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.

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


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

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.


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.


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

Elastix Support Knowledge Base

Setting up extensions in Elastix

Elastix Asterisk IPPXThis is a short video on the setting up of extensions on the Elastix Asterisk based IPPX.


Elastix Support Software Releases

Elastix 2.4 Released have announced the release of 2.4 stable.elastix240_en

Key changes are:

Changes in Elastix Framework:

  • The instalation of the Elastix system now its much cleaner.
  • The Migration to Privileged Scripts its completed. Now, there its no need to use commands such as /bin/touch, /bin/chmod, etc.
  • We improve readability on blackmin theme.
  • Fixed readout of FreePBX database password.
  • The internal jQuery was updated to 1.8.3 .
  • Some minor bug fixes for the Elastix Framework.
  • Changes in Elastix Addons :
  • Correction for Postgresql repo in ARM architecture.
  • Some minor bug fixes for Elastix-Addons.

Changes in Elastix Firstboot :

  • Make an update of password in manager.conf more robust in the case it falls out of sync with elastix.conf file
  • The Cancel option that used to appear in the dialog_password was removed, because if someone pressed, it no allows to continue configuring passwords. Now only appears the Cancel option after the firstboot if its necesary to change the password already seted.
  • Some minor bug fixes for Elastix-Firstboot.

Changes in Elastix Email_Admin :

  • Change of files owners for more security in the web path. Creation of new helper scripts (s
  • pamconfig,remotesmtp,mailman_conig,relayconfig).
  • Was made changes in the module email_account in order to better interaction at moment to create a new email account.
  • Some minor bug fixes for Elastix-Email_Admin.

Changes in Elastix Fax :

  • NEW MODULE Fax Queue.
  • Now errors are displayed when the fax job failed to submit and do not ignore them.
  • Remove useless code that could potentially error out the module.
  • Implementation of fax job cancelation.

Changes in Elastix PBX :

  • Add support and features to following phones: Elastix LXP200, Yealink model SIP-T38G, VP530 model, Alcatel Temporis IP800, Escene 620, Fanvil C62, Damall D3310 and Grandstream model GXV280.
  • Modified the way of displaying Reasons for Status in module weak keys.
  • In module Control Planel was made changes in function showChannel in order to fix bugs in wich the call made through a sip trunk have not been displayed in control panel.Some minor bug fixes for Elastix-PBX.

Changes in Elastix Security:

  • The instalation of this module now its much cleaner.
  • Change of files owners for more security int he path web path.
  • Some bug fixes for Elastix-Security.

Changes in Elastix System :

  • Reimplementation of GUI backup and restore operations on top of backupengine.
  • Add options to active o inactive services when reboot system in Process Status Applet.
  • Some minor bug fixes for Elastix-System.
  • Centos version was updated to 5.9
  • Kernel version was updated to 2.6.18-348.1.1
  • FreePBX version was updated to 2.8.1-16
  • Rhino version was updated to 0.99.6-0.b2
  • Asterisk version was updated to 1.8.20
  • Dadhi version was updated 2.6.1-4
  • Amongst others…

For Product details on Elastix see Here


Knowledge Base

VoIP – Per Call Bandwidth

These protocol header assumptions are used for the calculations:

  • 40 bytes for IP (20 bytes) / User Datagram Protocol (UDP) (8 bytes) / Real-Time Transport Protocol (RTP) (12 bytes) headers.
  • Compressed Real-Time Protocol (cRTP) reduces the IP/UDP/RTP headers to 2or 4bytes (cRTP is not available over Ethernet).
  • 6 bytes for Multilink Point-to-Point Protocol (MP) or Frame Relay Forum (FRF).12 Layer 2 (L2) header.
  • 1 byte for the end-of-frame flag on MP and Frame Relay frames.
  • 18 bytes for Ethernet L2 headers, including 4 bytes of Frame Check Sequence (FCS) or Cyclic Redundancy Check (CRC).

Note: This table only contains calculations for the default voice payload

Codec Information Bandwidth Calculations
Codec & Bit Rate (Kbps) Codec Sample Size (Bytes) Codec Sample Interval (ms) Mean Opinion Score (MOS) Voice Payload Size (Bytes) Voice Payload Size (ms) Packets Per Second (PPS) Bandwidth MP or FRF.12 (Kbps) Bandwidth w/cRTP MP or FRF.12 (Kbps) Bandwidth Ethernet (Kbps)
G.711 (64 Kbps) 80 Bytes 10 ms 4.1 160 Bytes 20 ms 50 82.8 Kbps 67.6 Kbps 87.2 Kbps
G.729 (8 Kbps) 10 Bytes 10 ms 3.92 20 Bytes 20 ms 50 26.8 Kbps 11.6 Kbps 31.2 Kbps
G.723.1 (6.3 Kbps) 24 Bytes 30 ms 3.9 24 Bytes 30 ms 33.3 18.9 Kbps 8.8 Kbps 21.9 Kbps
G.723.1 (5.3 Kbps) 20 Bytes 30 ms 3.8 20 Bytes 30 ms 33.3 17.9 Kbps 7.7 Kbps 20.8 Kbps
G.726 (32 Kbps) 20 Bytes 5 ms 3.85 80 Bytes 20 ms 50 50.8 Kbps 35.6 Kbps 55.2 Kbps
G.726 (24 Kbps) 15 Bytes 5 ms 60 Bytes 20 ms 50 42.8 Kbps 27.6 Kbps 47.2 Kbps
G.728 (16 Kbps) 10 Bytes 5 ms 3.61 60 Bytes 30 ms 33.3 28.5 Kbps 18.4 Kbps 31.5 Kbps
G722_64k(64 Kbps) 80 Bytes 10 ms 4.13 160 Bytes 20 ms 50 82.8 Kbps 67.6Kbps 87.2 Kbps
ilbc_mode_20(15.2Kbps) 38 Bytes 20 ms NA 38 Bytes 20 ms 50 34.0Kbps 18.8 Kbps 38.4Kbps
ilbc_mode_30(13.33Kbps) 50 Bytes 30 ms NA 50 Bytes 30 ms 33.3 25.867 Kbps 15.73Kbps 28.8 Kbps

Explanation of Terms

Codec Bit Rate (Kbps) Based on the codec, this is the number of bits per second that need to be transmitted to deliver a voice call. (codec bit rate = codec sample size / codec sample interval).
Codec Sample Size (Bytes) Based on the codec, this is the number of bytes captured by the Digital Signal Processor (DSP) at each codec sample interval. For example, the G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval).
Codec Sample Interval (ms) This is the sample interval at which the codec operates. For example, the G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval).
MOS MOS is a system of grading the voice quality of telephone connections. With MOS, a wide range of listeners judge the quality of a voice sample on a scale of one (bad) to five (excellent). The scores are averaged to provide the MOS for the codec.
Voice Payload Size (Bytes) The voice payload size represents the number of bytes (or bits) that are filled into a packet. The voice payload size must be a multiple of the codec sample size. For example, G.729 packets can use 10, 20, 30, 40, 50, or 60 bytes of voice payload size.
Voice Payload Size (ms) The voice payload size can also be represented in terms of the codec samples. For example, a G.729 voice payload size of 20 ms (two 10 ms codec samples) represents a voice payload of 20 bytes [ (20 bytes * 8) / (20 ms) = 8 Kbps ]
PPS PPS represents the number of packets that need to be transmitted every second in order to deliver the codec bit rate. For example, for a G.729 call with voice payload size per packet of 20 bytes (160 bits), 50 packets need to be transmitted every second [50 pps = (8 Kbps) / (160 bits per packet) ]


Bandwidth Calculation Formulas

These calculations are used:

  • Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size)
  • PPS = (codec bit rate) / (voice payload size)
  • Bandwidth = total packet size * PPS
Blog Case Studies Knowledge Base

Simple CRM integration.

CRM’s have changed a lot since the days of ACT and Goldmine costing many thousands of pounds, for example a 3 user licence for Goldmine is $2700 and $430 a year, compare that to a SaaS solution, for example Capsule is $12 a month you can see why users are flocking to reasonably prices web solutions.
Capsule being web based is accessible anywhere you want on PC, Tablet or smartphone, it also integrates with other services such as LinkedIn, Twitter or if you wanted Facebook. It integrates with other SaaS products such as Kashflow and Mailchimp to name but a few allowing accounting direct mailing and contacts to all be linked.

capsule-logo-on-blackSo why did we suddenly show interest in these again. Well we were contacted by business who used another CRM package and wished to migrate to Capsule but wanted to be able to call direct from the application. This feature is not available at the moment with Capsule, But we have previously used a Firefox plugin called greasemonkey that can rewrite pages and convert numbers it recognises into clickable links. On investigation there is a similar product for Chrome called Tampermonkey and they are compatible scriptwise. With minor changes to our CyJax dialer script we got this working on elastix as well as a modified version for users of Gradwell‘s hosted VoIP solution. With this script now installed Capsule allows simple click dialing of contacts from the database.

On Asterisk we have automated the updating of the contacts history notes with details of the Call, The notes are linked to the Case or opportunity that you were working on if you dialled out from case notes. Also we can track incoming calls** so that they show up in Capsule, at the moment they show up as a task as you cannot create a History note with out an owner but this hopefully will change if the Capsule API is extended . This means no more trying to remember when you called someone its all there.

We have recently extended this to now log callback requests from callers who dont want to wait in a queue. This additional feature is important as it lets people keep track of the callback requests dialing them, assigning them to a contact and adding further detail.

** To Track incoming calls required changes to the freepbx module in Elastix so that a macro is called on answer, Details on setting this support and patching Elastix is available  here

The scripts are available for free download at you will see that these are only scripts for your browser, If you want the server script for your ippbx please contact us to discuss this.


Knowledge Base Support

24×7 Asterisk server monitoring with Nagios.

We offer an economical solution for end users and resellers to monitor their Asterisk and Linux servers.

Our platform monitors servers 24 hours a day 7 days a week. Hosted in a state of the art US based data centre with connections to major UK data centres and multiple connections to the internet.

We offer different levels of monitoring from simple uptime and email alerts to system load, disk space and channel usage with email and SMS notification. Web panel and firefox/Chrome plugin available to all levels to view system status.

The service is primarily aimed at Asterisk based IPPBX server but we can monitor other Linux based servers and Mitel systems as well. Our checks on Asterisk servers were customised by us to allow easy and secure deployment as we only require SSH access to make checks and this is secured by server keys. 

Nagios monitor screen


Service levels

Silver Level £10 setup – £2.50 per month £25.00 per year

  • Single Server, 4 services from list below & email alerts.
  • Ping test
  • SIP/IAX Peer availability
  • Asterisk channels
  • ISDN availability
  • Disk Space
  • System Load
  • Heartbeat Status
  • SIP/IAX2 registration status
  • Mitel SNMP Alarm status

Gold Level £10 setup per server – £5.00 per month £50.00 per year

  • Upto 2 Servers, 4 services per server, email and SMS alerts by subscription

In addition to the silver list:-

  • Asterisk Database status
  • Custom checks, (cost for design may be inured)

Additional options.

SMS alerts by arrangement, if using Gradwell Numbers and outbound we can integrate with the SMS API

Extra contact £5 setup

Extra server £10 setup £2.50 per month £25 per year

Extra service £5 setup £0.50 per month £5 per year

Partner options are available, Please contact us for details.  Pdf  download cymon 

FIrmware releases

Gigaset N300 IP, N300A IP, N510 IP PRO – Firmware update 12/2012 (version 075) released

New version of firmware released for N300 bases, Upgrade to this if on 072 firmware to fix instability issues

– Problem of instability, which occurred only very sporadically with version 72, and reset of base station after intensive usage solved

– Problem with call transfer of an external party to an external target behind Cisco Manager solved

– de telefoongids (Netherlands): online phonebook search is working again

– Security:

· Password is masked in VOIP Wizard, no longer visible in clear text

· PIN entry delayed if user repeatedly enters wrong PIN

– S68H handset: CLIP presentation is working again

– Blind Call Transfer problem solved with and

– URI dialling: Problem with added international/local area codes fixed

– Problem with consultation call and “Use Area Code Numbers for Calls via VoIP” setting fixed
FIrmware releases Knowledge Base

Yealink release V70 firmware for their T2X Sets

Yealink has announced the release of the latest Firmware V70 for its award winning IP phone SIP-T2X series.

The key feature of this new Firmware V70 is “M7”, also known as the “unified auto-provision template”. With Firmware V70, the configuration files and the deployment methods of T2X, T3X and VP530 have now been unified.

With the deployment of “M7”, end users now no longer need to maintain different templates of T2X, T3x or VP530. In other words, it lowers the learning curve and increases the business efficiency remarkably.

End users can easily convert their old templates of Yealink IP Phone T2X series and T3X series to “M7” through Yealink Configuration Conversion Tool (CCT). Firmware V70 is now available for download free of charge at

Download release notes here