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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
Asterisk Support Knowledge Base Products and services Technical

Gradwell IP Address ranges

At Gradwell, they send internet traffic from different addresses (known as IP addresses) to allow their telephony systems to work smoothly. Below is the list of IP addresses where their VoIP (Voice over IP) traffic will come from. It’s important that your firewall allows traffic from these addresses however they recommend you don’t set it to allow only from these, just that they are included.

The reason they say don’t allow only these addresses is that there network is dynamic and may shift or new items added and we don’t want this to affect your service.

There are a couple of things you should do to ensure you get the most from the Gradwell Voice services:

  • Check your firewall filtering – is there anything being excluded?
    • If yes – Allow the IP range traffic – this will most likely be in your firewall settings or tools (they all differ so they can’t exactly point you there)
    • If no – you’re good to go
  • If you use UDP traffic then you’ll need to allow Media ports (known as RTP) with the numbers 1024 to 65535

Current ranges as of summer 2021

109.224.232.0/22 109.224.232.0 to 109.224.235.255
109.224.240.0/22 109.224.240.0 to 109.224.243.255
109.239.96.132/31 109.239.96.132 to 109.239.96.133
141.170.24.21/31 141.170.24.21 to 141.170.24.22
141.170.24.5/31 141.170.24.5 to 141.170.24.6
141.170.50.16/28 141.170.50.16 to 141.170.50.31
185.47.148.0/24 185.47.148.0 to 185.47.148.255
194.145.188.224/27 194.145.188.224 to 194.145.188.255
194.145.189.52/31 194.145.189.52 to 194.145.189.53
194.145.190.128/26 194.145.190.128 to 194.145.190.191
194.145.191.128/27 194.145.191.128 to 194.145.191.159
195.74.60.0/23 195.74.60.0 to 195.74.61.255
213.166.3.128/26 213.166.3.129 - 213.166.3.190
213.166.4.128/26 213.166.4.129 - 213.166.4.190
213.166.5.0/24 213.166.5.0 to 213.166.5.255
78.40.243.192/27 78.40.243.192 to 78.40.243.223
87.238.72.128/26 87.238.72.128 to 87.238.72.191
87.238.73.128/26 87.238.73.128 to 87.238.73.191
87.238.74.128/26 87.238.74.128 to 87.238.74.191
87.238.77.128/26 87.238.77.128 to 87.238.77.191

To simplify things a bit listed below are the ranges in common formats.

Rules for Freepbx Custom file “firewall-4.rules”

-A fpbxreject -s 109.224.232.0/22 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s 109.224.240.0/22 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	109.224.222.16/28	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	109.224.232.0/22	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	109.224.240.0/22	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	109.239.96.132/31	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	141.170.24.20/30	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	141.170.24.5/31	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	141.170.50.16/28	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	185.47.148.0/24	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	194.145.188.224/27	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	194.145.189.52/31	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	194.145.190.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	194.145.191.128/27	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	195.74.60.0/23	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	212.11.68.144/28	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	213.166.2.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	213.166.3.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	213.166.4.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	213.166.5.0/24	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	78.40.243.192/27	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	87.238.72.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	87.238.73.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	87.238.74.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A fpbxreject -s	87.238.77.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT

Rules for IPtables file

-A INPUT -s 109.224.232.0/22 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s 109.224.240.0/22 -p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	109.224.222.16/28	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	109.224.232.0/22	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	109.224.240.0/22	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	109.239.96.132/31	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	141.170.24.20/30	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	141.170.24.5/31	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	141.170.50.16/28	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	185.47.148.0/24	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	194.145.188.224/27	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	194.145.189.52/31	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	194.145.190.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	194.145.191.128/27	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	195.74.60.0/23	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	212.11.68.144/28	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	213.166.2.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	213.166.3.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	213.166.4.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	213.166.5.0/24	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	78.40.243.192/27	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	87.238.72.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	87.238.73.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	87.238.74.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
-A INPUT -s	87.238.77.128/26	-p udp -m udp --dport 4569:5270 -j ACCEPT
Categories
Handsets Products and services Sangoma Phones

Sangoma P310 and P315

These phones are very competitively priced; and are perfect for anyone that needs reliable, basic calling functionality or needs a lot of phones either in one place or over a distributed set of large-facility locations. 

However, you can be rest assured that they are manufactured to very high standards and have all the qualities you would expect in more expensive models.

With these two new phones you can utilise essential Unified Communications features without sacrificing the function and performance that is usually only available with more expensive IP phones!

Want a closer look? Watch the first P315 being unboxed in Europe! Contact us for how you too can get your hands on the P310 or P315 and for bulk pricing.

Unboxing the P3xx handsets

Categories
Products and services

FreeStats Installation

Installing FreeStats requires a good working knowledge of Asterisk and Freepbx. Documented here is the step by step process.

 
 To install Asterisk FreeStats you need:

 Enable writing queue logs to the MySQL database through ODBC . (more: Asterisk: queue _log in MySQL via unixODBC )
 Set the username and password of the database in the Asterisk 

Call Center Stats config.
 Let's turn on the Asterisk queue log in the MySQL database 
 By default, queue statistics data is saved to the queue_log file. 
 Let's assign a MySQL database table for data storage - asteriskcdrdb.queuelog
 CREATE TABLE IF NOT EXISTS asteriskcdrdb.queuelog (
  id INT NOT NULL AUTO_INCREMENT,
  time char (32) DEFAULT NULL,
  callid char (64) DEFAULT NULL,
  queuename char (64) DEFAULT NULL,
  agent char (64) DEFAULT NULL,
  event char (32) DEFAULT NULL,
  data char (64) DEFAULT NULL,
  data1 char (64) DEFAULT NULL,
  data2 char (64) DEFAULT NULL,
  data3 char (64) DEFAULT NULL,
  data4 char (64) DEFAULT NULL,
  data5 char (64) DEFAULT NULL,
  PRIMARY KEY (id)
  ) ENGINE = MyISAM DEFAULT CHARSET = utf8;
 CREATE TABLE IF NOT EXISTS agents_new (
  id MEDIUMINT NOT NULL AUTO_INCREMENT,
  agent char (64) DEFAULT NULL,
  PRIMARY KEY (id)
  ) ENGINE = MyISAM DEFAULT CHARSET = utf8;
 CREATE TABLE IF NOT EXISTS queues_new (
  id MEDIUMINT NOT NULL AUTO_INCREMENT,
  queuename char (64) DEFAULT NULL,
  PRIMARY KEY (id)
  ) ENGINE = MyISAM DEFAULT CHARSET = utf8; 

We have included in the download a file to do this for you. to install it just run the following command
mysql -ufreepbxuser -pSECRECT  < ./freestats.sql
 
In the /etc/asterisk/asterisk.conf file , add to the options section:
 [options]
 queue_adaptive_realtime=yes
 
 edit /etc/asterisk/logger_general_custom.conf 
 and add
 
queue_adaptive_realtime = yes
queue_log_to_file = yes
queue_log => odbc,asteriskcdrdb,queuelog

In the realtime configuration file /etc/asterisk/extconfig_custom.conf, we define the driver, family, and target table:
[settings]
queue_log => odbc,asteriskcdrdb,queuelog 
; where asterisk is the db config, for example in /etc/asterisk/res_odbc_custom.conf

In freepbx the following are already set you just need to add the settings to /etc/asterisk/extconfig_custom.conf as above

If extconfig.conf does not exist, create it with the command: 
touch /etc/asterisk/extconfig.conf 
and set permissions:
chown asterisk.  /etc/asterisk/extconfig.conf 
/etc/asterisk/res_odbc_custom.conf 
[asteriskcdrdb]
enabled => yes
dsn => asteriskcdrdb
username => dbuser
password => dbpass
pre-connect => yes 

where is DSN , settings from file
/etc/odbc.ini 
[asteriskcdrdb]
Description = MySQL connection to 'asteriskcdrdb' database
driver = MySQL
server = localhost
database = asteriskcdrdb
Port = 3306
Socket = /var/lib/mysql/mysql.sock 
 
Apply configuration with 

asterisk -rx 'core restart now'
 
NOTE 'core restart now' - drops all current calls
Check the connection of Asterisk with the database through odbc :
asterisk -rx 'odbc show'
ODBC DSN Settings
Name:   asteriskcdrdb
DSN:    MySQL-asteriskcdrdb
Number of active connections: 3 (out of 5)
Logging: Disabled

Now move freestats and set permissions:
mv freestats /var/www/html/
Asterisk FreeStats config 
 
Edit the config.php file according to your data, where
 $ DBServer - host (localhost)
 $ DBUser - DB user.
 $ DBPass - DB Password.
 $ DBName - DB name.
 $ DBTable - The name of the database table.
 example:
 $ DBServer = 'localhost' ;
 $ DBUser = 'freepbxuser' ;
 $ DBPass = 'STRONG_SECRET' ;
 $ DBName = 'asteriskcdrdb' ;
 $ DBTable = 'queuelog' ; 

Open statistics in your favorite web browser:
http(s)://ip.address/freestats
To add agents and queues, click the appropriate buttons: 
Sync : 
The query will fetch the unique names of agents and queues and place them in the agents_new and queues_new tables . 
In order for entities to appear , there must be at least one entry in the queuelog table about the incoming call to the queue. 
In the future, if you have new agents or queues, simply synchronize the data to display statistics on them. Call data is saved regardless of synchronization, because the Asterisk queue logger is responsible for this.
FreePBX conversation recordings in the statistics interface 
To display conversation records in statistics, copy the following context into the /etc/asterisk/extensions_override_freepbx.conf file. 
The example shows the context [sub-record-check] of FreePBX Distro 14.
extensions_override_freepbx.conf
Or modify the context from your system in the same way by adding a custom queue event - REC . 
exten = > recordcheck, n, Queuelog ( recordcheck, $ {UNIQUEID} , NONE, REC, $ {CALLFILENAME} ) 
 You cannot edit the FreePBX context directly in the extensions_additional.conf file, but you can put the modified version in extensions_override_freepbx.conf
If you use Asterisk 'clean' , you can do the same in the context of calling the queue by adding instead of the ${CALLFILENAME} variable the name of the conversation recording file in accordance with the scheme you use. 
Additional information about queue log and QueueLog () command.
 
Authorization through accounts FreePBX 
To enable authorization with the password FreePBX, uncomment the following code in the config.php file and change the password freepbxpassword to the password of your installation. The password can be viewed in the /etc/asterisk/res_odbc_additional.conf file (for the latest freepbx versions).

Configure AMI and AJAM for Realtime module 
The php-curl package must be installed. 
The ajam_cookie file must be created in the root directory of the application:
touch ajam_cookie && chmod 777 ajam_cookie 
Asterisk  needs to have AJAM  enabled  /etc/asterisk/manager.conf
as below in freepbx go to advanced settings and set "Enable Static Content" to yes 
[general]
 enabled = yes
 port = 5038
 bindaddr = 0.0.0.0
  webenabled = yes
  httptimeout = 60 
 create user AMI
 [ajamuser]
  secret = PASSWORD
  deny = 0.0.0.0/0.0.0.0
  permit = 127.0.0.1/255.255.255.0
  read = system, agent, reporting
  write = system, agent, reporting 
 
and enable the built-in http server in 
 
/etc/asterisk/http.conf
 [general]
  enabled = yes
  enablestatic = yes
  bindaddr = 0.0.0.0
  bindport = 8088
  prefix = asterisk 
 To test AJAM Interface is active 
  freepbx * CLI> http show status 
  HTTP Server Status:
  Prefix: / asterisk
  Server: Asterisk / 15.5.0
  Server Enabled and Bound to 0.0.0.0:8088
 HTTPS Server Enabled and Bound to [::]: 8089
 Enabled URI's:
  / asterisk / httpstatus => Asterisk HTTP General Status
  / asterisk / amanager => HTML Manager Event Interface w / Digest authentication
  / asterisk / arawman => Raw HTTP Manager Event Interface w / Digest authentication
  / asterisk / manager => HTML Manager Event Interface
  / asterisk / rawman => Raw HTTP Manager Event Interface
  / asterisk / static / … => Asterisk HTTP Static Delivery
  / asterisk / amxml => XML Manager Event Interface w / Digest authentication
  / asterisk / mxml => XML Manager Event Interface
  / asterisk / ws => Asterisk HTTP WebSocket
 Enabled Redirects:
    None.
 Finally, set the data to connect to AMI / AJAM in the application config.php file :
 $ config [ 'urlraw' ] = 'http://127.0.0.1:8088/asterisk/rawman' ;
  $ config [ 'admin' ] = 'ajamuser' ;
  $ config [ 'secret' ] = 'PASSWORD' ;
  $ config [ 'authtype' ] = 'plaintext' ;
  $ config [ 'cookiefile' ] = null ;
  $ config [ 'debug' ] = false ; 
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
Knowledge Base Products and services

Aastra 6753i Transfer

Step By step instructions for call transfer when using the Aastra 6753i with firmware 3.x.x and above.

Phone Idle screen.

Once a call is answered their number will show and an icon of a ‘off hook phone’ will also show

To transfer the call press your ‘Transfer key’. Another ‘line’ will show numbered 2 with a ‘ > ’ next to it.

Enter the number you want to dial and press ‘>‘ dial if the call isn’t immediately dialed.

To ‘Blind’ transfer the call press the Transfer Button again or put the Handset down. NOTE if you do this you will not be able to get the call back.

After pressing dial the Phone Icon will show ‘ringing’

To get the call back while it is ringing press the ‘ < ‘ button shown on the display next to ‘Cancel’. Then L1 in this example will flash and ‘call held’ will show on the display as below, you need to get the call back by pressing the Flashing Line Key.

If the call goes to Voicemail or the caller answers the display will show the ‘off hook’ icon against 2

If the Caller wants the call then Press the ‘Transfer key’ the Red ‘Hangup key’ or put the handset down and the call will be transferred to them. Do not press the ‘>‘ Drop button.

If they don’t want the call or it goes to voicemail and you want to get the caller back, Press the ‘ > ‘ Drop Button and that call will be dropped and as before ‘call held’ will show on the screen and you press the L1 button to get the caller Back

Categories
Handsets

Gigaset SL750 Pro

SL750H £84.00

The SL750H PRO handset is the thinnest pro handset yet and has a large 2.4″ illuminated display with an intuitive, icon-based user interface. This handset delivers fantastic call clarity based on Gigaset’s patented HDSPTM audio quality. The SL750H allows users to easily control call and notification alerts with a choice of audio profiles to suit your business needs.

The SL750H PRO has been given a special coating that makes it resilient to scratches on the display, handset or keys. This coating also helps to protect it from disinfectants, making it an ideal choice for industrial and manufacturing companies, hospitals and similar environments.

This handset is fully compatible with both Gigaset N510IP PRO and N300IP singlecell solutions as well as the N720IP PRO multicell DECT solution.

Categories
Covid-19 System Status

Covid 19 update June 2020

Coronavirus: service updates and supplier information

03 APRIL 2020

Last updated: 17:00 5 June 2020

This page contains important updates from us and our suppliers that may affect our partners or customers during this unprecedented time. We’ve also included useful links and support information and a full breakdown of any service updates as we have them. This will continue to be updated at least once a day, so please check back here regularly.

Key points:

BT Openreach (who help us supply internet connectivity services) have advised that provision and repair of their services are now on a “best endeavours” basis only across the UK and will impact almost all connectivity products: DSL, Ethernet, FTTC, FTTP, FIBRE, PSTN, ISDN2, ISDN30, LLU and SMPF

Openreach will be assisting providers to bring forward certain WLR and MPF new line installs to an earlier date. They will be contacting customers directly to try to agree an earlier install date to complete the work on behalf of the Provider.

Priority restoration of services will only take place for organisations deemed “critical”, including NHS, pharmacies, utilities etc (full list below)

Ongoing repairs will continue for all, but with priority given to critical organisations and customers deemed “at risk” and you will now be asked if you are an “At Risk Customer” or “suffering COVID-19 Symptoms” before they assess priority of restoration

All newly-appointed provisioning will be pushed back to Monday 1st June 2020 at the earliest

All other Gradwell products (Wave, 3CX, SIP Trunks etc) remain unaffected with the exception of number porting, which may be delayed

Openreach/BT have declared that the current situation they and we are all facing is a matter beyond their control in light of COVID-19.

As such, appointments for provision and repair are on a best endeavours basis only, this covers everyone nationwide and will impact the following products that we provide: DSL, Ethernet, FTTC, FTTP, FIBRE, PSTN, ISDN2, ISDN30, LLU MPF and SMPF, including adjustment of orders.

Fault Repair

During this time, priority and restoration of services will take place for critical functions only, such as NHS, pharmacies, utilities, emergency services, retail and wholesale food distribution outlets and financial services. This also applies to customers deemed “at risk”.

Ongoing repairs to the BT/Openreach infrastructure will continue, but with high priority focus being on the above-mentioned critical core services ahead of all others. You will be asked by Gradwell if anyone is deemed within the building or location is considered at “at risk” of exposure to Covid-19:
Customer meets criteria for repair and IS symptomatic
Customer meets criteria for repair and IS NOT symptomatic

Ongoing repairs to the BT/Openreach infrastructure will continue, but with high priority focus being on the above-mentioned critical core services ahead of all others. We appreciate your patience during these times, and we’re sure you understand that the protection and restoration of core services is paramount to sustaining core infrastructure services.

Openreach have advised on Fault repair with immediate effect: 

To support efforts to reduce human contact and promote social distancing, all installation activities that require access to a customer’s premises are now being assessed and the customer to state the following criteria to ensure if repairs are deemed essential that appropriate teams can deal with the issue at hand:

  • Customer meets criteria for repair and IS symptomatic
  • Customer meets criteria for repair and IS NOT symptomatic

Openreach will attempt to complete as much of the inflight orders from outside of the premises as possible, but if internal work is required, these will be done on a priority and risk assessed basis. Any orders that do not require on premise installation will proceed as normal, but we appreciate these cases are limited.
The exception to these rules are those businesses identified as critical national infrastructure. This includes; NHS, pharmacies, utilities, emergency services, retail and wholesale food distribution outlets, and financial services businesses. On site installation work will be permitted for these businesses but we will have to justify the criticality of the service to Openreach.

The Openreach Emergency Helpdesk will remain open. However, all business as usual contact centres have now closed.

Porting of Numbers

Following on from previous updates on BT’s porting backlog, we are pleased to announce this has now been cleared and we are able to place new orders again. There still may be a slight delay in contrast to normal timescales, as it’s likely that BT and our direct suppliers will receive a large amount of new port requests all at once, but this will likely only add on 24-48 hours to the normal processing time. In conclusion, we will not be operating at our normal timescales for another few days, but we anticipate everything to be back to normal in the short term. We appreciate your patience during this time.

New Orders

All new orders received with immediate effect that require BT or Openreach infrastructure (the majority) are being moved to commence from 1st June 2020 onwards. This will be under constant review and may have further delays should the situation worsen. Exceptions apply for customers deemed “at risk”.

All existing orders in the system with BT or Openreach will continue to progress, but with a view to completing the orders as soon as possible and may be delivered well outside of expected and industry standard timelines.

We are still accepting new orders, however, there will be a delivery backlog for any products requiring BT or Openreach provision until at least 1st June 2020 onwards. All other Gradwell products, eg Wave, Multi User VoIP, 3CX and SIP Trunk services, are currently not impacted and delivery times will remain unaffected.

Openreach have announced the below with regards to WLR and MPF new line installs.

Openreach will be assisting providers to bring forward certain new line installs to an earlier date. They will be contacting customers directly to try to agree an earlier install date to complete the work on behalf of the Provider.

Only lines that have relatively simple installation journeys will be eligible for this support. Orders that are deemed to have complex delivery that would require additional engineering work or orders where Openreach feel that it would not be possible to gain access to the premises safely in adherence to their COVID-19 safe working practices will be excluded from this process. For example, orders that require entry to blocks of flats etc will be out of scope.

Core infrastructure service providers

We will of course continue to share more information from core suppliers like BT Openreach on this page as soon as we have it. A full breakdown of Openreach’s current service status is in the table below.

Openreach Critical Functions
DamageOpen (limited service)
WelfareOpen (limited service)
DSOOpen (limited service)
ProductRepairProvision
Escalations (EPOC)No new work acceptedNo new work accepted
Premium Business HelpdeskNo new work acceptedNo new work accepted
WLRBAU if non-appointedBAU if non-appointed
LLUBAU if non-appointedBAU if non-appointed
FTTCBAU if non-appointedBAU if non-appointed
SOGEABAU if non-appointedBAU if non-appointed
GFASTBAU if non-appointedBAU if non-appointed
SOGFASTBAU if non-appointedBAU if non-appointed
Migration ServicesBAU if non-appointedBAU if non-appointed
ISDNBAU if non-appointedBAU if non-appointed
FTTPBAU if non-appointedBAU if non-appointed
EthernetBAU if non-appointedBAU up to curtilage

BAU: For Total Loss of Service Faults: Openreach will attempt to activate the service remotely. If they are unable to, then they will visit the premises and ask two risk assessment questions:

1) Is there anyone on-site who has Covid-19 symptoms?

2) Has there ever been anyone on-site with symptoms?

3) Has anyone on-site been advised by the NHS via a letter to isolate for 12 weeks?

If the answer is yes to either of these questions then they will not access the premises and the fault repair will be postponed until restrictions have been lifted. If not, they will schedule a repair.

For all other repair faults, Openreach will only resolve if it’s possible to fix remotely and without access to the premises.

Provisions: Will only attempt to access the premises if the provider is classed as CNI/welfare/at risk/specific key worker (see below).

CNI: refers to “Critical Network Infrastructure.”

All Welfare/At Risk/Specific Key Worker cases will be reviewed by Openreach after a case submission by a customer.

Categories
Covid-19

New 119 Covid line number

Ofcom announced this afternoon the new 3-digit telephone number ‘119’ to be used as part of the NHS’s Covid-19 response, the number will be used as part of the NHS’s effort to handle the response to the Coronavirus situation.

The 119 number is supported by networks and so calls will connect as soon as the Government announce the service has been launched.

The number will be free to dial, in line with the NHS 111 service, as directed by Ofcom.

Categories
Applications Remote Working Services Special Offers

Sangoma Meet, Free multi-party, video conferencing and desktop sharing

Sangoma Technologies announced the general availability of its new video conferencing and collaboration service Sangoma Meet. Sangoma Meet is a multi-party, video conferencing and desktop sharing, cloud-based service, specially designed to enable a great work from home or collaboration experience. HD video support is standard with Sangoma Meet, ensuring the highest quality video calls possible. 

Sangoma Meet is also highly secure, fully encrypting password-protected video conferences to avoid problems with unauthorised users hacking into a video meeting, and Sangoma will not rent or sell Sangoma Meet user information to any third parties, including any social networks. It can also be used as a stand-alone service or in conjunction with its Unified Communication Business Phone Systems. 

Connect

Its simple to use with all the features you need

Moderator Privileges: If you are holding a team call, you have full moderator privileges. You can mute/unmute attendees, send them private messages, or kick anyone off the call. You also have the option to invite audio-only participants to limit video sharing.

Chat: If you prefer not to talk on your video call, use the chat feature in Sangoma Meet to share your thoughts with the group. A full suite of emojis will keep the conversation engaging and light!
Screen Sharing: Give your coworkers real-time examples and streamline meetings with screen sharing in Sangoma Meet. Instantly share your screen and choose what you want to show to others.
Inviting users: Its as simple as, copy your meeting link and send it to the desired parties. As soon as  users click on the link, they immediately join the conference.

Sangoma Meet is available for use today in the company’s General Availability release. It has already been in beta usage, both internally and with partners, with excellent reviews to date.

The Sangoma Meet cloud service has been updated to include additional functionality such as mobile apps (for iOS and Android), calendar integration (for Google G-suite and Microsoft Outlook), desktops apps (for both Windows and Mac) to further enhance user experience, and dial-in functionality to the video conference using a phone (with enhanced security via a PIN).

To help its end-users communicate more effectively during this global pandemic, Sangoma decided to launch this new service completely free of charge to everyone, during these troubled times. 

The full datasheet explaining all the features can be viewed here and the help site is located here