Categories
Asterisk Support Blog Design FreePBX Knowledge Base Software

G.729 Goes Royalty Free

G.729 – IMPORTANT INFORMATION

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.

 

Categories
Knowledge Base

Resetting the Polycom Soundpoints admin password to default

If the default Polycom password of 456 does not work, or if someone has changed the admin password on the phone, please do the following:

  1. Find and write down the MAC address (serial number) of the phone you want to reset. It should be twelve characters, and look something like ‘0004F2ABCDEF’.  If you can’t read the back label, you can find the MAC address by pressing Menu, Status, Network, Ethernet.
  2. Power down the phone.
  3. Power up the phone.
  4. While powering up the phone (you have about 6-8 seconds to complete this step):
    • For SoundPoint IP 320, 321, 330. 331, 335, 430, and 450 press and hold the 1, 3, 5, and 7 on the dial pad at the same time.
    • For SoundPoint IP 301, 501, 550, 600, 601, and 650 press and hold the 4, 6, 8, * on the dial pad at the same time.
  5. After holding down the numbers for few second, you will be prompted to enter the admin password.  Enter the MAC address of the phone. No colons and the alpha characters must be entered as lowercase letters
  6. The Set will restart. You may need to restart again to get access to the menus with 456 password.

 

Categories
IPPBXs Software

FreePBX

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 powappliances-headerer 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.

 

 

Categories
Blog Knowledge Base

Planning for a Successful VoIP deployment

Before you deploy voice-over-IP or a Hosted PBX service in your office there are a few considerations you must first address.  Switching from traditional telephone service to voice-over-IP (VoIP) requires sufficient bandwidth, a proper switch and router, and a good battery backup solution to protect you from power failures.

The key voice-over-IP requirements discussed in this article are:

Bandwidth – Determining how much bandwidth you will need for voice-over-IP in your office is your first step.

The Router – Choosing a low quality or under performing router is a costly mistake which will degrade your call quality.

Quality of Service – You must decide whether voice traffic will be separated from regular internet users or if it will share the same network.

VoIP Equipment – There are many digital office phones, soft phones, headsets and telephone adapters on the market to choose from.

Power Failures – Voice over IP does not work when the power goes out so you should install a battery backup system and possibly a Power-over-Ethernet switch if your budget permits it.

How much bandwidth do I need?
Voice over IP needs a certain amount of bandwidth in order to keep your conversations clear and free of disruptions.  Bandwidth is the amount of information which your internet connection can send and receive in a certain period of time.  Your first step should be to use an online speed test to find out what your maximum upload stream and download stream is.  We suggest you do this test using a fixed connection to the internet rather than using your wifi (wireless) connection to get accurate results.  Try to use numerous tests during different times of the day to get a good average of what you can expect from your internet connection.  Bandwidth is normally measured in kbps or kilobits per second.
You will need to have a high speed (broadband) connection to use voice-over-IP.  A typical DSL connection will be rated at 600 kbps for the upload stream and 5000 kbps on the download stream.  You will notice that your upload stream is almost always smaller than your download stream which becomes your limiting factor for using VoIP service.
Your next step is to determine how many people in your office are likely going to be using the phone at the same time.  For instance, having ten people on the phone will require ten times as much bandwidth as having one person on the phone.  Below is a chart which will help you calculate how many people can be on the phone at one time:
Ask your voice-over-IP service provider what audio codecs they offer as there is a trade off between audio quality and bandwidth usage…

Full Quality Audio (G711 Codec)\- Uses 87 kbps for each concurrent phone call (NEB)
Compressed Audio (G729 Codec)\- Uses 33 kbps for each concurrent phone call (NEB)

So the calculation for a typical DSL connection would be:

DSL connection:600 kbps upload / 5000 kbps download
Gives us (Full Quality):600 kbps / 87 kbps = 6 concurrent calls
Gives us (Compressed Quality):600 kbps / 33 kbps = 18 concurrent calls

Notice we used the upload bandwidth in our calculation as this is the limiting factor for voice-over-IP.  You also don’t want to push your connection to the limit as most cable and DSL connections do not have guarantees in terms of how much bandwidth they will deliver.  If you Internet connection drops in bandwidth at some point during the day you don’t want your call quality to be affected.  Other factors affecting voice-over-IP are the latency of your connection and how much packet loss there is on it.

Choosing a router
A router is the device that connects all your computers and network equipment to your Internet connection.  It is an often overlooked piece of the puzzle that can have a major impact on the success or failure of your voice-over-IP implementation.  There are many routers on the market, some are very cheap (less than $40) and others can cost you thousands of dollars.  There is nothing worse than putting a poor quality or underpowered router in your office which could cause an otherwise good VoIP installation to go bad.
Your router needs to be powerful enough to handle the number of phones you will have in your office and should also work flawlessly with voice-over-IP equipment.  A good place to start when deciding on your router is to speak with your voice-over-IP service provider. We also recommend checking to make sure that your router is compatible with voice-over-IP services.
The following is a list items which will help you to determine whether your router is right for voice-over-IP:
How many voice-over-IP phones will you be connecting to the router? The more phones you will be connecting, the more powerful the router needs to be. Don’t use a £40 router to run an office with 10 IP Telephones.
Will your voice-over-IP phones have their own dedicated Internet connection? If not, a router with a quality of service (QoS) setting to prioritize voice traffic over regular traffic is an absolute must. Without QoS you will encounter poor quality telephone calls regularly.
What other functions will the router need to perform? You might need your router to handle VPN connections, allow wifi (wireless) connections or perform other tasks.
Make sure you can bridge your router to your modem. Routers that are not bridged can cause problems with voice-over-IP installations.
Never use more than one router or nat gateway on the network at a time as this will cause problems for IP Telephones when they attempt to do NAT.
Make sure your router is compatible.
It is always best to get a recommendation from your voice-over-IP service provider as some routers are known to perform very poorly with VoIP phones.

Quality of service
Call quality is a function of your network and the public internet. Some delays and network congestion cannot be avoided due to information traveling over the public internet while other types can be avoided. Good network design is critical to a stable and reliable voice-over-IP implementation.
Quality of service (QoS) refers to the ability for your router to prioritize voice traffic (VoIP) differently than regular internet traffic on your network or the separation of voice traffic.  Voice over ip is a real-time protocol which means that if information is lost or delayed it will result in a noticeable drop in call quality or a complete loss of it. Symptoms of network congestion include garbled audio, dropped calls and echo.   When setting up voice-over-IP in your office there are three possible ways handle voice traffic. Some customers report perfectly good results without any quality of service (especially in a small 1-2 person office) and others report worse results with quality of service enabled on their router as some routers do a poor job of implementing this. Generally speaking however the best way to deliver reliable voice-over-IP service is through a dedicated internet connection that is only used by the voice-over-IP equipment rather than sharing the internet with computers. Below are the different methods of doing quality of service:

No QoS – Voice traffic and regular internet traffic in your office are sharing the same internet connection.  No prioritization of voice traffic over regular traffic is being performed and thus there is the high potential that voice quality could be degraded if there is insufficient bandwidth for both voice and regular traffic. Some customers experience very few problems using this method while others report a high frequency of poor quality calls, dropped calls and garbled voices. It all depends on how much network congestion your office has. Most internet connections are more likely to be upload bound which generally results in people not being able to hear you, because all of your upload bandwidth is being consumed by something on your network.

Router enabled QoS – Voice traffic and regular internet traffic in your office are sharing the same internet connection, but your router is able to distinguish between voice traffic and regular internet traffic and give the voice traffic a higher priority.  The problem with this method is that routers can only prioritize upload bandwidth which means your voice will be clear but the router cannot ensure that download bandwidth will be prioritized. If employees on your network are downloading often this will cause a noticeable drop in call quality but this method is better than no quality of service. Some internet providers can prioritize the download bandwidth using TOS or COS methods from their end which will create an end to end quality of service solution. Most customers find that even prioritising upload bandwidth for voice-over-IP offers a dramatic improvement in call quality because most internet connections are limited by their upload bandwidth and have lots of download bandwidth free.

Separated Traffic – Voice traffic and regular internet traffic are separated onto two different internet connections and networks. This is especially critical for larger offices with 5 or more employees.  Voice traffic is carried on one internet connection and data from computers is carried on the other connection. In this case no prioritization is required by your router because voice traffic has its own dedicated internet connection.  This is the best way to ensure clear voice communications and the method we generally recommend customers whenever possible.

The method you decide on largely depends on how much bandwidth you have, what you are using your internet connection for besides voice-over-IP and the level of call quality desired.  Many offices report perfectly good results without using any QoS, while others find that it makes a major difference in the quality of their calls.

Choosing VoIP phones and equipment
Before deploying voice-over-IP in your office you will need to decide how each employee will be connected to your voice-over-IP provider.  There are many choices on the market today.
Digital IP Telephones – These types of phones look just like regular multi-line business telephones except that they connect directly to your internet connection using a network cable.
Soft Phones – A soft phone is a software program running on your computer that looks and feels just like a real telephone.  This requires you to purchase a USB headset which connects to your desktop or laptop so you can make and receive calls.
Wifi Phones – A wifi phone looks and feels very much like a regular cell phone except that it connects to your wireless router in the office.
Analog Telephone Adapters (ATA) – An ATA is a small box which connects to your router and allows you to plug in regular analog telephones so they can work with voice-over-IP.  ATAs are generally low cost alternatives to digital office phones and are easy to take with you when you travel.
Battery backup and Power-over-Ethernet
With voice-over-IP and most office telephone systems you must consider what happens when the power goes out.  For some offices this can be a regular occurrence and for others it might happen with a very low frequency.  Once of the things you will need to decide is whether or not you will install a battery backup system.
Here are a few important terms your should know:
Power over Ethernet (PoE) – Is a technology that allows VoIP over ip telephones to be powered using regular network cables rather than power adapters which plug into the wall.  This has the advantage that you can power all the phones in your office from a single source and makes installing a battery backup unit much easier.
Uninterruptible Power Supply (UPS) – Is a device that powers your equipment when you lose power at the office.  The system has a built in battery which keeps your network devices operational when the power goes out.
The easiest way to protect your phone system from a power outage is to power all the phones using a Power-over-Ethernet switch that would normally be connected in the back room where your router and cable/DSL modem is located.  This has the advantage that all your phones are drawing power from a single source which you can backup using an uninterruptible power supply (UPS).  All you need to do is plug in your PoE switch, router, and DSL/cable modem into a sufficiently powerful UPS device so that when the power goes out all your phones remain up and running.

Categories
Blog

A new patent troll.

SIP Trunking

In the last few weeks a large sleeping troll has come out of hibernation and seems set on disrupting the whole voip market.

Quote

“BT is engaged in licensing an extensive range of standards related patents that address the key features of SIP Trunking providers and VOIP operators providers.

BT’s Patents address a wide range of fundamental capabilities now in widespread deployment, such as:

  • Setting up a call
  • Breaking out to other networks
  • Managing resources efficiently
  • Registering terminal to a network
  • Cost effective call completion
  • Monitoring and alerting of IP call quality”

Well that pretty much covers all of the workings of a SIP network. A full list of the patents is here  .

But its not Just BT, AT&T also have claims over SIP as well see here for a list.

It seems that some of the major patent holders see more money in the licencing of the now ubiquitous SIP protocol than maybe supplying it to customer. Which is a shame as the only ones who will make any money will be the Lawyers in the end.

More to follow on this I’m sure….

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
Products

Handsets,Gateways and Cards

We supply VoIP Hardware, Phone Systems, IP Phones & VoIP Equipment from all Major Manufacturers.

Aastra Polycom Snom
Digium Openvox Patton
Grandstream Vegastream Redfone
Citel Linksys  We also supply

YeaLink

Gigaset

Please contact us with

your requirements


Please contact us for pricing