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Asterisk Support Elastix Support Knowledge Base Technical

Running a Macro on answer for Asterisk queues.

asteriskThe Asterisk Queue application has an option that will run a macro on answer, This can be very useful when integrating with CRM such as Capsule or call centre applications.

This option isnt included in freepbx, Though this can be hand coded it isn’t best to do this when using Elastix, AsteriskNoW or any other freepbx based system.

To add this option We have written a couple of patched versions of the relevant freepbx pages that can be downloaded here , You will also need to add a extra field to the mysql database as follows

  1. Log in to mysql:   mysql -u root -p
  2. Enter password
  3. mysql> use asterisk
  4. mysql> ALTER TABLE `queues_config` ADD `qmacro` VARCHAR( 255 ) NULL;
  5. mysql> describe queues_config;

You should now have something like this:- | qmacro | varchar(255) | YES | | NULL | | as the last line of the table.

Now download the tar file and unpack it. then copy the two files to the /var/www/html/admin/modules/queues directory.

On loading the queue page in freepbx you will now have the “Queue macro on answer” box

queuemacro

In this box you put the macro name you wish to run when a member answers a call.

For example:-

[macro-logit]
exten => s,1,Noop( capsule crm intergration ${crminfo} ${CALLERID(all)})
exten => s,n,Set(foo=${CURL(http://127.0.0.1/directory/capsual.php?strCallid=${crminfo})})
exten => s,n,Noop(${foo})
exten => s,n,Hangup()

This a simple dialplan that runs a php script to log calls to the capsule crm

capsual.php

<?php
$today = date(“F j, Y, g:i a”);
$duedate1 = date(“Y-m-d”);
$duedate2 = date(“H:i:s”);
$Token = ‘YOUR CAPSUAL API CODE’;
$number = $_GET[‘strCallid’];
$datetime = $today;
$duedate = “$duedate1″.”T”.”$duedate2″.”Z”;
echo $duedate;
$myxml=”<?xml version=”1.0″ encoding=”UTF-8″?>n
<task>n
<description>Call recieved from $number at $datetime. Please update and assign this task if required</description>n
<dueDateTime>$duedate</dueDateTime>n
<category>incoming call</category>n
</task>”;
// The URL to connect with (note the /api/ that’s needed and note it’s person rather than party)
// SEE: http://capsulecrm.com/help/page/api_gettingstarted/
$capsulepage = “https://youraccount.capsulecrm.com/api/task”;
echo $capsulepage;
echo $number;
// Initialise the session and return a cURL handle to pass to other cURL functions.
$ch = curl_init($capsulepage);
// set appropriate options NB these are the minimum necessary to achieve a post with a useful response
// …can and should add more in a real application such as
// timeout CURLOPT_CONNECTTIMEOUT
// and useragent CURLOPT_USERAGENT
$options = array(CURLOPT_USERPWD => “$Token:x”,
CURLOPT_HTTPHEADER => array(‘Content-Type: application/xml’),
CURLOPT_HEADER => true,
CURLOPT_RETURNTRANSFER => true,
CURLOPT_POST => true,
CURLOPT_POSTFIELDS => $myxml
);
curl_setopt_array($ch, $options);
// Do the POST and collect the response for future printing etc then close the session
$response = curl_exec($ch);
$responseInfo = curl_getinfo($ch);
curl_close($ch);
echo $responseInfo;
echo $response;
?>

Have fun

 

Categories
Knowledge Base Technical

Flushing your sendmail queue.

Whenever sendmail has to deliver mails to other hosts which cannot be reached at that time, the messages are kept in the queue and are marked as “Deferred: Connection timed out”. Although the other hosts could be reached again and you want to tell sendmail to flush the mail queue, the command

sendmail -q -v

does not really try to reconnect to these hosts and still assumes that the connection timed out. The reason is that the hoststatus is cached, per default for a period of 30 minutes. Using

sendmail -OTimeout.hoststatus=0m -q -v

you can re-run the mail queue and force sendmail to reconnect to the hosts. You may want to define an alias for that, say, ‘sendmail-flush-timeouts’.

You can set further options in /etc/sendmail.cf.

Categories
Blog

Zen like pondering about telephony and Asterisk

Categories
Knowledge Base

ECO DECT mode on your Gigaset phone

Find out how to reduce the transmission power of your Gigaset phones and learn the benefits of enabling ECO DECT mode.

 

Find more Tutorials here: http://www.gigaset.com/tutoriallibrary
Or visit Gigaset on Facebook: http://www.facebook.com/gigaset

Categories
Knowledge Base

How to transfer the directory of your Gigaset phone

With your Gigaset phone you can easily transfer your directory as a vCard to an internal device or via Bluetooth to an external device.

Find more Tutorials here: http://www.gigaset.com/tutoriallibrary
Or visit Them on Facebook: http://www.facebook.com/gigaset

 

Categories
Knowledge Base

ETHERACCESS LA210

The unit is referred to as the Network Termination Equipment (NTE) or RAD box.

It is used to ‘bond’ the multiple lines that are used on an EFM connection into one circuit. There are RJ45 Network ports on the rear that allow you to plug your own equipment such as routers in.

Front View

The front of the LA-210 includes LED lights that can indicate the state of the unit and the connection.

LA210_front

Rear view

The LA-210 has 4 RJ-45 network ports that the you can plug your own router into.

LA210_rear

ls120led LED STATUS


Categories
Knowledge Base Support

Mitel SNMP Alarm monitoring

As part of our ongoing improvements to our Alarm and fault monitoring service we are now pleased to be able to offer proactive monitoring of the Mitel 3300ICP snmp alarm output.3300
This monitoring is proactive, meaning we check the system at regular intervals from our Nagios platform and will raise alarms on power failing as well as all mitel snmp alarm levels.

mitel alarm example

The alarm can be emailed or txt’d to single or group of addresses.

All that is required is fixed external hostname or IP address and port 161 or another random port forwarded to port 161 so we can connect and the snmp configuration on the Mitel system to allow our systems IP address to connect.

If you are interested in this service the standard charge £25 per site per year for more details please email or call us.

Categories
Knowledge Base Technical

Skype for SIP name to DDI with Asterisk

When using Skype for SIP trunks with Asterisk a simple an neat way to enable DDI calling for the skype names is to use the “extension” option.
This means that the ‘To’ in in the sip header is set to what you set.

This can then be picked out with a simple little bit of dialplan

exten => 99051000000000,1,Set(CALLERID(num)=${CALLERID(name)})
exten => 99051000000000,2,Set(cNum=${SIP_HEADER(TO):5:6})
exten => 99051000000000,3,Noop(${cNum})
exten => 99051000000000,4,Goto(from-pstn,${cNum}|1)

In the above example we have 6 digit ddi numbers in the context from-pstn.

Setting up the Skype end is as simple as logging into your BCP and then the relevent profile and clicking on the calling tab

and setting as below

Image

This lets you now use one account and have all your BCP accounts have DDI calls directed at the PBX

Categories
Knowledge Base

Installing Asterisk 11 on Centos 6.3

asteriskThis is a short video tutorial on the installation of Asterisk 11, I have included the blog and video in one place for ease of viewing

First, you will want to be sure that your server OS is up to date.

yum update -y

Disable SELinux by changing “enforcing” to “disabled” in /etc/selinux/config. Use a text editor or copy and paste this command.

sed -i s/SELINUX=enforcing/SELINUX=disabled/g /etc/selinux/config

After you update and disable SELinux, you’ll need to reboot.

reboot

Next, you will want to resolve basic dependencies. (More information on Asterisk dependencies.)

yum install -y make wget openssl-devel ncurses-devel  newt-devel libxml2-devel kernel-devel gcc gcc-c++ sqlite-devel

Change into the /usr/src/ directory to store your source code.

cd /usr/src/

Download the source tarballs. These commands will get the current release of DAHDI 2.6, libpri 1.4 and Asterisk 11.

wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz
wget http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4-current.tar.gz
wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz

Extract the files from the tarballs.

tar zxvf dahdi-linux-complete*
tar zxvf libpri*
tar zxvf asterisk*

For the next set of commands it is important to follow the proper order: DAHDI first, then libpri, then Asterisk.

Install DAHDI.

cd /usr/src/dahdi-linux-complete*
make && make install && make config

Install libpri.

cd /usr/src/libpri*
make && make install

Change to the Asterisk directory.

cd /usr/src/asterisk*

In the next step, running the “configure” script will vary depending on whether your system is 32-bit or 64-bit. (Watch the video for more details.) When the menuselect command runs, select your options, then choose “Save and Exit” and the install will continue.

Use this command if you are installing Asterisk on 32bit CentOS.

./configure && make menuselect && make && make install

Use this command if you are installing Asterisk on 64bit CentOS.

./configure --libdir=/usr/lib64 && make menuselect && make && make install

Optional: If you ran into errors you will want to clean the install directory before recompiling.

make clean && make distclean

Once you have an error-free install, copy the sample files from the configs subdirectory into /etc/asterisk.

make samples

Then add the Asterisk start script to the /etc/init.d/ directory

make config

Start DAHDI.

service dahdi start

Start Asterisk.

service asterisk start

Connect to the Asterisk CLI.

asterisk -rvvv

And now you have Asterisk 11 running on CentOS 6!

Original Article written by Billy Chia @ digium

Categories
Handsets

Digium Handsets

The Only Phones Built Specifically For Asterisk

  • asteriskEasy provisioning from Asterisk or AsteriskNOW
  • Integrated with Asterisk voicemail, directory, parking, call recordings, call queues and more
  • Build custom phone apps with a simple JavaScript API

Digium’s family of IP Phones are the first on the market built specifically for use with Asterisk and Asterisk-based systems. All models include HD audio and plug-and-play deployment at a price that fits any budget. With multiple line appearances, context-aware soft keys, and advanced applications that integrate directly with Asterisk features, the Digium phones offer a better user experience than any other phone on the market.

Asterisk Phone Features

Smart Software

Access to information is the key to productivity in today’s business environment. The integrated applications that come standard with all Digium phones put critical information at your fingertips. With voicemail, call log, contacts, phone status, user presence, parking, call recording and call queue interface, the Digium phones provide simple, intuitive access to a wealth of information, saving valuable time.

 Simplified Provisioning

Standards-based IP phones have a reputation for being difficult to install and configure. Most systems require changes to network configurations or additional components to facilitate deployment. Digium phones support plug-and-play provisioning. Simply plug in the phone and it will automatically discover Asterisk systems on the network. Select the user you want to assign to the phone and the proper configuration is instantly loaded. For larger deployments you can pre-assign phones by tying a MAC address to an Asterisk user. It’s that simple.

Custom Applications

Most desktop phones come with a fixed feature-set that is determined exclusively by the manufacturer. Digium phones are different. All models include the Digium app engine, an innovative feature that makes it remarkably simple to build and deploy custom apps. All of the productivity apps that ship with a Digium Phone are written with the JavaScript API that is used by the app engine. A BETA version of the phone firmware with app development tools is available at phones.digium.com, along with documentation for developing your custom apps.

Getting Started With Digium Phones

Get AsteriskDigium phones will work with any version of Asterisk. However, we’ve added some compelling features that are only available today in Asterisk 11 or in special branches of Asterisk 1.8 (seeCertified Asterisk) and Asterisk 10 (the -digiumphones branch). To take advantage of simple provisioning, integrated applications and the app engine, you will need to use one of these versions

Call or email for sales enquiries