Categories
Blog Handsets Products Sangoma Phones Software

Sangoma Connect Mobile

We know that right now, communication is key. We also know that you need to be able to work where it makes the most sense for you, so mobility is a must. Connect Mobile is here to help you communicate more effectively, It is the next-generation mobile softphone app for FreePBX and PBXact.

Sangoma Connect Mobile combines the best of our existing solutions with new core mobile technologies to provide a greater communication experience for calling and collaborating with coworkers.


It allows staff to work remotely using the Sangoma Connect Mobile
app for iOS and Android devices. Employees can enjoy voice, SMS, and multi-party video collaboration*, using their business extension,
while keeping their personal device information private.
The intuitive onboarding process sends an auto-generated email to
each user upon activation from the PBX with simple instructions for
them to download the app with instant log-in. It’s as simple as that!
Secured Communications
Sangoma Connect users are protected by TLS/SRTP encryption
which means their phone calls are kept private while on cellular
or WiFi connections, and your business is secured from any
unauthorized connections.

Sangoma Connect Mobile provides rich voice and video calling for PBXact and FreePBX, on iOS and Android devices. We’ve made the onboarding experience very simple for system administrators to provision, and for users to install:

  • Audio & Video Calls using your Business Extension, Make and receive calls using your mobile device, while protecting your,personal phone number, using mobile data or WiFi connection.
  • Do-Not-Disturb (DND), Silence all inbound business calls when you don’t want to be disturbed using the scheduling feature for automatic behavior.
  • Voicemail, Receive business voicemail directly to the app, allowing users to quickly listen to calls they have missed.
  • QuickDial with BLF, Create a list of favorite contacts for easy access with BLFs (Busy Lamp Field) that also show the current status of your colleagues.
  • Powerful Call Control, Users can blind & Warm transfer calls, record and have 3-way conversations

And that’s it! It just works. Issues like certificate management, which can be a manual process with other apps, are handled automatically.

Another great feature is the integrated Sangoma Meet, the award-winning video conferencing platform within Sangoma Connect Mobile, making it super simple to create or join a multi-party video conference.

How to Get It

Sangoma Connect Mobile is included, free-of-charge, for all new and existing PBXact systems.  It is also free-of-charge to FreePBX customers with an active connect license, for the same amount of users already purchased. Simply perform an update on your system, and Sangoma Connect will be available within the User Management Module.

Purchasing Options

FreePBX customers can dowload a free 2 user 1 year licence, and those wishing to purchase a Sangoma Connect Mobile license can do so from us.  Sangoma have integrated Sangoma Connect Mobile with Connect Desktop , users can take advantage of even further collaboration tools for your MacOS or Windows computer. Simply purchase the Connect license from the portal store and choose to use Sangoma Connect Mobile, or both! If needed, here are some helpful instructions on how to purchase a license and how to install on your FreePBX system.

Download the Mobile Client

End users can download the iOS or Android app, for free, by searching for ‘Sangoma Connect’ in the respective app stores. A direct download link is also available within the invitation email end-users receive. Here is information on how to use the app.

Email or Call for current licence pricing

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
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
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
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.

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Knowledge Base

Setting up Postfix to use Office 365 mail

FreePBX uses centos 7 and postfix fom its mail delivery, normally this is fine unless the customer is using Office 365 mail then there can be delivery issues.

Firstly you will need to set up a user in Office 365 for the system.

Postfix’s main configuration file is main.cf and that is where we make the required change as follow:

[root@localhost ~]# vi /etc/postfix/main.cf

Append the following lines

masquerade_domains = domainname
myhostname = USERNAME.domainname
mydomain = USERNAME.domainname
myorigin = USERNAME@domainname
relayhost = [smtp.office365.com]:587

mynetworks = 127.0.0.0/8
inet_interfaces = loopback-only
smtp_use_tls = yes
smtp_always_send_ehlo = yes
smtp_sasl_auth_enable = yes
smtp_sasl_password_maps = hash:/etc/postfix/sasl_passwd
smtp_sasl_security_options = noanonymous
smtp_sasl_tls_security_options = noanonymous
smtp_tls_security_level = encrypt
smtp_generic_maps = hash:/etc/postfix/generic 
# smtp_tls_CAfile = /etc/ssl/certs/ca-certificates.crt

Save and exit from the file.
Next we need to edit the configuration for the postfix SASL credentials:

[root@localhost ~]# vi /etc/postfix/sasl_passwd

Add a line below

[smtp.office365.com]:587 user@domainname:password

Replacing user@domainname:password with your sender account details

Save and exit from file

A Postfix lookup table must now be generated from the sasl_passwd text file by running the following command.

[root@localhost ~]# postmap /etc/postfix/sasl_passwd

Now change permission for this file

[root@localhost ~]# chown root:postfix /etc/postfix/sasl_passwd

[root@localhost ~]# chmod 640 /etc/postfix/sasl_passwd

Next, we need to configure generic file in order to be able to send emails as a valid user (this is required for Office365).

[root@localhost ~]# vi /etc/postfix/generic

Go the end of file and append following lines.

root@localhost.localdomain UserName@Domain.com

Again replacing localhost.localdomain and UserName@Domain.com with your service hostname and the email user are using

Save and exit from file.

Next let’s correct the file permission.

[root@localhost ~]# chown root:root /etc/postfix/generic

[root@localhost ~]# chmod 0600 /etc/postfix/generic

[root@localhost ~]# postmap /etc/postfix/generic

Now restart Postfix service.

[root@localhost ~]# systemctl restart postfix

Now try to send a test email using the command below:
FOR Centos:

echo "This is the body of the email"| mail -r"Sender-Display-Name<sender@domain.com>" -s "This is the subject(E-Mail from SMTP Relay) line" recipeat@gmail.com

In FreePBX under Voicemail admin you must change the senders address to match your account as well as the sender for notifications such as backups etc. otherwise you can get errors and mail wont be delivered.