Zulu UC Desktop and softphone integration unifies the most popular business communication tools & applications enhancing user productivity and mobility. Designed specifically for FreePBX and PBXact phone systems, Zulu enables features such as:
Zulu Softphone enabling users to make/receive phone calls from their desktop or mobile stations, including Chat for team collaboration.
Faxing directly from the Zulu widget & softphone.
Click-to-call to make calls directly from your web browser and /or email client.
With Click-to-Call integration, users can instantly call any phone number that is seen on their web browser or MS Outlook client which a click!. Simply click on the phone number and Zulu will initiate an outbound call via the softphone client or your desk phone, whichever is with you at the time. Great for mobile users who come and go from their workstation.
Click-to-Call also recognizes extensions and phone number prefixes, so you never have to worry about having to modify the phone number or extension you wish to click to dial.
Call Pop
Ideal for CRM and Help Desk Integration, Call Pop will automatically open your desktop web browser on an inbound call with all the information of the caller. This feature helps users expedite phone calls and provide the caller with the best customer service experience.
* For additional CRM integration check out our CRM Link Module
Presence
Improve communication between staff members by allowing them to see each other’s presence via the Zulu Softphone. This feature can save your employees time by reaching out to members who they know are available to take their request.
Users can set their presence using a variety of pre-set statuses or create their own. And because Presence is server side, a user’s presence will be updated across all communication endpoints automatically too.
Zulu Softphone
At the Center of the Zulu is the all new softphone enabling users to take their office with them and never miss a call.
Make and receive phone calls using Desktop
Send and receive FAX*
Control Presence status which will update your status on all your devices
Flexible Calling Options- generate a phone call from either the client on your desktop or your desk phone. Great for mobile users who come and go from their workstation.
The Zulu UC Softphone features integrated Chat functionality so that staff members can communicate with each other more effectively. Features like 1-to-1 messaging, group chat, file transfer and auto-archiving will improve employee collaboration and improve business results. Finally, one tool to do it all!
*Compatibility
Operating System: Zulu UC is compatible with Windows, Mac and Linux operating Systems. Browsers: Click-to-call and Screen POP work with Firefox and Chrome (Safari coming soon).
The Most Resilient VoIP Digital Gateways in Their Class
Vega VoIP digital gateways are small appliances that seamlessly connect your legacy telephony infrastructure, made up of PRI (T1, E1) or BRI lines, to IP networks. They are great for businesses with legacy phone equipment (such as a TDM PBX) who want to connect to SIP trunking services without having to spend money altering their current network infrastructure. They are also great for businesses that are already VoIP enabled at the core (with an IP-PBX) that need PSTN connectivity and require a SIP-to-TDM converter. Simply place the Vega VoIP Digital Gateway at the edge of your network, plug in your existing internet cable for VoIP connectivity and E1,T1 or BRI cables from your phone system and let the Vega VoIP Digital Gateway automatically handle the SIP signalling and voice media conversion for seamless voice and T.38 Fax integration.
Advanced Web GUI
Features an intuitive Quick Wizard which does all the hard work for you for new deployments. Flexible dialplan to allow you to make your own routes, including automatic failure detection with failover routing.
Diagnostic Tools
Web GUI based PCAP tracing tool to capture full signaling and media, eliminating the need to connect equipment for line tracing, fully compatible with wireshark.
Low and High Density Models
The Vega 100G and Vega 200G are our low density models with a maximum capacity for 30 and 60 SIP-TDM simultaneous calls. The Vega 400G is our high density model and the most flexible field upgradable unit for a maximum capacity of 120 simultaneous SIP-TDM calls.
E1/T1 & BRI Interface
Each E1/T1 interface (for Vega 100G, 200G, 400G) and BRI interface (Vega 50 BRI) can be independently configured as network side or terminal side. The Vega gateway can therefore be connected to a PBX or the PSTN.
Built-in Local Survivability
In the event of a WAN failure, IP phones behind the Vega gateway can continue to call each other, be routed to a backup switch or connected directly to the PSTN.
Vega VoIP Digital Gateway Models
Vega VoIP Digital Gateways are one of the most reliable fault tolerant SIP-to-TDM media Gateways on the market, sized for your business needs. All Sangoma hardware carries a one year warranty with options to extend.
Vega 50 BRI
Sangoma’s Vega 50 BRI VoIP Digital Gateways are a 2-4-8 port BRI appliance for up to 16 simultaneous BRI calls
Web GUI for configuration and troubleshooting
Featuring Quick Wizard for rapid d
eployment
Onboard DSP for media translation
Interoperable with most legacy and VoIP carriers worldwide
Advanced flexible call routing with automatic failover and bypass routing
Built in Local Suitability in the case of WAN failure
Vega 100G
Sangoma’s Vega 100G VoIP Digital Gateways are a single port T1/E1/PRI appliance supporting up to 30 simultaneous calls.
Web GUI for configuration and troubleshooting
Featuring Quick Wizard for rapid deployment
Onboard DSP for media translation
Interoperable with most legacy and VoIP carriers worldwide
Advanced flexible call routing with automatic failover and bypass routing
Built in Local Suitability in the case of WAN failure
Vega 200G
Sangoma’s Vega 200G VoIP Digital Gateways are a dual port T1/E1/PRI appliance supporting up to 60 simultaneous calls.
Web GUI for configuration and troubleshooting
Featuring Quick Wizard for rapid deployment
Onboard DSP for media translation
Interoperable with most legacy and VoIP carriers worldwide
Advanced flexible call routing with automatic failover and bypass routing
Built in Local Suitability in the case of WAN failure
Vega 400G
Sangoma’s Vega 400G VoIP Digital Gateways are a quad port T1/E1/PRI supporting up to 120 simultaneous calls.
Web GUI for configuration and troubleshooting
Field upgradable licensing
Dedicated bypass ports for High availability
Support for Private Wire or Point-to-Point applications
Onboard DSP for media translation
Interoperable with most legacy and VoIP carriers worldwide
Advanced flexible call routing with automatic failover and bypass routing
2N offer a range of stylish solutions for door communication. The 2N® Helios IP door and security intercoms will ensure comfort for you and your visitors and with a range of different models and feature enhancing accessories there will be an option to suit your needs.
Due to the complexity and multiple configurations possible please call or email for pricing and options.
As of January 1, 2017 the patent terms of most Licensed Patents under the G.729 Consortium have expired.
With regard to the unexpired Licensed Copyrights and Licensed Patents of the G.729 Consortium Patent License Agreement, the Licensors of the G.729 Consortium, namely Orange SA, Nippon Telegraph and Telephone Corporation and Université de Sherbrooke (“Licensors”) have agreed to license the same under the existing terms on a royalty-free basis starting January 1, 2017.
For current Licensees of the G.729 Consortium Patent License Agreement, no reports and no payments will be due for Licensed Products Sold or otherwise distributed as of January 1, 2017.
For other companies selling G.729 compliant products and that are not current Licensees of the G.729 Consortium, there is no need to execute a G.729 Consortium Patent License Agreement since Licensors have agreed to license the unexpired Licensed Copyrights and Licensed Patents of the G.729 Consortium Patent License Agreement under the existing terms on a royalty-free basis starting January 1, 2017.
As soon as we hear how this is going to affect Digium Asterisk we will update here.
Leo D’Alessandro, Product Marketing Manager at Sangoma, and Frederic Dickey, VP of Product Management at Sangoma, will in this webinar explain how to build an efficient contact center cost-effectively with Sangoma’s FreePBX / PBXact UC.
In this webinar, you’ll learn how the many ways FreePBX / PBXact UC can solve your contact center requirements:
• How calls are best routed using call queues
• Maximizing Agent Productivity and Customer Satisfaction with automated Queue Callbacks
• Integration with desktop and CRM
• Monitoring live call metrics
• Reporting tools to analyze overall performance
Introducing the FreePBX appliance! The FreePBX appliance is a purpose-built, high-performance PBX solution. Designed and rigorously tested for optimal performance, this is the only officially supported hardware solution for FreePBX. The appliance comes preloaded with the FreePBX Distro and includes a one-year warranty!
Featuring the FreePBX Distro, this appliance is an ideal fit for businesses looking to get more from a PBX. With millions of deployments throughout the world, FreePBX is relied upon daily by everyone from enterprises to startups. Leveraging the power of FreePBX has enabled businesses to grow while keeping communication expenses minimal. The FreePBX Distro has made deploying, configuring and using a PBX system easier than ever! With an easy-to-use GUI (Graphical User Interface), getting started is a breeze!
With over 1 MILLION production systems worldwide and 20,000 new systems installed monthly, the FreePBX community continues to out-perform the industry’s commercial efforts. The FreePBX EcoSystem has developed over the past decade to be the most widely deployed open source PBX platform in use across the world. The openness of the project allows users, resellers, enthusiasts and Partners to utilize the FreePBX EcoSystem to build robust communications solutions that are powerful but at the same time easy to implement and support. Sangoma is proud to be the sponsor of FreePBX project. If you are new to FreePBX you can get started quickly by downloading and installing the FreePBX Distro. The FreePBX Distro is an all in one platform that installs everything you need to build a phone system. Once You have a basic PBX in place you can add commercial modules to add advanced features to an already feature rich base install of FreePBX.
As an open source GPL, web-based PBX solution, FreePBX is easy to customize and adapt to your changing needs. FreePBX can run in the cloud or on-site, and is currently being used to manage communications of all sizes and types of environments from small one person SOHO (Small Home, Small Office) businesses, to multi-location corporations and call centers. The FreePBX ecosystem provides you with the freedom and flexibility to custom design business communications around your needs.
FreePBX Commercial Modules are add-ons that enhance the already feature rich base install of FreePBX! These modules are not Open Source GPL and are only designed to work with CentOS or RHEL systems. The FreePBX Distro is already preconfigured to work with these modules. For custom installations please see: Install Commercial Modules on CentOS and RHEL based systems
The FreePBX appliance is a purpose-built, high-performance PBX solution. Designed and rigorously tested for optimal performance, this is the only officially supported hardware solution for FreePBX. The appliance comes preloaded with the FreePBX Distro and includes a one-year warranty!
Featuring the FreePBX Distro, this appliance is an ideal fit for businesses looking to get more from a PBX. With millions of deployments throughout the world, FreePBX is relied upon daily by everyone from enterprises to startups. Leveraging the power of FreePBX has enabled businesses to grow while keeping communication expenses minimal. The FreePBX Distro has made deploying, configuring and using a PBX system easier than ever! With an easy-to-use GUI (Graphical User Interface), getting started is a breeze!
Sangoma IP Phones Designed Exclusively for FreePBX are Designed to work with FreePBX, Sangoma IP phones are so smart you can quickly and easily use them right out of the box. Each phone in the series features industry standard Power over Ethernet, so no power cable or outlets required. They have full duplex speakerphones, dual Ethernet Ports, multi-way conference calling, high definition voice quality, and they’re Virtual Private Network (VPN) capable.
Full Integration with FreePBX, FreePBX phone apps are available right on the phone, straight out of the box with no requirement for additional licenses. Users can control complicated features directly from their phones. There’s no need to remember feature codes. User applications include: Call Parking, Follow Me, Do Not Disturb, Conference Rooms, Call Forwarding, Time Conditions, Presence, Queues, Transfer to Voice Mail, Visual Voice Mail, and Log in/out.
Why is Sangoma Zero Touch Better? VoIP telephones can be complex to install, and manually configuring many different parameters and hundreds of extensions can take hours. When you buy and install your Sangoma IP phones, the redirection server automatically points the phone to the Sangoma FreePBX for configuration. Other vendors have redirection servers, but they have to be programmed with the details of the IP PBX. Only Sangoma can provide Zero Touch provisioning with FreePBX.
EndPoint Manager Included When using a Sangoma phone, EndPoint Manager software inside FreePBX is automatically enabled. This lets your users control global settings, program their phone keys, map extensions, upload images, download new firmware, and much more.
Zulu UC is a unified communications desktop program that interfaces with FreePBX and PBXact. It provides call notifications, SMS, faxing capabilities and much more! Supported applications include Microsoft Outlook, Mozilla Firefox and Google Chrome.
User connections are available in 20-seat packages, so get yours today by visiting the Store tab in your FreePBX Portal!
Click to Call
Zulu UC enables you to click-to-call within email, contact lists and websites. The Click to Call feature is currently supported for Microsoft Outlook 2010 or newer, Google Mozilla Firefox and Google Chrome.
URL Pop
Call pop settings allow you to have the PBX inform Zulu to open a web browser on a per inbound route or queue level in your PBX GUI. This is ideal integration for CRM and Help Desk systems.
Faxing
If you have the Fax Pro module and Faxing enabled, you can send faxes from both Outlook and the Zulu UC Widget in addition to UCP. You can also view received faxes the same way you view emails under your Outlook Inbox, where you then save the attached fax wherever you’d like.
Browser Plug-ins
Zulu UC is currently supported for Mozilla Firefox and Google Chrome on Windows machines. Make sure to install the Zulu Firefox Plugin or the Zulu Chrome Plugin before you start using Zulu UC in your browser. Support for Mac OSX is coming soon.
From the 1st September 2015 Yealink units purchased from us as an official Yealink (UK) reseller, will carry a Lifetime Warranty.
This Life Time Warranty will commence from the date of purchase; until 12 months after the official end of life date of that product.
This Lifetime Warranty excludes the following products:
– T18P, T20P, T20PN, T22P, T22PN, T26P, T26PN, T28P, T28PN, T32G, T32GN, T38G, T38GN, EXP38, EXP39, YHS32 and all VP series units.
– All Spare Parts including but not limited to : Network Cable (CAT5), Base Stand, Handset, Handset Curly Cord, Power Supply Unit (PSU)
The products listed above will be covered by the current 12 months from date of purchase or 24 months from date of manufacture, whichever is the longer.
Any extended warranty purchased prior to 1st September 2015 will continue to be honoured until the end of the purchased warranty period.
This warranty covers manufacturing faults and component defects. It does not cover wear and tear, physical damage, or problems associated with the network or platform it is connected to.
Please contact the distributor or ITSP the unit was purchased from for further details.
In the Event of a Failure
In the unlikely event that you have a faulty unit, you should contact us first.
We may require some additional information or additional testing and procedures to be carried out to establish the unit is faulty and one which they have supplied.
If the unit is deemed to be faulty we will request the unit is returned to us at your cost and subject to further tests, confirming the unit is faulty and that the warranty seal hasn’t been broken, a replacement unit will be supplied at no cost.
If the model returned under the Lifetime Warranty is no longer available we will supply an equivalent or better model.
This warranty is in addition to any statutory rights.
The SIP2SIM service is a very simple concept which provides you with control of your mobile communications. It is ideal for an office of any size and even for more technical home users.
The service consists of a SIM card, which you put in a mobile phone and it makes that phone appear as if it is a SIP extension (e.g. SIP phone) on a phone system of your choice.
An extension on your office phone system
The basic service allows you to specify, on our control pages, the server name, login and password for a SIP server. This could be your office phone system whether an asterisk box, or a FireBrick or whatever. As long as it handles normal UDP SIP with G.711 a-law audio then we will register as a phone and allow calls both ways.
This means your phone can simply be an office extension, like any other.
Call office extensions using short extension numbers from your mobile phone.
Office policies on callable numbers, such as premium rate, enforced like any other extension.
Office voicemail system working, just like any other extension.
Office call logging, just like any other extension.
Office call recording, just like any other extension.
Use in hunt groups, just like any other extension.
Even use features like call steal to transfer calls if you want, or in-band DTMF to control call transfer and related features.
Manage costs
The costs are very simple for using the SIM in the UK on O2. Higher costs apply for roaming SIMs, even roaming to other UK networks.
£5+VAT to purchase the SIM card.
£2+VAT pcm once activated (activates on first use).
2p+VAT per minute for calls either way.
2p+VAT per text either way.
2p+VAT per MB for data usage either way.
No minimum term – just monthly rolling contract.
Calls are charged per second, and data per kilobyte. Charges apply to the SIP2SIM service separately from any charges you may have for text or voice calls using your VoIP/SMS provider.
In some cases a SIM will be set up with a free trial which allows some usage without assigning to an account or setting up an account. For these trial SIMs, once assigned to an account an activation fee of 1p+VAT is charged and usage charges commence as normal.
Telephone numbers
Just like a SIP handset, the service does not come with any sort of telephone number.
You can, if you wish, have the phone register on a VoIP provider’s service. This would mean you get calls to a number operated by the VoIP provider, and can make calls from that number, just like any other SIP phone. If registered with your own telephone system, it would have internal extension numbering, and even direct dial in numbering as you have chosen to configure on your phone system.
There is no question of porting numbers to or from the service, it has no numbers.
We can, of course, provide telephone numbers as part of our VoIP service if you wish, and we can even pre-configure these to connect with your SIM as part of the order process.
International roaming
The SIMs are available with world wide roaming. Costs are higher when roaming, obviously. The following are charges for use within EU. See full roaming price list for more details.
£5+VAT to purchase the SIM card.
£2+VAT pcm once activated (activates on first use).
10p+VAT per minute for calls either way.
5p+VAT per text either way.
10p+VAT per MB for data usage either way.
No minimum term – just monthly rolling contract.
In the UK and rest of EU, calls are charged per second, and data per kilobyte. Charges apply to the SIP2SIM service separately from any charges you may have for text or voice calls using your VoIP/SMS provider.
UK roaming for the best coverage of the UK
The SIMs can even roam in the UK. This means the SIM can change its identity automatically – being an O2 SIM when O2 is available (at the lowest costs), but switching to a foreign identity (Dutch Vodafone) to roam on to other UK networks. Whist costs are then higher this means you stay in touch even when there is no O2 signal.
When the SIM is using the O2 profile and on O2 in the UK, the UK prices apply. When using the EU profile on any UK network (including O2) or elsewhere in the EU, the EU roaming prices (as above) apply. If on EU profile on O2, it will normally switch back to O2 profile automatically within a few minutes. Currently the SIM will not roam to Three in the UK.
Text
Texts are operated separately. Our text interface using HTTP can send texts to the phone, and texts from the phone can be posted to an HTTP gateway of your choice. If you have an A&A VoIP telephone number then texts can be associated with that number directly (not all of our numbers ranges can handle inbound texts).
Mobile Data
Data currently allows simple NAT, unfiltered, Internet access. We hope to offer data via A&A in the future.
Third Party SIP services
The service involves entering SIP registration details in to our control pages. Where these are the details of your own SIP server such as an office phone system, you can make the decision as to whether or not you trust us with those SIP details in order to provide the SIP2SIM service. We will, of course, use all reasonable skill and care to ensure the details remain confidential and are not disclosed.
We have, however, designed the service so that it can work with a wide variety of third party SIP services, not just in the UK but in various countries. There are a lot of unusual systems out there and we continue to work to ensure that such services operate with SIP2SIM. However, using third party SIP details may well be in breach of your terms with the third party SIP provider as it means giving us your SIP details. It is up to you to check the terms and we would not suggest anyone breaks a contract they have. You may find that the provider is happy to trust us, especially if they do any other work with us, so it is worth asking. We are also happy to discuss contracts with other providers for the SIP2SIM service and we may be able to provide them with branded SIMs to sell to their customers.
Unfortunately we cannot guarantee that the service will always work with all other providers, and whilst we aim to resolve any technical issues (with reference to the standards), if a provider simply will not deal with our service and blocks us there is little we can do.
The Sip2Sim service is provided by Andrews and Arnold Ltd and can be purchased direct from them or via ourselves where we will assist with setting up the service for you.
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Sangoma’s Vega 50 BRI VoIP Digital Gateways are a 2-4-8 port BRI appliance for up to 16 simultaneous BRI calls
Sangoma’s Vega 100G VoIP Digital Gateways are a single port T1/E1/PRI appliance supporting up to 30 simultaneous calls.
Sangoma’s Vega 400G VoIP Digital Gateways are a quad port T1/E1/PRI supporting up to 120 simultaneous calls.
er of FreePBX has enabled businesses to grow while keeping communication expenses minimal. The FreePBX Distro has made deploying, configuring and using a PBX system easier than ever! With an easy-to-use GUI (Graphical User Interface), getting started is a breeze!
