Categories
Handsets Special Offers

Gigaset Maxwell C DECT Desk Phone

The DECT cordless desktop phone in a Maxwell housing offers great flexibility from the desk and until January 2020 we are offering this unique handset for £82.00 Plus VAt and delivery or an N510 Base and Handset for the special price of £125 + Vat, Bundle pricing is also available for example An N510 Base and 4 handsets will cost only  £375 + vat. You can download the Datasheet here.

In an office dominated by wired desk phones, the Maxwell C is one of a kind. This is Gigaset’s most advanced professional cordless phone in a Maxwell housing.

The wireless office of today has shed the need for physical network connections. Our computers and office phones are all cordless – and the Gigaset Maxwell C leaves you with supreme flexibility from the desk.

All devices can now be connected to DECT single N510 or Multicell system from Gigaset. This brings the ideal IP phone solution with extraordinary HD-audio, crystal clear TFT-display delivering an intuitive business companion to the front of the wireless office.

The Maxwell C’s stylish design and flexible mounting options make it the perfect phone for anything from the office to the home; the hospitality environment to the warehouse and the garage.

Ready for use with all Gigaset’s professional base stations including Multicell systems to deliver complete coverage from any desk, meeting-room or office environment.

Categories
Knowledge Base Sangoma

Building FreePBX CallCenters

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

WEBINAR: Building Your Contact Center with FreePBX / PBXact UC from Sangoma on Vimeo.

Categories
FreePBX Software

FreeHMS , A Web based call management package for small Hotels and Serviced offices

FreeHMS is a web based call management package for small Hotels, Guest Houses and managed offices. It is designed to work with FreePBX and Asterisk.

It allows owners to bill guests or users for the phone usage allowing guests to make calls, setup wakeup calls and access voicemail. Rooms are initially blocked from calling other rooms but can call Admin extensions with out being checked in. When a room is checked in they can make trunk calls and set up wakeup calls. When checking out any wakeup calls are removed and the voicemail is defaulted and all Voicemail messages are deleted.

Setting up the system is simple for Installer with minimal changes to the dialplan which are included in the custom configuration file. The system can be set to any language as all text is from a single configuration file which also includes currency and tax rate for billing. Users are created in FreePBX user admin so are easily changed and added.

Call rates are set using the rates page only available to the Admin users

Administration is simple and password controlled using the ‘User Managment’ module of Freepbx so changes to rates and rooms can only be carried out by the admin users, Reception users can log guests in and out, Create Bills and mark rooms for cleaning as well as set or cancel wakeup calls, The Housekeeping login only allows setting of rooms clean or not*. If a room is not marked as clean then that room cannot be checked in.

The software is fully web based and can be used on PC, Mac, Tablet or even smartphone.

iphonehms

The software is OSS and can be customised to suit most customers.

Features Include:

Checkin /Checkout

Billing : Rates are set by the admin user only, Bill can printed with relevant sales tax added.

Cleaning : Rooms are marked unclean on checkout and can only be checked in when marked as clean by reception or the cleaning staff. A cleaning list can be printed off for stall without a tablet or smartphone.

The software is here to download  and as its released as OSS you can modify and extend it as you wish

For more details and pricing for complete system please email info@cyber-cottage.co.uk or call us.

If you just want the software its free to download,  Limited email support will be available, All we ask is if you add a feature or make a change let us have it so everyone can benefit from it.

Finally if there is a feature you want let us know how we can work with you to make it come about.

Categories
Blog

Gigaset Pro Bundle Offer

gigsetpromo

Until further notice you can get a free N510IP PRO base when ordering three Gigaset pro handsets. To get this deal order one of the following bundles, If you want more handsets get in contact and we will put a budget beating price together for you.

Buy 3 Get the Base Free

 

  • N510IP PRO base and 3 x R650H PRO handsets
    • Normal RRP Price £322+vat     Our Price £190+vat
  • N510IP PRO base and 3 x S650H PRO handsets
    • Normal RRP Price £277+vat     Our Price £160+vat
  • N510IP PRO base and 3 x SL750H PRO handsets
    • Normal RRP Price £367+vat    Our Price £250+vat
  • All orders £10 UK p&p for NI please confirm p&p

Call 01225 580025 or email us to place an order.

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
Peripherals Products

Algo 8128 SIP/VoIP Strobe light

Algo

The Algo 8128 SIP Strobe light is the ideal solution for visual ringing in such areas as noisy factories, cafeterias, and public areas.

Or alternatively it can also be used as a silent visual alert where loud ringing may be disruptive in areas such as hospitals, theatres and, churches etc.

Other applications include emergency and security notification where the press of a single phone key can be used to activate one or many strobe lights. And it can also be integrated within a Call Centre system to provide visual notification when queues and waiting times are exceeding there maximum thresholds

Algo 8128 SIP Strobe Light Key Features

  • 360° Visibility
    Flash patterns are visible in every direction or may be chosen specifically for ceiling and wall mount applications.
  • PoE SIP Endpoint with Web Interface
    Integrates easily into a VoIP Unified Communications environment, hosted or premise PBX.
  • Auto-Multicast
    Trigger one – trigger many. Multiple strobes may be operated simultaneously and synchronously using just one SIP extension.
  • Colourful Options
    Available blue, red, and amber caps to distinguish events in the workplace.
  • LEDs for High Intensity and Long Life
    The 8 brilliant LEDs splash 198 candela light in all directions with greater efficiency than xenon strobes.

Please call or email for pricing and avalibility

Categories
Blog Calls and Lines

Revised call charges.

Gradwell has moved to a new carrier for outgoing phone services, giving us the opportunity to offer a simpler and clearer pricing structure for our call charges.

In summary, the current standard rate charges remain the same as before; however, new levels of detail have now been added to help give clarity regarding how call tariffs are being charged.

The table below outlines the standard rate tariffs currently in place and now includes the new ones that have been added for UK calls.

Gradwell’s Standard UK Call Charges (pence per minute):    

Call TypePeak*Off-peakWeekend
Geographic (01, 02, 03)1.251.001.00
Mobile**9.008.006.00
Local rate (0845)**3.003.003.00
National rate (0870)**1.251.001.00
National rate (0871)**12.0012.0012.00
Directory Enquiriesdependent on the route, use the telephone lookup tool to establish exact cost
NEW   
Isle of Man6.004.004.00
Geographic Islands (Jersey/Guernsey)3.002.002.00
Premium Mobile
07520, 07744, 07755, 078222, 074416, 07777, 078228
20.0020.0020.0

Bundled minutes.

PPM Number of MinutesPrice
1p 500 £5.00
0.8p1,000£8.00
0.6p2,500£15.00
0.5p5,000£25.00
0.4p10,000£40.00
0.35p25,000£87.50
0.3p50,000£150.00
0.25p100,000£250.00
UK Landline Minutes
Number of MinutesPrice
250 £20.00
500 £30.00
1,000£55.00
2,000£100.00
4,000 £160.00
10,000 £350.00
20,000 £600.50
40,000 £1,200.00
60,000 £1,500.00
80,000 £2,000.00
100,000 £2,500.00
Mobile Minute Bundles

*Peak: 8am-6pm UK London time.

**Calls to numbers in these ranges start at the specified prices. Charges to some numbers will vary. To find the exact cost of any number we encourage you to use the telephone lookup tool as featured on the Gradwell call charges webpage.

Key Information Regarding Call Charges:    

•    Calls between your VoIP-enabled offices are free
•    Call to anyone else on Gradwell VoIP are free
•    0845 numbers have a 3p per call charge in addition to the pence per minute rate
•    0871 numbers for Services Fixed Fee FF15, FF28 and FF29 are charged at a flat 12p per call only
•    Calls are billed by the second, with a minimum call charge of 1.5 pence on all calls except Freephone calls
•    Inbound calls to 0800 numbers cost 3 pence per minute
•    Calls that are diverted to landlines are charged at standard rates

Charges for Directory Enquiry services will vary dependent on the route used and may be subject to a per call charge and / or a pence per minute rate. There are a large number of Directory Enquiry routes from which to choose, each with specific charges that can be viewed using the telephone lookup tool. Listed below are the most popular route charges per minute:

RoutePeakOff-peakWeekendPer Call Charge
1181185p5p£6.505p
(£3.85 weekend)
118226£1.00£1.00£1.0055p
118365£1.50£1.50£1.5045p
118212£5.50£5.50£5.50£3.20
118500£6.50£6.50£6.50£1.05


International Call Charges:
    

International call charges to the 34 headline countries have been simplified, not only are the landline calls currently set at the low cost of 1.5 pence per minute, but the mobile charges to these countries have now been set at a flat 11 pence per minute. The qualifying 34 headline countries are listed below:

ArgentinaChinaHungaryMalaysiaSouth Korea
AustraliaCzech RepublicIndiaNetherlandsSpain
AustriaDenmarkIrelandNew ZealandSweden
BelgiumFranceIsraelNorwaySwitzerland
BrazilGermanyItalyPolandTaiwan
CanadaGreeceJapanPortugalUnited States
ChileHong KongLuxembourgSingapore 

Please note that Estonia and Finland are no longer part of the headline country group and  Brazil and India have been added.

In addition to the headline countries listed above, Call charging rates to other International destinations have been simplified and now offer lower cost calls for both landline and mobile:

CountryLandlineMobileCountryLandlineMobile
Afghanistan£0.30£0.30Macedonia£0.20£0.60
Albania£0.20£0.50Madagascar£1.00£1.00
Algeria£0.10£0.50Malawi£0.25£0.45
American Samoa£0.05 Maldives£1.50£1.50
Andorra£0.05£0.25Mali£0.45£0.45
Angola£0.10£0.15Malta£0.05£0.15
Anguilla£0.15£0.35Marshall Islands£0.35 
Antarctica£7.00 Martinique£0.05£0.15
Antigua & Barbuda£0.25£0.45Mauritania£0.70£0.80
Armenia£0.15£0.30Mauritius£0.15£0.15
Aruba£0.15£0.35Mayotte£0.30£0.15
Ascension£3.00 Mexico£0.05£0.15
Azerbaijan£0.30£0.50Micronesia£0.35 
Bahamas£0.15£0.15Moldova£0.35£0.35
Bahrain£0.10£0.15Monaco£0.10£0.50
Bangladesh£0.05£0.15Mongolia£0.05£0.15
Barbados£0.20£0.40Montenegro£0.20£0.60
Belarus£0.50£0.50Montserrat£0.45£0.45
Belize£0.35£0.35Morocco£0.05£0.60
Benin£0.40£0.40Mozambique£0.10£0.20
Bermuda£0.05£0.15Myanmar£0.40£0.40
Bhutan£0.10£0.15Namibia£0.05£0.15
Bolivia£0.20£0.30Nauru£1.50 
Bosnia and Herzegowina£0.20£0.50Nepal£0.20£0.15
Botswana£0.10£0.35Netherlands Antilles£0.20£0.20
Brunei Darussalam£0.05£0.15New Caledonia£0.30 
Bulgaria£0.05£0.20Nicaragua£0.35£0.40
Burkina Faso£0.40£0.45Niger£0.50£0.50
Burundi£0.70£0.70Nigeria£0.10£0.15
Cambodia£0.10£0.15Niue£1.50 
Cameroon£0.25£0.45North Korea£1.00 
Cape Verde£0.30£0.40Northern Cyprus£0.10£0.25
Cayman Islands£0.10£0.30Northern Mariana Islands£0.05 
Central African Republic£0.60£0.60Oman£0.15£0.40
Chad£0.60£0.60Pakistan£0.20£0.20
Colombia£0.05£0.15Palau£0.40 
Comoros£0.60£0.60Palestinian Territories£0.30£0.30
Congo£0.70£0.70Panama£0.05£0.20
Cook Islands£1.50 Papua New Guinea£0.90 
Costa Rica£0.05£0.15Paraguay£0.05£0.15
Cote D’Ivoire£0.80£0.45Peru£0.05£0.15
Croatia£0.05£0.15Philippines£0.20£0.25
Cuba£1.00£1.00Puerto Rico£0.05 
Cyprus£0.05£0.15Qatar£0.25£0.25
Diego Garcia£2.00 Reunion£0.05£0.15
Djibouti£0.60£0.60Romania£0.05£0.15
Dominica£0.15£0.35Russia£0.05£0.30
Dominican Republic£0.05£0.15Rwanda£0.50£0.50
DR Congo£0.60£0.45Saint Kitts And Nevis£0.20£0.45
East Timor£1.50 Saint Lucia£0.30£0.45
Ecuador£0.15£0.30Samoa£1.00£1.00
Egypt£0.15£0.15San Marino£0.05£0.35
El Salvador£0.30£0.30Sao Tome And Principe£1.50£1.50
Equatorial Guinea£0.40 Saudi Arabia£0.15£0.20
Eritrea£0.35£0.35Senegal£0.60£0.60
Estonia£1.00£0.70Serbia£0.20£0.60
Ethiopia£0.35£0.40Seychelles£0.80£0.80
Falkland Islands (Malvinas)£2.50 Sierra Leone£0.80£0.70
Faroe Islands£0.10£0.35Slovakia£0.05£0.15
Fiji£0.30£0.30Slovenia£0.05£0.35
Finland£0.05£0.15Solomon Islands£1.50 
French Guiana£0.05£0.15Somalia£0.70£0.70
French Polynesia£0.30£0.50South Africa£0.05£0.15
Gabon£0.70£0.70South Sudan£1.50£1.50
Gambia£0.90£1.00Sri Lanka£0.20£0.15
Georgia£0.10£0.15St. Helena£3.00 
Ghana£0.35£0.35St. Pierre And Miquelon£0.40£0.70
Gibraltar£0.05£0.35St. Vincents£0.25£0.45
Greenland£0.90£0.90Sudan£0.25£0.25
Grenada£0.25£0.40Suriname£0.20£0.30
Guadeloupe£0.05£0.15Swaziland£0.10£0.25
Guam£0.05 Syrian Arab Republic£0.15£0.25
Guatemala£0.15£0.20Tajikistan£0.20£0.25
Guinea£0.70£0.70Tanzania£0.50£0.45
Guinea-Bissau£0.60£0.90Thailand£0.05£0.15
Guyana£0.40£0.40Togo£0.60£0.60
Haiti£0.45£0.45Tokelau£2.50 
Honduras£0.15£0.25Tonga£0.70 
Iceland£0.05£0.15Trinidad And Tobago£0.10£0.30
Indonesia£0.05£0.15Tunisia£0.70£0.70
Iran£0.10£0.15Turkey£0.05£0.25
Iraq£0.20£0.30Turkmenistan£0.15 
Jamaica£0.20£0.35Turks And Caicos Islands£0.20£0.40
Jordan£0.15£0.15Tuvalu£1.50 
Kazakhstan£0.05£0.25Uganda£0.50£0.50
Kenya£0.20£0.20Ukraine£0.20£0.35
Kiribati£1.50 United Arab Emirates£0.25£0.25
Kuwait£0.05£0.15Uruguay£0.10£0.25
Kyrgyzstan£0.25£0.25Uzbekistan£0.10£0.15
Laos£0.05£0.15Vanuatu£0.80 
Latvia£0.05£0.30Vatican£0.05 
Lebanon£0.15£0.25Venezuela£0.05£0.15
Lesotho£0.30£0.30Viet Nam£0.10£0.15
Liberia£0.70£0.70Virgin Islands (British)£0.25£0.45
Libya£0.35£0.45Virgin Islands (U.S.)£0.05 
Liechtenstein£0.10£0.30Wallis & Futuna£1.50 
Lithuania£0.05£0.15Yemen£0.20£0.20
Macau£0.10£0.15Zambia£0.10£0.20
   Zimbabwe£0.15£0.70
Categories
Asterisk Support Elastix Support Knowledge Base

IAX2 Cause code

Here is a table of the IAX2 to assist with debugging IAX2 call issues

More IAX2 information can be found here and the RFC is here


CSV
 download is here
Number Cause Reference
1 Unassigned/unallocated number [RFC5457]
2 No route to specified transit network [RFC5457]
3 No route to specified transit network [RFC5457]
4-5 Unassigned
6 Channel unacceptable [RFC5457]
7 Call awarded and delivered [RFC5457]
8-15 Unassigned
16 Normal call clearing [RFC5457]
17 User busy [RFC5457]
18 No user response [RFC5457]
19 No answer [RFC5457]
20 Unassigned
21 Call rejected [RFC5457]
22 Number changed [RFC5457]
23-26 Unassigned
27 Destination out of order [RFC5457]
28 Invalid number format/incomplete number [RFC5457]
29 Facility rejected [RFC5457]
30 Response to status enquiry [RFC5457]
31 Normal, unspecified [RFC5457]
32-33 Unassigned
34 No circuit/channel available [RFC5457]
35-37 Unassigned
38 Network out of order [RFC5457]
39-40 Unassigned
41 Temporary failure [RFC5457]
42 Switch congestion [RFC5457]
43 Access information discarded [RFC5457]
44 Requested channel not available [RFC5457]
45 Pre-empted (causes.h only) [RFC5457]
46 Unassigned
47 Resource unavailable, unspecified (Q.931 only) [RFC5457]
48-49 Unassigned
50 Facility not subscribed (causes.h only) [RFC5457]
51 Unassigned
52 Outgoing call barred (causes.h only) [RFC5457]
53 Unassigned
54 Incoming call barred (causes.h only) [RFC5457]
55-56 Unassigned
57 Bearer capability not authorized [RFC5457]
58 Bearer capability not available [RFC5457]
59-62 Unassigned
63 Service or option not available (Q.931 only) [RFC5457]
64 Unassigned
65 Bearer capability not implemented [RFC5457]
66 Channel type not implemented [RFC5457]
67-68 Unassigned
69 Facility not implemented [RFC5457]
70 Only restricted digital information bearer capability is available (Q.931 only) [RFC5457]
71-78 Unassigned
79 Service or option not available (Q.931 only) [RFC5457]
80 Unassigned
81 Invalid call reference [RFC5457]
82 Identified channel does not exist (Q.931 only) [RFC5457]
83 A suspended call exists, but this call identity does not (Q.931 only) [RFC5457]
84 Call identity in use (Q.931 only) [RFC5457]
85 No call suspended (Q.931 only) [RFC5457]
86 Call has been cleared (Q.931 only) [RFC5457]
87 Unassigned
88 Incompatible destination [RFC5457]
89-90 Unassigned
91 Invalid transit network selection (Q.931 only) [RFC5457]
92-94 Unassigned
95 Invalid message, unspecified [RFC5457]
96 Mandatory information element missing (Q.931 only) [RFC5457]
97 Message type nonexistent/not implemented [RFC5457]
98 Message not compatible with call state [RFC5457]
99 Information element nonexistent [RFC5457]
100 Invalid information element contents [RFC5457]
101 Message not compatible with call state [RFC5457]
102 Recovery on timer expiration [RFC5457]
103 Mandatory information element length error (causes.h only) [RFC5457]
104-110 Unassigned
111 Protocol error, unspecified [RFC5457]
112-126 Unassigned
127 Internetworking, unspecified [RFC5457]
128-255 Unassigned

 

Categories
QueueMetrics Support Software

QueueMetrics,  The Advanced Call Center Software Solution Suite. Measure your targets, conversion rates and agent activities. Create accurate reports and statistics. Set security and privacy on individual queues. Support virtual and multi-tenant production environments.

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Categories
Asterisk Support Elastix Support Knowledge Base Security

Keeping the Bots at bay out and allowing your friends in

Recently we have seen an upsurge in Bots attacking Asterisk servers, Interestingly its not good old sipvicious anymore but a Windows program called sipcli and originating mainly from the US and Germany.

Normally our iptables firewalls are updated but for some reason these keep getting through, So we have now based rules on the User-Agent in iptables as well

Here are a few examples to get rid of many of the favourites

-A INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: friendly-scanner" --algo bm --to 65535 -j DROP
-A INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: sipcli" --algo bm --to 65535 -j DROP
-A INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: sipvicious" --algo bm --to 65535 -j DROP
-A INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: VaxSIPUserAgent" --algo bm --to 65535 -j DROP

Also its worth adding these ranges as little good will ever come from them

# Ponytelecom ranges
-A INPUT -s 62.210.0.0/16 -j DROP
-A INPUT -s 195.154.0.0/16 -j DROP
-A INPUT -s 212.129.0.0/18 -j DROP
-A INPUT -s 62.4.0.0/19 -j DROP
-A INPUT -s 212.83.128.0/19 -j DROP
-A INPUT -s 212.83.160.0/19 -j DROP
-A INPUT -s 212.47.224.0/19 -j DROP
-A INPUT -s 163.172.0.0/16 -j DROP
-A INPUT -s 51.15.0.0/16 -j DROP
-A INPUT -s 151.115.0.0/16 -j DROP

# VITOX TELECOM
-A INPUT -s 77.247.109.0/255.255.255.0 -p udp -j DROP 
-A INPUT -s 185.53.88.0/24 -p udp -j DROP 
-A INPUT -s 185.53.89.0/24 -p udp -j DROP 
-A INPUT -s 37.49.224.0/24 -p udp -j DROP 
-A INPUT -s 37.49.230.0/24 -p udp -j DROP 
-A INPUT -s 37.49.231.0/24 -p udp -j DROP 
-A INPUT -s 77.247.110.0/255.255.255.0 -p udp -j DROP

For Freepbx format add following to /etc/firewall-4.rules

-A fpbxreject -s 37.49.231.0/24  -m udp -p udp -j DROP
-A fpbxreject -s 37.120.129.0/19   -p udp -j DROP
-A fpbxreject -s 185.53.88.0/24  -p udp -j DROP
-A fpbxreject -s 185.53.89.0/24  -p udp -j DROP
-A fpbxreject -s 185.53.90.0/24  -p udp -j DROP
-A fpbxreject -s 185.53.91.0/24  -p udp -j DROP
-A fpbxreject -s 37.49.224.0/24  -p udp -j DROP
-A fpbxreject -s 37.49.225.0/24  -p udp -j DROP
-A fpbxreject -s 37.49.227.0/24  -p udp -j DROP
-A fpbxreject -s 37.49.228.0/24  -p udp -j DROP
-A fpbxreject -s 37.49.229.0/24  -p udp -j DROP
-A fpbxreject -s 37.49.230.0/24  -p udp -j DROP
-A fpbxreject -s 37.49.231.0/24  -p udp -j DROP
-A fpbxreject -s 77.247.108.0/24  -p udp -j DROP
-A fpbxreject -s 77.247.109.0/24  -p udp -j DROP
-A fpbxreject -s 77.247.110.0/24  -p udp -j DROP
-A fpbxreject -s 77.247.111.0/24  -p udp -j DROP
-A fpbxreject -s 62.210.0.0/16 -p udp -j DROP
-A fpbxreject -s 195.154.0.0/16 -p udp -j DROP
-A fpbxreject -s 212.129.0.0/18 -p udp -j DROP
-A fpbxreject -s 62.4.0.0/19 -p udp -j DROP
-A fpbxreject -s 212.83.128.0/19 -p udp -j DROP
-A fpbxreject -s 212.83.160.0/19 -p udp -j DROP
-A fpbxreject -s 212.47.224.0/19 -p udp -j DROP
-A fpbxreject -s 163.172.0.0/16 -p udp -j DROP
-A fpbxreject -s 51.15.0.0/16 -p udp -j DROP
-A fpbxreject -s 151.115.0.0/16 -p udp -j DROP

If you are still getting problems check out a sip trace and look for the contact part of the

Contact: <sip:100@xxx.www.rrr.zzz:5070>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE
User-Agent: sipcli/v1.8                       <<<<<<<<<<<<<<<< here it is 
Content-Type: application/sdp
Below is a simple Bash script to create iptables entry for Linux. Create a script and paste the code in , if you just run it it created entries ready for Centos iptables id you run  ‘scriptname freepbx’ it created the entry for /etc/firewall-4.rules
#!/usr/bin/bash
SCANNERS='owenee Custom SIP gazllove pplsip sipcli sipvicious sip-scan sipsak sundayddr friendly-scanner iWar CSipSimple SIVuS Gulp sipv smap friendly-request VaxIPUserAgent VaxSIPUserAgent siparmyknife Test'
SYS=$1
if [ "$SYS" != "freepbx" ]
then
SYS=NOOP
fi
echo This is for a $SYS system
echo Copy and paste below
echo
PORTS='5060:5261'
PROTOS='udp tcp'
for scanner in $SCANNERS; do
for port in $PORTS; do
for proto in $PROTOS; do
if [ "$SYS" = "freepbx" ]
then
echo -A fpbxreject -p $proto -m $proto --dport $port -m string --string '"User-Agent:' $scanner'"' --algo bm --to 65535 -j DROP
else
echo -A INPUT -p $proto -m $proto --dport $port -m string --string '"User-Agent:' $scanner'"' --algo bm --to 65535 -j DROP
fi
done
done
done

In this case just set as we have in iptables and it will catch all versions.

Hope this helps you as much as it has helped us

Also this idea can be reversed to only allow user agents (phones) you want to accept.

Here are a few examples of common soft and hardphones

-A ELASTIX_INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: Yealink" --algo bm --to 65535 -j ACCEPT
-A ELASTIX_INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: FPBX" --algo bm --to 65535 -j ACCEPT
-A ELASTIX_INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: Linphone" --algo bm --to 65535 -j ACCEPT
-A ELASTIX_INPUT  -p udp -m udp --dport 5060 -m string --string "User-Agent: DX800" --algo bm --to 65535 -j ACCEPT
-A ELASTIX_INPUT  -p udp -m udp --dport 5060 -m string --string "User-Agent: 3CX" --algo bm --to 65535 -j ACCEPT
-A ELASTIX_INPUT  -p udp -m udp --dport 5060 -m string --string "User-Agent: Grand" --algo bm --to 65535 -j ACCEPT

Again to find others just do a sip trace and note down the user agent.

This can also be extended to make you system more secure by only allowing in devices that register to you FQDN and not just ip address

# Firewall configuration written by system-config-firewall
# Manual customization of this file is not recommended.
*filter
:INPUT ACCEPT [0:0]
:FORWARD ACCEPT [0:0]
:OUTPUT ACCEPT [0:0]
-A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
-A INPUT -p icmp -j ACCEPT
-A INPUT -i lo -j ACCEPT
# -A INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j ACCEPT
#ssh moved from 22 to random port
-A INPUT -m state --state NEW -m tcp -p tcp --dport 65432 -j ACCEPT
#Web interface moved to new port.
-A INPUT -m state --state NEW -m tcp -p tcp --dport 8765 -j ACCEPT
#drop sipvicious traffic
-A INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: sipvicious" --algo bm --to 65535 -j DROP
-A INPUT -i eth0 -p udp --dport 5060 -m state --state ESTABLISHED,RELATED -j ACCEPT
-A INPUT -i eth0 -p udp --dport 4569 -m state --state ESTABLISHED,RELATED -j ACCEPT
#only allow Yealink phones
-A ELASTIX_INPUT -p udp -m udp --dport 5060 -m string --string "User-Agent: Yealink" --algo bm --to 65535 -j ACCEPT
#That register to your domain name directly
-A INPUT -i eth0 -p udp --dport 5060 -m string --string "REGISTER sip:yoursip.yourdomain.co.uk" --algo bm -j ACCEPT 
#only allow iax from known server
-A INPUT -s xxx.xxx.xxx.0/22 -p udp -m udp --dport 4569 -j ACCEPT
-A INPUT -i eth0 -p udp --dport 5060 -j DROP
-A INPUT -i eth0 -p udp --dport 10000:20000 -j ACCEPT
-A INPUT -j REJECT --reject-with icmp-host-prohibited
-A FORWARD -j REJECT --reject-with icmp-host-prohibited
COMMIT

The above example should keep you secure. (but things and methods change so keep your eye on the ball)