Tag: digium

Categories
Blog

Yealink T20P the new entry level IP phone

Yealink T20P is the entry level phone, of the Yealink VoIP desk phone range.

The Yealink T20P provides an entry level phone for the Yealink range of VoIP phones. The T20P boasts features beyond what you may expect from the lowest specification of the range. Ideal for use for the home or as a small office phone.

The T20P telephone offers power over ethernet (PoE), two SIP accounts, two line keys and a clear 2x 16 line LCD screen. A cost effective entry level enterprise IP phone with 2 lines.

Strong provisioning is in place for the Yealink range, making the models perfect for ITSP’s or large scale deployments.

Features

T20P-large

  • 2 SIP Accounts
  • 2 Line, 2×16 LCD Display
  • 2 Programmable Keys
  • 5 Hard Function Keys
  • Power over Ethernet (PoE)
  • 3 Way Conference Calls
  • Speakerphone
  • Call Hold, Waiting and Transfer.
  • Wall mountable
  • Compatible with a range of corded headsets:

 

RRP £69.99 +vat , (Contact us for volume and Special pricing)

Categories
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 

Categories
Gateways

Redfone FoneBRIDGE

The foneBRIDGE is a T1/E1 PRI-to-Ethernet Bridge. It is an integrated black box appliance designed to streamline installation and enable redundant design of open source telephony based VoIP systems such as Asterisk®, Elastix, Trixbox, FreeSwitch and others.

Features

Failover and HA Enabled
• Flexible Configuration
• Highly reliable design
• Simple Install
• Form factor independent
• Single, Dual, Quad and Octal Port models

Applications

• T1/E1 PRI Trunk termination
• Legacy PBX-to-Asterisk integration
• Simple, single server installs
• Complex, HA Asterisk clusters
• Channel Bank connectivity
• Mixed telephony environments (T1 and E1)
• Blade Servers where PCI slots are not available

Download Datasheet here

Price excluding VAT:

  • Non ec Single e1 £430
  • Non ec Dual e1 £850
  • Non ec Quad e1 £1200
  • ec Single e1 £640
  • ec Dual e1 £1200
  • ec Quad e1 £1700
*ec = Hardware echo cancelation
Categories
Gateways

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.

Applications

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:

Specifications

Interfaces / Connections

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

Benefits

  • Hardened
  • Cost effective
  • Low power consumption

Features

  • 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

Categories
Services

VoIP Design and Sales

At Cyber-cottage.co.uk 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 TableBook.me 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.

Categories
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

 

Categories
Case Studies

Elastix solution for a Bristol IT company

We were recently approached by a Bristol IT company to replace their ageing Avaya system. The proposed solution was an Elastix 2.3 solution running on Vmware 5.1, with their ISDN30e line connected via a Digium G100 gateway.

A key requirement was the ability to recharge usage to tenants in the building and replace an aging and expensive Oak call logger, This was simple with the Elastix solution as this option is included free of charge in the system and just required the uploading of a rates table.

The customer decided on Yealink T28 handsets for the office, utilising the BLF to have visibility of who is engaged on calls.

Categories
Case Studies

Outdoor pursuits company.

We were approached to supply  a hosted PBX solution to a Southwest England based Outdoor pursuits company replacing their on site AsteriskNow system with a more manageable and streamlined hosted Elastix based solution.

This gave them the ability to have handsets located where ever they were needed and add handsets easily with the user friendly Elastix interface. It also gave them increased viability of call usage with the inbuilt reporting and FOP2 let them see users engaged on calls.

We ported their existing BT Analogue lines to Gradwell VoIP trunks, This gave them an immediate increase in call capacity and combined with inclusive landline and mobile call packages meant call costs were kept under control.

Categories
Case Studies

Xorcom Solution for Kensington Office

An existing client was moving offices from Carnaby St London to Kensington, Since the original system was installed the usage had changed with the majority of staff now being based in South Africa where we also have remotely installed an Asterisk solution previously.

For the new kensington office it was decided to use a Xorcom IPBX as this single box solution would make management easier and as ISDN2 Lines were required the overall cost would be lower than using a dedicated server and ISDN2 gateway.

xr2000-analog-250

The system was preconfigured and tested in our Lab and then taken to site and installed in one day. The clean interface allows for easy addition of handsets using the endpoint manager.

Since the company make a large volume of international calls it was decided to use Gradwell for outgoing calls as this means a great saving over BT for call charges. They also had a EFM circuit installed for both Voice and office internet usage.

Categories
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.