Author: cyberco

Categories
IPPBXs Products Services

Multi User Hosted PBX

Use the Internet to make calls – it’s simple, cost effective, and perfect for small businesses and call centres.

Voice over Internet telephony reduces telephony bills; connects mobile, remote and office workers; and gives a consolidated impression of your business – same number no matter where employees are located.

From 2 to 10 people wanting to stay in touch, Our feature-rich packages can offer a local or International number from which to operate – or you can bring your old number with you (number porting). There are also a wide range of add-ons that offer inclusive minutes. We can go through setting it up for you or you can do it yourself. To signup follow this link or call us on 01225580025

  • Multi User VoIP £8.00 per month
  • 4000 UK landline minutes £ 20.00 per month Lower amounts available
  • 4000 UK & International landline minutes £25.00 per month
  • 500 UK mobile minutes £30.00 per month

*Prices exclude VAT.

Key features

An online customer control panel allows you to manage your own account, and you can expand the system when ever you need to. All you need to get started is broadband, a router, and an adapter or a VoIP phone.

  • Make immediate savings: Free internal calls. Competitively priced calls and inclusive landline and mobile minutes package add-ons.
  • Quick and easy to set up: No difficult installations.
  • Excellent call quality: With no compromising on functionality.
  • Never be out of touch: Call forwarding available.
  • Keep your old number: Seamless transition with ‘number porting’.
  • Global presence: International numbers available.
  • Stay in control: Online customer administration, call logs and invoicing.
  • Voicemail and voicemail notification
  • Call forwarding to any number including mobiles
  • Online contacts directory, call logs and invoicing
  • Customised CLI (caller line identity)
  • Time of day routing

 

Features

  • Set up £4.99
  • Monthly £8.00
  • Included phone number UK and International*
  • Concurrent calls per number 2
  • Internal extensions 10
  • Call packages FREE VoIP-to-VoIP
  • 999 Emergency Services access YES
  • Minimum contract length 12 months
  • Voicemail YES
  • Voicemail notification SMS or Email
  • Call forwarding YES
  • Codecs supported G729a, G711u, G711a
  • Online call logs and invoicing YES
  • Online contact directory YES
  • Customised CLI (caller line ID) YES
  • Time of day routing YES
  • Audio call conferencing YES
  • IVR/Auto-attendant YES
  • Music on hold YES
  • Hunt call groups YES
  • 4000 UK landline minutes add-on £20.00 per month
  • 4000 UK & Int. min. add-on £25.00 per month
  • 500 UK Mobile minutes add-on £30.00 per month
  • Additional Number^^ £3.00 per month
  • Personal Number** £10 setup, £10 per month

* Surcharges apply for International numbering.

^ Subject to fair use check at 4000 minutes per month.

^^ Can only point at a Gradwell VoIP number on the same account, they can act as a mainline phone number however they must take an existing route. Please note that they cannot be used to increase concurrent calls.

** Can only point directly at an extension number. You cannot direct these numbers towards hunt groups, call queues or any other type of functionality. Does not provide an additional line.

All phone services (inc Unlimited packages) are subject to Terms and Conditions and standard call charges. All prices exclude VAT. Range of hardware and accessories available.

 

Categories
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