Digium G100/G200

Built on a powerful combination of the Asterisk Open Source communications engine and a state-of- the-art embedded platform, Digium VoIP Gateways provide the best value for Asterisk connectivity.

The gateway software is based on the Asterisk communications engine and is managed through Digium’s intuitive point-and-click GUI interface, which allows for easy navigation and effortless setup. VoIP gateways feature a power-saving embedded design with a highly efficient digital signal processor (DSP) handling all media-related operations.

The Digium G100 VoIP Gateway includes a single software-selectable T1/E1/PRI interface and supports up to 30 concurrent calls. It is built to support TDM-to-SIP, SIP-to-TDM and SIP-to-SIP (transcoding) applications. In a TDM-to-SIP deployment the VoIP gateway significantly reduces operating costs by connecting a legacy business phone system with dynamic SIP trunking services. SIP-to-TDM deployments use the VoIP gateway to connect a modern SIP communications system with T1/E1/PRI service from legacy carriers.


Digium VoIP Gateways are flexible solutions that fit a variety of communications applications. The applications listed below represent some of the most widely used, today. The flexible configurtation options and standards-based connectivity mean Digium’s gateway appliances can support a wide range of custom applications.

Public Switched Telephone Network T1/E1/PRI to VoIP:

VoIP Provider to Legacy PBX:


Interfaces / Connections

  • 1/2 T1/E1/PRI w/ RJ-45
  • 1 10/100/1000 Ethernet


  • Hardened
  • Cost effective
  • Low power consumption


  • Intelligent call routing
  • Easy-to-navigate GUI
  • Fax and modem support
  • Solid state (no moving parts)
  • Remote configuration and software updates
  • Octasic™ DSP processor
  • Up to 60 concurrent calls

Price excluding VAT : G100 £820.00 G200 £1370.00

Knowledge Base

Gigaset Dect test mode

Setting test mode on the Gigaset handsets can be very useful. Detailed here is how to set a handset into test mode and what the numbers then mean. Once set go off on a walk round your site to find dead spots. Then change the base position to get the best coverage




The above screen Shows RX power at 100% , Frequency is 3 , TimeSlot is 02, Basestation code is 78 and finally Bit error rate is 100% (This means 100% Good not 100% error rate)

A short document on Setting test mode on Gigaset dect handsets for site surveys is available for download here. This shows you how to enable it and what each of the numbers mean.


VoIP Design and Sales

At we provide support,design and installation services. We have over 25 years of experience of the telecommunications industry and have the depth of knowledge to assist you in all aspects of telecommunications needs.

We have been working with VoIP systems since 1999, and VoIP networks from the Mid 1990s everything from small offices of 15 extensions to large multi-site networks with bespoke platforms. Our primary deployments are now based on the Asterisk open source platform from Digium.

Solutions have included:-

  • High capacity conference servers.
  • High availability redundant servers for emergency services dispatch.
  • Click2Call solutions
  • Call Centres
  • Office PABX systems

Asterisk is a complete telecommunications platform. From caller ID to multi-site networks, anything your telephone system can do, Asterisk can do better and maybe cheaper.

It includes a whole host of telephony features such as CTI, Voicemail, call conferencing and CRM integration.

We have tailored our Asterisk solution to behave like a normal PBX, with call barring, day and night service, call re-routing, DND, voice mail for all users and new features can be added easily at any time.

With Asterisk we can replace your PBX or complement an existing PBX by adding more functionality at a competitive price.

Recent systems have included a large hosted callback platform for a Major UK Car Parking company allowing drivers to make calls to the office at no charge to themselves.A system for a “online” Solicitors group to allow the tracking of calls and work-flow. We have recently deployed a system for to allow them to take table reservations for restaurants.

Recently customers have included Mendip Outdoor Pursuits, Purple CarParks, NorthCott Global Solutions and Qwtanet. These have been a mixture of onsite systems, hosted systems and solutions based on Asterisk running in a VMware environment.

Call or email us to discuss your requirements.

Case Studies

Restaurant Booking Solution.

We have been working with a client on a Hosted restaurant booking solution, providing the CTI and call tracking systems. This was complicated by the simple fact that the booking system was a closed system by another supplier.

The system works by the restaurant diverting their line to a DDI number we provide that sends the call to the system with the destination matching a defined number for that restaurant when the call enters the system the relevant settings are looked up in a database and audio message file, IVR options and CallerID name are set and the call is passed to the IVR. The caller then chooses their preferred option, The call can be depending of the time of day be passed to the call centre for a booking to be taken on the restaurants behalf or the call is passed to the restaurant where in many a Hosted Gigaset Dect handset is provided for them to take the booking or call the call centre for free.

The system has changed and evolved over time and by using Asterisk has meant that we can accommodate most requests for changes, Most recently we updated the statistics package to Asternic Pro

Asternic stats
Pro stats

Statistics. This has allowed much more detailed reports to be created for queues and agents.

The calls are delivered to the platform over a EFM circuit from providing quality and reliability combined with increased capacity over the original ISDN30 circuit.

For disaster recovery we provide a backup system in a data-centre that is kept in sync with the office system so in the case of power outage or system failure calls can be diverted to this system and calls take on mobile phones. Switching to back-up system is completed by the single click of a button on a web-page that instigates the diversion of the lines and starts the backup system automatically.

Currently we are migrating the database services off to a separate VMware server with 3 VMs, one for each of the core web or mysql servers. This will allow the service to scale as there are now over 1 million records per datatbase and it is showing no sign of slowing.

Knowledge Base

Digium G100/200 Gateways and UK CallerID Number

The current firmware in the Digium G series gateways have a quirk that if they don’t receive caller ID name they move the caller Id number to be the Caller Id name but don’t leave the Caller Id number in place. The relies on you setting  “trustrpid=yes” in teh sip trunk configuration.

We have produced a short document on settings for using the gateway with any freePBX based asterisk solution. It can be downloaded here



Snom Handsets

We Supply the full Snom range of handsets and feel that they provide both solid reliable business handsets to  touch-screen handsets that integrate with door-entry systems and much more.



For full details of handset features click on the handsets image




Gigaset Handsets

All the Gigaset DECT bases and handsets we sell are compatible with each other, but some handsets may only offer basic or limited compatability find out more information contact us



For full details of handset features click on the handsets image


Knowledge Base

Trusting Linux servers

This hopes to explain in simple steps setting up a pair (or more) servers as a trusted group.
So what do we want to achieve ? Well we wnat to be able to ssh, sftp, rsync etc between servers and not need to enter passwords
Steps required
1 Hosts File
2 Editing sshd_config
3 Create the ssh keys
4 Setting up the Auth. users file
Hosts File

Firstly we need to make sure all servers are in the hosts file
# Do not remove the following line, or various programs
# that require network functionality will fail. localhost asterisk2.local
# We point to eth0 on our own box asterisk2.local asterisk2
# We point to eth1 on the other box asterisk1

Editing sshd_config

Now we need to edit the /etc/ssh/sshd_config file
so that the following

RSAAuthentication yes
PubkeyAuthentication yes
AuthorizedKeysFile /root/.ssh/authorized_keys


#RSAAuthentication yes
#PubkeyAuthentication yes
#AuthorizedKeysFile .ssh/authorized_keys

Now restart the sshd
/etc/init.d/sshd restart

Create the ssh keys

We now need to create the keys on each server
ssh-keygen -t rsa
and hit return for all the questions.
this will create 2 files in /root/.ssh

go the /root/.ssh directory and copy the to the other server and get its

sftp asterisk1


Setting up the Auth. users file

In the /root/.ssh directory you will now have for example :- id_rsa known_hosts

We now need to copy the to the authorized_keys file

cat >> authorized_keys

Do the same on the other server.

You should now be able to ssh and rsync between servers.


Xorcom Astribank

Astribank is a versatile and powerful channel bank that was specifically designed for the Asterisk IP-PBX. Astribank supports all the common telephony lines and trunks: FXS, FXO, BRI, E1/T1 PRI, T1 CAS and E1 R2. The Astribank driver is a part of the standard Asterisk distribution.


Knowledge Base

Sip debugging with wireshark

Wireshark and Cloudshark are invaluable tools for debugging sip and iax issues on your Asterisk server.

Here we have a short Video that goes over the basics of getting a call captured and opened in Cloudshark

we also have a short tutorial for download here in PDF format

First we need to get the packets we want. This is far simpler than its thought. We use a simple command line tool called tcpdump, if its not installed install it now, You wont be able to live without it.

Here we have 2 commands, The first captures packets on interface eth0, -n means we won’t convert addresses, -w means we just capture raw packets and udp means its only the udp packets we want and finally port 5060 means its only the sip messaging we want. In the second we dont specify port 5060 so that we get the rtp stream as well.

/usr/sbin/tcpdump -n -i eth0 -w /tmp/wireshark.pcap -s2000 udp port 5060
 /usr/sbin/tcpdump -n -i eth0 -w /tmp/wireshark.pcap -s2000 udp
screen -S "udpDump" -dm tcpdump -n -i eth0 -C 9 -W 15 -w /var/log/asterisk/dumpsip.pcap -s2000 udp port 5060

The command above will write to file in the background and will rotate at 9 meg so suitable for cloudshark

Once you have started the capture and made a call as required you will get a file called for example /tmp/wireshark.pcap copy this to your workstation via ftp or sftp as you would copy any file.