Categories
Blog FreePBX Software

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.

The software is opensource 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

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.

If you do download and like it, maybe think about buying me a coffee

Categories
Asterisk Support Blog Elastix Support FreePBX Knowledge Base Security

Keeping the Bots out and allowing your friends in

Since this post was originally written things have advanced, FreePBX has an integrated firewall with intrusion detection using Fail2Ban, and this should always be enabled even if system is on premise.

Another major step forward in protection is APIBAN this is a client program that helps prevent unwanted SIP traffic by identifying addresses of known bad actors before they attack your system. Bad bots are collected through globally deployed honeypots. To use APIBAN you will need a key these are obtained from here . More details on API ban are here if you are interested in using it in different situations.

To simplify installation on Freepbx based systems I have simple script that downloads and install it, this can be downloaded here or from the command line of the server as follows:

wget https://freeaccesspublic.s3.eu-west-2.amazonaws.com/apiban.sh
Make it an executable : chmod +x  apiban.sh
then run the script : ./apiban.sh your_api_key

If you dont add your APIKEY on the command line vi will open and you can add it manually. The script will then initially run the client which will take a few seconds to download the initial set of bots, then it will add a line to the crontab file and restart the cron daemon. the timing of the cronjob is randomised to be between every 4 and 22 minutes.

We have seen many Bots attacking Asterisk servers, Interestingly its not always 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

For Freepbx format add following to the Firewalls custom rules


-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "sip:a'or'3=3--@" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: PolycomSoundPointIP SPIP_550 UA 3.3.2.0413" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: Avaya IP Phone 1120E" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: PolycomVVX-VVX_401-UA5.4.1.18405" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: eyeBeam release 3006o stamp 17551" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: owenee" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: owenee" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: Custom" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: Custom" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: SIP" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: SIP" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: gazllove" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: gazllove" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: pplsip" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: pplsip" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sipcli" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sipcli" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sipvicious" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sipvicious" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sip-scan" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sip-scan" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sipsak" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sipsak" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sundayddr" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sundayddr" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: friendly-scanner" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: friendly-scanner" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: iWar" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: iWar" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: CSipSimple" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: CSipSimple" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: SIVuS" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: SIVuS" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: Gulp" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: Gulp" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: sipv" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: sipv" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: smap" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: smap" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: friendly-request" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: friendly-request" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: VaxIPUserAgent" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: VaxIPUserAgent" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: VaxSIPUserAgent" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: VaxSIPUserAgent" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: siparmyknife" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: siparmyknife" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp -m udp --dport 5060:5261 -m string --string "User-Agent: Test" --algo bm --to 65535 -j DROP
-A fpbxreject -p tcp -m tcp --dport 5060:5261 -m string --string "User-Agent: Test" --algo bm --to 65535 -j DROP
-A fpbxreject -p udp --dport 5060:5261 -m string --string "REGISTER sip:server.domain.co.uk" --algo bm -j ACCEPT
-A fpbxreject -p udp --dport 5060:5261 -m string --string "REGISTER sip:" --algo bm -j DROP
-A fpbxreject -p tcp --dport 5060:5261 -m string --string "REGISTER sip:server.domain.co.uk" --algo bm -j ACCEPT
-A fpbxreject -p tcp --dport 5060:5261 -m string --string "REGISTER sip:" --algo bm -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
Categories
Blog Handsets Products Sangoma Phones Software

Sangoma Connect Mobile

Revolutionise Your Business Communications: In the modern corporate landscape, the traditional office desk is no longer the sole hub of productivity. Whether your team is working from home, travelling for client meetings, or managing critical on,call shifts, the need for seamless connectivity is paramount. Enter Sangoma Connect, the mobile softphone application designed to turn your smartphone into a fully featured business extension.

For businesses looking to simplify workforce management and reduce the complexities of on,call rotations, Sangoma Connect offers a robust, secure, and cost,effective solution. This guide explores the extensive feature set of the application and demonstrates how it can transform your operational efficiency and bottom line.

What is Sangoma Connect?

Sangoma Connect is a high,performance mobile application for iOS and Android that integrates natively with Sangoma PBXact and FreePBX phone systems. It is engineered to provide a consistent “in,office” experience regardless of a user’s physical location. By leveraging VoIP (Voice over Internet Protocol) technology, the app allows staff to make and receive high,quality voice and video calls using their business phone number, ensuring professional identity remains intact while keeping personal mobile numbers private.

The application is more than just a dialler, it is a unified communications (UC) hub. It bridges the gap between traditional telephony and modern collaboration tools, offering everything from instant messaging to multi,party video conferencing within a single, intuitive interface.

The core of Sangoma Connect is its ability to deliver crystal,clear audio and high,definition video calls. Because the app uses the business’s PBX infrastructure, calls are routed via the internet (Wi,Fi or mobile data), providing a reliable connection even when cellular signals are weak. This is particularly vital for on,call staff who must remain reachable in various environments.

Gone are the days of manual configuration and complex setup codes. Sangoma Connect features a “magic link” onboarding process. Once an administrator enables a user in the PBX management module, the employee receives an auto,generated email. By clicking the link within the email, the app automatically configures itself, logging the user in without the need for hostnames or passwords.

Understanding the availability of your team is crucial for efficient workforce management. The Presence feature allows users to set their status, such as “Available”, “Away”, or “Do Not Disturb”. This status is synchronised across the entire Sangoma ecosystem, including desk phones and desktop clients. For managers, this provides an at,a,glance view of who is ready to take a call, which is essential for coordinating on,call responses.

Managing messages is simplified through the Visual Voicemail interface. Instead of dialling into a system and listening to prompts, users can see a list of their voicemails, see who called, and play back messages in any order. This allows on,call staff to prioritise urgent queries quickly without wasting time on administrative navigation.

Professionalism is maintained through powerful call control features. Users can perform “blind” transfers (sending a call directly to another extension) or “warm” transfers (speaking to the colleague before passing the call over). This ensures that customers are always directed to the right expert with the necessary context.

Collaboration often requires more than two people. Sangoma Connect supports native 3,way audio conferencing, allowing users to add a third party to an active call instantly. For larger groups, the app integrates with Sangoma Meet, providing a platform for multi,party video collaboration and screen sharing directly from the mobile device.

Security is a non,negotiable for modern businesses. Sangoma Connect uses advanced encryption protocols, specifically TLS (Transport Layer Security) and SRTP (Secure Real,time Transport Protocol). This ensures that every conversation is private and protected from eavesdropping, whether the user is on a private home network or a public Wi,Fi hotspot.

Push Notifications

To preserve battery life without missing important calls, the app utilises push notifications. The application does not need to run in the foreground to receive an alert, the system “wakes” the app when a call or message arrives, ensuring reliability for on,call personnel who cannot afford to miss a notification.

Strategic Advantages for On,Call Workforce Management

Managing a workforce that is frequently on the move or “on,call” presents unique logistical challenges. Sangoma Connect is specifically tailored to address these pain points, creating a more agile and responsive team.

Simplifying the On,Call Rotation

When an employee is on,call, they are often tethered to a physical handset or forced to give out their personal mobile number. Sangoma Connect removes these barriers. Because the app acts as a secondary extension, managers can use the PBX’s “Follow Me” and “Call Forwarding” features to route calls to the on,call staff’s mobile device automatically.

If an on,call technician is busy, the “Presence” indicator lets the rest of the team know immediately, allowing the system to route the call to the next available person in the queue. This level of transparency reduces “phone tag” and ensures that urgent client issues are resolved faster.

Professionalism and Privacy

One of the greatest benefits for staff is the separation of business and personal life. When making an outgoing call via Sangoma Connect, the recipient sees the company’s caller ID, not the employee’s personal mobile number. This maintains a professional image and protects the privacy of your staff, which is a significant factor in employee satisfaction and retention for those working outside traditional hours.

Increasing Productivity Through Unified Communications

Productivity is often lost in the “spaces between” tasks, the time spent checking various apps, returning to the office to check a desk phone, or trying to track down a colleague. Sangoma Connect recaptures this time.

Real,Time Collaboration

With integrated team chat and file sharing, employees can collaborate on the fly. An on,call engineer at a client site can instantly message a senior architect for advice, send a photo of a technical issue, or even start a video call to show the problem in real,time. This immediate access to collective knowledge drastically reduces the “Time to Resolution” for complex tasks.

Geographic Flexibility

Sangoma Connect empowers businesses to hire the best talent regardless of location. Because the communication system is entirely portable, a business based in London can have on,call support staff in Manchester or Edinburgh, all appearing as if they are sitting in the same office. This flexibility allows for better coverage across different time zones without the need for expensive regional offices.

Unlocking Cost Savings

While the productivity gains are significant, the financial benefits of adopting Sangoma Connect are equally compelling for any business owner or IT manager.

Elimination of Hardware Costs

Traditional business telephony requires a desk phone for every employee. With Sangoma Connect, your employees’ existing smartphones become their primary or secondary business devices. For remote or mobile workers, this eliminates the need to purchase, ship, and maintain physical hardware, leading to substantial savings on capital expenditure (CAPEX).

Reduced Mobile Tariffs and Roaming Charges

Since Sangoma Connect uses the data network to route calls, it bypasses traditional mobile voice minutes. For businesses with international on,call requirements, this is a game,changer. Employees can make “internal” calls to the office or other colleagues for free from anywhere in the world, provided they have an internet connection, effectively eliminating expensive roaming charges.

Simplified IT Infrastructure

Sangoma Connect is managed via the same interface as your main phone system. This centralisation reduces the administrative burden on your IT department. With automated provisioning and cloud,based certificate management, the time required to manage mobile extensions is reduced from hours to minutes, allowing your IT team to focus on higher,value projects.

In Conclusion: Sangoma Connect is more than an app, it is a strategic tool that empowers your business to be more responsive, more professional, and more efficient. By simplifying the management of on,call staff and unifying your communication channels, you create an environment where productivity thrives and costs are kept firmly under control.

Whether you are a small business looking to scale or a large enterprise seeking to modernise your workforce management, Sangoma Connect provides the features and flexibility required for success in a mobile,first world.

Categories
Blog Case Studies

An out of the normal Customer request

and how we solved it:

We were approached by one of our customers who provides support services to travellers and global companies who had a client that provides maritime engineering services world wide and required an emergency helpline that “followed the sun” 

Detailed Specification

A single number that called dependant on time the on call support staff.

The calls cannot go to users Voicemail.

The staff members are to be notified by email that the call was taken and who took it.

If the on call staff do not answer the call it is forwarded to our clients call centre.

On completion of the call a copy of the recording in mp3 format is emailed to the on call staff.

Solution.

Our customer uses FreePBX so the core of the project is the use of the Queue application but with some custom dial plan and scripts to exploit and enhance features that are not exposed, 

The inbound numbers destination is a “custom destination” that first sends it to some custom dial plan explained later and then to the “Call Flow Control” application to allow the system to be overridden, It is then sent to the “Time Conditions” application that uses UTC as its time zone to over come issues with daylight saving in different hemispheres, this then send the call to the correct queue depending on time 

To allow an email to be sent to staff we used the qgosub variable that is explained HERE , this sub routine sends the email on answer. this variable is set by a small dial plan snippet that sets the qgosub variable and an additional one to set a channel variable as the callers callerID number, as its lost when the call is made to the staff members by the queue application. 

To make sure calls do not go to voicemail, the queue option “call confirm” this forces the called staff to press 1 to accept a call, This much overlooked option is useful for many queue scenarios.

If the call is unanswered the call has to be passed to the callcenter with the callerID name tagged with the customers Name, We achieve this with the “SetCallerID” application passing the call onto the client call centre.

Finally when the call is complete we need to email the recording to the customer. To do this with the “Post Call Recording Script” option in Advanced options. (You may need to enable “Display Readonly Settings” and “Override Readonly Settings”), This did require a little lateral thinking as we were already using this script to convert recordings to MP3 and save them to AWSS3 storage, But we didn’t want an email sent after all recordings do we included an additional ‘if’ statement to check if the qgosub variable was passed over to the script and if it was email the attachment otherwise do nothing.

I hope this shows the flexibility of FreePBX and asterisk and how fairly complex call routings and requests can be fulfilled in a manner that doesn’t require complex dial plans or require high support overheads.

If you want to achieve similar don’t hesitate to get in touch as by using modules already in FreePBX you’re not paying to reinvent the wheel.

Categories
Blog FreePBX Knowledge Base

Running Subroutines on answer for Queues

Some years ago we wrote a post on running macros on queue answer here. this was very useful for integration with backends, At the time we raised a feature request to get it added to Freepbx, But this never happened.

Now the variable QGOSUB is in the dialplan for freepbx queues, But still there is no way of setting this in a default freepbx installation and it requires a snip-it of custom dialplan that is called from freepbx by a ‘custom destination’ . For example at its simplest the dialplan to set it could be :-

[qmacro-set]
exten => .,1,Noop(ians test) 
exten => .,n,Set(_QGOSUB=ians_routine) 
exten => .,n,Goto(app-daynight,1,1)  

and this sets the variable for all channels in this call, and when the Queue command is run in the default freepbx dialplan

Queue(9471,${QOPTIONS},,${QAANNOUNCE},${QMAXWAIT},${QAGI},,${QGOSUB},${QRULE},${QPOSITION})  

This allows simple or more complicated routines to be run. For example sending an email on answer which was a request we had that caused us to revisit this.

[ians_routine]
exten = s,1,Set(origtime=${EPOCH})
exten = s,n,Noop(${CHANNEL})
exten = s,n,Set(Agent11=${CUT(CHANNEL,@,1)})
exten = s,n,Set(Agent12=${CUT(Agent11,/,2)})
exten = s,n,Noop(${Agent11} , ${Agent12} )
exten = s,n,Set(fulltime=${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)})
exten = s,n,system(echo "There has been a call , Callers Details: ${CALLERID(number)} ,  ${CRM_SOURCE} , Date and Time: ${fulltime} ,  Agent: ${Agent12} ,Timestamp: ${origtime} , Queue Number: ${QUEUENUM} " | mail -s "failed recall at ${fulltime}" email@address.com)
same = n,Return()

If you think that you would like to be able to set this variable in the freepbx gui give it a vote https://issues.freepbx.org/browse/FREEPBX-22274

Categories
Blog

3D Printers and why you need one.

I recently had the need for a wall bracket for a Sangoma S505 handset and the S300/S400 handsets.

Looking at my suppliers price list there were none in stock and at nearly £10 each this seemed like a perfect project for the newly acquired 3d printer. After a bit of design work on paper then in Tinkercad (Ill be moving on to fusion 360) it was ready to print.

I’ve put a link to download of the STL code here for S505 and here for the S300 so anyone can download and modify it, Its robust and angles the phone so that the handsets arnt knocked off easily, angling and supporting the phone. The S300 Bracket can also be used as an alternative desk base allowing the phone to sit flatter on a desk.

It is surprising how useful the 3d printer has become, and how readily available designs are. Another recent request was for an additional mount for a Ubiquiti wireless access point, We found a design on Thingiverse printed it out and sent it to the customer. Add to this all the little widgets that we have been making since getting it i’m not sure how we lived without one for so long.

I would urge any IT company to invest in a printer, The price point is now so low that you don’t need to produce many items to have it pay for itself, Just the 2 items mentioned here would have cost nearly £20 plus postage, and very little margin, They now cost £0.20 worth of filament and maybe £0.20 worth of renewable electricity.

Categories
Blog Calls and Lines Covid-19

Calls to 101 Non-Emergency number to be free from April 1st

Given the pressure on the emergency services presently, the Home Office have requested that the 101 Non-Emergency number be made free to all callers from 1st April 2020. Therefore these calls placed through our network will be free from midnight this evening.

We understand that this change is intended to be permanent, and not just for the duration of the Covid-19 emergency.

Please note that although this change will be in place from midnight tonight it may not be reflected in the ratesheet immediately .

Categories
Blog Knowledge Base

Presenting Local or national CLI from outside a country

Over the last six months an increasing number of countries will not allow internationally originated calls that are presenting a local or national CLI for the country called. 

It is believed that the reason behind this is part of a global wide crack down on fraud and nuisance calls, as some of the regulations and requirements introduced may have a more widespread impact on the CLI passed on calls.

Currently, the countries we are aware that are implementing these controls are:-

Australia – ACMA  ( Australian Communications & Media Authority ) are actively cracking down on common phone fraud, including malicious “spoofing”.

Australian networks are also trying to stamp out  “Wangiri” fraud,  where victims are called once from an international number with high premium charges for those who call back.

https://www.acma.gov.au/articles/2019-11/acma-recommends-immediate-action-combat-scams

Canada – CRTC (Canadian Radio-television & Telecommunications Commission ) have issued a requirement for telecoms providers to implement a system to block calls within their network or offer call-filtering service.

https://crtc.gc.ca/eng/phone/telemarketing/identit.htm

China – China have always been very strict on caller ID, as well as not permitting internationally originated calls with a local or national CLI, they do not allow calls presented with a withheld flag. 

These calls will either present the withheld CLI or the call will fail. 

France – ARCEP ( France’s regulatory body for electronic communications, portal and print media distribution ) have issued a statement to carriers that due to in country regulations, calls sent internationally with a French CLI are not permitted.

Malaysia – we are seeing that calls may fail if a local or national CLI is presented.

Turkey – ICTA ( Information & Communication Technologies Authority ) have introduced CLI regulations which require operators to block some voice calls with CLIs that may be confused with local numbers due to the format of the number presented.

United Arab Emirates – in our experience calls may fail if a local or national CLI is presented.  In addition, the UAE are blocking calls with a Tunisian or Algerian CLI.

Categories
Blog Gateways Services

ISDN Switchoff…

With the end of ISDN looming, the switch off could affect over two million businesses in the UK which will come at a cost for those affected. For many, the cost implications to replace their entire Telphone system would simply be too much, but with Sangoma’s offering, the transition from PSTN and ISDN to VoIP can provide a long-term, cost-effective solution.

Why remove a PBX that gives you everything you need? Sangoma’s Vega Gateways provide seamless connectivity to SIP providers, with no need to change any existing hardware or to re-cable your system, these gateways are designed to migrate from PSTN to SIP with minimal downtime.

Available in a range of varieties, Sangoma Vega Gateways suit businesses of all sizes from the SME to enterprise corporations. What’s more, users can benefit from advanced functionalities such as least cost routing (selecting the cheapest route for a call) and enhanced network proxy features which can help with the failover of calls from VoIP to PSTN in the event of a loss of internet connection.

If you would like more information or advice on what is the best option for your business please call or email us. Unlike many we have ISDN simulators so that we can configure your new gateway and install it with limited disruption to your business

Categories
Blog Knowledge Base

SSL, FOP2 and All that..

There are a few common questions and gotchas when using fop2 on a FreePBX server using ssl.

The symptom is simple you cant connect and at the bottom left if tels you that it cant connect over websockets to port 4445 and will finally time out saying Flash is required.. None of this is awfully helpful to be honest.

But the fix is simple and its a shame by default its not this .

In your fop2.cfg file local the lines below and edit them so they match .

It maybe that your certificate names aren’t as below, In that case substitute webserver.XXX with the correct file names.

; If you access fop2 via https, browsers will try to use wss (Secure 
; web sockets) and for that it requires a certificate file and key file,
; the same ones you have in your web server configuration. Be sure to
; specify the correct certificates, the defaults are the ones for a 
; regular Centos installation:
;

ssl_certificate_file=/etc/httpd/pki/webserver.crt
ssl_certificate_key_file=/etc/httpd/pki/webserver.key
;ssl_certificate_file=/etc/pki/tls/certs/localhost.crt
;ssl_certificate_key_file=/etc/pki/tls/private/localhost.key

Thats its. Fop2 will now work over HTTPS