We have noted that with some Fortigate routers and firewalls come with SIP helpers enabled by default.
The customer may initially not think that there is any issue and inbound and outbound calls work as expected, But we had noted on one customer site that when they did a call pickup on another phone that was ringing in the office they would not be able to hear the caller. The caller could hear them and if they put the call on and off hold they could speak normally.
On further investigation with wireshark we noted that the RTP port changed when the pickup took place. We tested this on other sites not using the Fortigate hardware and did not have this issue.
Below are listed the commands to clear the SIP helper settings from the Fortigate hardware.
Open the Fortigate CLI from the dashboard.
Enter the following commands in FortiGate’s CLI:
config system settings
set sip-helper disable
set sip-nat-trace disable
reboot the device
Reopen CLI and enter the following commands – do not enter the text after //:
config system session-helper
show //locate the SIP entry, usually 12, but can vary.
delete 12 //or the number that you identified from the previous command.
Disable RTP processing as follows:
config voip profile
edit default
config sip
set rtp disable
And finally:
config system settings
set default-voip-alg-mode kernel-helper based
End
on a fortigate 200D the following is the method to use
Step 1) Removing the session helper.
Run the following commands:
config system session-helper show
Amongst the displayed settings will be one similar to the following example:
edit 13 set name sip set protocol 17 set port 5060
In this example the next commands would be:
delete 13 end Step 2) Change the default –voip –alg-mode.
Run the following commands:
config system settings set default-voip-alg-mode kernel-helper based end Step 3) Either clear sessions or reboot to make sure changes take effect
a) To clear sessions
The command to clear sessions applies to ALL sessions unless a filter is applied, and therefore will interrupt traffic.
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.
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:
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.
In freepbx there is a feature that is quite well hidden but actually does a very useful job.
In The “Advanced Settings” page if you enable both “Post Call Recording Script” As the name suggests this is a script that run after a recorded call has ended. We created a script called postrecord.sh and in the text field on the menu we have put as below. This emails both inbound and outbound calls.
For calls to and from an extension we can pull the email address from the voicemail.conf and send the email to that address.
Its also set to delete the wav file away after a defined number of days.
The Script below will first convert the recording then email it to you or your customer.
A couple of prerequisites are required, these are sox and lame. sox is probably already installed, lame maybe not.
Installing Lame is simple for centos as below.
wget http://sourceforge.net/projects/lame/files/lame/3.99/lame-3.99.5.tar.gz tar -zxvf lame-3.99.5.tar.gz
cd lame-3.99.5
./configure
make
make install
The script is fairly simple as below. the main variables are passed to it but we build the directory structure on the fly and file extension is fixed as wav. you can set the file_age variable to delete the wav file messages over that many days old.
Be careful if cutting and pasting this scripty as wordpress may have wrapped some lines
#!/bin/bash
#This script emails the recorded call right after the call is hung up. Below arethe variables passed through asterisk
# $1 - Time String
# $2 - Source
# $3 - File
# $4 - unique id
# $5 - Destination
# $dt - Date and Time
/bin/nice /bin/sleep 3
dy=$(date '+%Y')
dm=$(date '+%m')
dd=$(date '+%d')
file_age=35
dtpath=/var/spool/asterisk/monitor/$dy/$dm/$dd/
/bin/nice /usr/local/bin/lame -b 16 -m m -q 9-resample $dtpath$3.wav$dtpath$3.mp3
/bin/nice /bin/chownasterisk:asterisk $dtpath$3.mp3
dt=$(date '+%m/%d/%Y %r');
id=$(mysql -uUser -pPassword -s -N -e "SELECT descr from asterisk.queues_config where extension = $5");
email=recordings@yourdomain.com
file=$dtpath$3
if [ "$id" = "" ]; then
direction=callers
id=$(mysql -uUser -pPassword -s -N -e "SELECT name from asterisk.users where extension = $2");
IN=$(/bin/grep "$2 =>" /etc/asterisk/voicemail.conf)
echo $IN
set -- "$IN"
IFS=","; declare -a Array=($*)
email=${Array[2]}
else
direction=customers fi
echo -e "You have a new call recording to listen to\n\n
The call date and time was: $dat \n\n
The call was from: $2 \n\n The call was to: $5 \n\n
The $direction name was: $id \n\n
And the unique call id was: $4 \n\n
Please see the attached file \n\n" | mail -a $file.mp3 -s "New Recording at $dt" $email
/bin/nice /usr/bin/find /var/spool/asterisk/monitor/-type f -mtime +"$file_age" |grep wav | \
while read I; do
/bin/rm"$I"
done
We recently had a very puzzling issue with a customer who we supplied some T23 Yealink handsets. When making outgoing calls over Gamma sip trunks on their Elastix server we were getting one way audio, This was not an issue with their existing Snom handsets or a problem for internal or incoming calls over the same trunks. It also wasn’t an issue when using iax2 trunks.
It seems that there is some interoperability issue when using sip trunks and these handsets. and seems to be a little known issue as only affects a few operators.
It seems to addressed in 44.80.0.20 version software that isn’t on the Yealink UK site yet but is available here and should be loaded on all T23 handsets as they are being delivered as 44.80.0.5 firmware at the moment.
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.
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.
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):
Premium Mobile 07520, 07744, 07755, 078222, 074416, 07777, 078228
20.00
20.00
20.0
Bundled minutes.
PPM
Number of Minutes
Price
1p
500
£5.00
0.8p
1,000
£8.00
0.6p
2,500
£15.00
0.5p
5,000
£25.00
0.4p
10,000
£40.00
0.35p
25,000
£87.50
0.3p
50,000
£150.00
0.25p
100,000
£250.00
UK Landline Minutes
Number of Minutes
Price
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:
Route
Peak
Off-peak
Weekend
Per Call Charge
118118
5p
5p
£6.50
5p (£3.85 weekend)
118226
£1.00
£1.00
£1.00
55p
118365
£1.50
£1.50
£1.50
45p
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:
Argentina
China
Hungary
Malaysia
South Korea
Australia
Czech Republic
India
Netherlands
Spain
Austria
Denmark
Ireland
New Zealand
Sweden
Belgium
France
Israel
Norway
Switzerland
Brazil
Germany
Italy
Poland
Taiwan
Canada
Greece
Japan
Portugal
United States
Chile
Hong Kong
Luxembourg
Singapore
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:
We have recently installed and customised a Queuemetrics solution for a customer. Their key reason for choosing Queuemetrics was the ability to use dynamic agents without the need of major Elastix reprogramming and a clear and simple interface.
It was also decided that users needed to be able to log into the system from their handsets as well as from a web interface, as the customer is hoping to roll out an agent portal in the future.
To enable Hotdesk the following setting has to be set similar to below
# The value is interval time (in seconds) used by the analyzer to look back searching HOTDESK verbs in the queue log
default.hotdesking=86400
This meant some additional dialplans to allow logging in & out and pausing.
These are similar to the dialplans that the web portals use except that they have prompts and they also have to store the extension and agent id in the asterisk database.
;added dialplan for queuemetrics
; Add Member - 422
; User is asked for their loging agent id
exten => _422XXXX,1,Answer
exten => _422XXXX,2,Read(AGENTID,agent-login,4,,1,6)
exten => _422XXXX,3,Gotoif($["${AGENTID}" = ""]?end)
exten => _422XXXX,4,GotoIf($[${LEN(${AGENTID})} != 4]?2)
exten => _422XXXX,5,set(DB(qmagent/${CALLERID(num)})=${AGENTID})
exten => _422XXXX,6,Macro(queuelog,${EPOCH},${UNIQUEID},NONE,Agent/${AGENTID},HOTDESK,SIP/${CALLERID(num)})
exten => _422XXXX,7,AddQueueMember(${EXTEN:3:4},SIP/${CALLERID(num)})
exten => _422XXXX,8,SayDigits(${AGENTID})
exten => _422XXXX,9,Playback(agent-loginok)
exten => _422XXXX,10(end),Hangup
; extension 33: agent unpause with hotdesking
exten => 33,1,Answer
exten => 33,2,NoOp( "QM: Unpausing Agent/${AGENTCODE} at extension SIP/${CALLERID(num)} made by '${QM_LOGIN}' " )
exten => 33,3,UnpauseQueueMember(,SIP/${CALLERID(num)})
exten => 33,4,Playback(dictate/pause)
exten => 33,5,Playback(removed)
exten => 33,6,Hangup
These need to be added to your extensions_custom.conf file in a context that’s included in the from-internal context.
Also a change has to be made to the dialplans in the extensions_queuemetrics.conf to store and delete the database entry as well.
The system has proved to deliver what was expected and will shortly be expanded to track outbound calls and the addition of custom wallboards similar to what we recently produced for another customer.
If you would like to talk about adding QueueMetrics to your Asterisk system or are looking for a complete phone system and queuemetrics platform please contact us.
We have recently added the 5th system to the customers international VoIP network. This system was for their Polish office and is linked to their Tokyo, Sydney, Singapore and London office systems.
For this site a Sangoma FREEPBX 60 system was chosen for ease of remote deployment and reliability. The Tokyo & Sydney offices already has a Xorcom XR2000 systems whilst the London and Singapore offices have a OpenvoxAsterisk solutions. .
For the New International offices FreePBX systems were chosen as they provide a full turnkey system that can be sent out to the office plugged in. The systems initially obtain their IP address by DHCP and once a port is forwarded through the firewall to this address a fixed IP address is assigned and the customer firewall updated. Access to The GUI is by a SSH tunnel so that other than a random port for SSH and a port for IAX2 no other ports need to be opened on the customer firewall. Endpoint manager makes the deployment of handsets on the remote systems a simple and reliable process.
All systems have been linked by IAX2 trunks and the dial-plan configured so that desk to desk calls can be made between all offices and outgoing calls break out of the closest geographic system, for example a user in Sydney making a call to a UK number will have the call originate from the London system and the same goes for Tokyo, Singapore and Polish users calling UK or international numbers.
The network of systems is key to the support of the customers 24×7 support service. This is controlled by a dial-plan that is complicated by the fact that Japan does not have “Daylight saving” so even though the calls land on the UK system we had to configure the dial-plan to take into account local time in Tokyo and not base routing solely on UK time. This has proved reliable and very successful.
All systems on the network are monitored 24×7 by our Nagios monitoring platform, Not only monitoring Asterisk but also monitoring the status of the international IAX2 trunks.
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