sounds;;
q) exit;;
Q) exit;;

tar cfvz $FILENAME $VM
tar cfvz $FILENAME $MON
echo -en "Upload $FILENAME to FTP? (y or n) "
read INPUT
if [ "$INPUT" == "y" ]; then

### Unattended Mode Functions ###
tar cfvz $FILEDATE-$SERVERNAME-ETC-AST.tar.gz $ETC_AST ##Uncomment##$VM #$MON #$VAR_LIB

ftp -ivn <<EOF
open $FTP_IP

#### Start Program Flow ###
#Unattended Function Call(s)
if [ "$UNATTENDED" == "1" ]; then
if [ "$PUT_FTP" == "1" ]; then
#Interactive Function Calls

###–END CODE—###
Leave a comment if you have questions/suggestions or would like help setting up the script.

FreePBX Round Robin Trunks Load-Balance/Distribute Calls

Posted: February 4th, 2009 | Author: | Filed under: Asterisk VoIP, Linux, Tech | Tags: , , , , , , , , , , , | 2 Comments »

I’ve seen several posts around the net from users that want to distribute the calls across multiple trunks with FreePBX.  The main reason being that they have a deal with a provider, and each line has 4,000 minutes for free, etc.  After looking around at the different methods to accomplish this, it become very clear what must happen.

If your trunks are FXO ports/Analog then you are in luck.   :)

Go to the FreePBX interface, click on Trunks.
-Add a trunk
-Put ‘r0′ for the Zap identifier (that’s R zero)
-Submit Changes
-Set the round robin Zap as your outbound route and enjoy!

Simple and clean, enjoy.

I’m in Japan…

Posted: November 13th, 2008 | Author: | Filed under: Asterisk VoIP, Linux, Mac, Tech | Tags: , , , , , , , , , , , , , , , , , , , , , , , , | 1 Comment »

I’ve been wanting to go to Japan for a while now, its just that I haven’t gotten any of my friends to really commit to going.  So about 6 months ago, I decided that I was going to go by myself.  After thinking about it, it really is the best way to visit Japan for the first time.  I can allow myself to be totally immersed without having someone from my country bring me back to ‘feeling American’ every 5 minutes.  I am really glad that I have this opportunity to visit this great country.  I’ve been here 5 days now, and I almost feel like I live here.

To share my experiences, I’ve documented everything possible (mostly food) so future prospective-visitors can learn from my time here.

Check for the daily journal, pictures, and video updates.