Tag: freePBX

Categories
IPPBXs Sangoma

PBXact

The PBXact Business phone system is a fully-featured IP-PBX designed with unified communication features for organizations needing mobility, productivity and collaboration capabilities. The PBXact business phone system comes with an extensive set of built-in Unified Communications features:

  • “Phone Apps”- Advanced productivity features controlled by the IP-Phone’s display
  • UCP web-based Dashboard allowing end-users to control phone settings, voicemail, conference rooms and more…
  • Zulu UC Desktop and Soft-phone Integration for mobility and productivity.
  • CRM Integration for screenpop and click to call
  • End-Point-Manager to auto-provisioning Sangoma IP-Phones and manage each user’s programmable buttons and features
  • Built-in VPN to guarantee security for your remote workers connecting to the corporate PBX

PBXact Feature Support Included in All Systems

Business Features

  • Flexible Time-Based Call Routing
  • Built-In Conference Bridge
  • Fax to E-mail
  • Hunt/Ring Groups
  • Music on Hold
  • Voicemail Blasting
  • Find Me / Follow Me Calling
  • Personal IVRs
  • Wake Up Calls
  • Support for Video Calling
  • Secure Communications (SRTP/TLS)
  • Announcements
  • Text to Speech
  • Calling Queues (ACD)
  • Interactive Voice Response (IVR)

Calling Features

  • Zulu UC Desktop Application – Outlook, Browser and Softphone Integration
  • Three-Way Calling Support
  • Voicemail
  • Voicemail to E-mail
  • Caller ID Support
  • Call Transfer
  • Call Recording
  • Do Not Disturb
  • Call Waiting
  • Call History / Call Detail Records
  • Call Event Logging
  • Speed Dials
  • Caller Blacklisting
  • Call Screening

Telephony Support

  • Open Standards Support for Multiple Protocols
  • SIP, IAX2
  • PRI, T1, E1, J1, R2, POTS/Analog, ISDN, GSM (Excludes PBXact 10)
  • WebRTC
  • Softphone Support
  • Specialty Device Support
  • Door Phones
  • Overhead Paging
  • Strobe Alerts
  • Paging Gateways
  • Voice Gateways
  • Failover Devices
  • Desktop/Mobile Phone Support

Administration

  • Upgrade System with Granular Control
  • Bulk Import Utilities (Trunks, Extensions, Users, DIDs)
  • Localization in both GUI and Sound Files for Multiple Languages
  • Backup and Restore Utilities
  • Custom Destination Administration
  • Web-based Config File Management When Needed
  • System Recording Management
  • GUI Controls for DNS, Network Settings, and More!

User Control Panel

  • Responsive GUI (Desktop, Tablet, and Mobile Device)
  • WebRTC Softphone
  • Call History (Details and Recording Playback / Download)
  • Contact Management
  • Presence Management
  • Conference Room Management
  • Settings Management
    • Find Me / Follow Me
    • Call Forwarding, Call Waiting, Do Not Disturb
    • Call Confirmation
  • Voicemail
    • Visual Voicemail – Playback and Management
    • Notification Options
    • Greetings Management

Add-ons

The Base Platform includes a base of system enhanced features (see chart below)

 Included in BaseAdditional Add-Ons
PBXact Enhanced Featuresx 
  Call Recording Reportsx 
  Class of Servicex 
  Conference Prox 
  Extension Routingx 
  Fax Prox 
  Park Prox 
  Page Prox 
  SysAdmin Prox 
  Voicemail Notifyx 
  Voicemail Reportsx 
  XMPP Prox 
  Phone Apps for Sangoma Phonesx 
connect mobile and deskx 
 EndPoint Manager x
High Availability x
Sangoma Property Manager x
Call Center Features  
Appointment Reminder  Outbound Calling Campaign  CallerID Management  Outbound Call Limiting  PinSet Pro  Queue Pro  Queue Reporting  Web CallBack x

Additional functionality can be added as needed:

  • High Availability (License Required per PBX Node, Excludes PBXact 40 & 60)
  • Call/Contact Center Features (Enhanced Call Center Functionality)
  • Operator Panel / Wall Boards
  • Third Party Phone Support (for Non-Sangoma IP Phones)

Download Brochure here 

Categories
Gigaset Handsets Products Special Offers

Gigaset Maxwell C DECT Desk Phone

 

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

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

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

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

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

Email or Call for current pricing and qty discounts

Categories
Knowledge Base

Connecting to Serial console ports with Macs

Many devices and servers still require connection to them with console cables. Sangoma IPPBX and SBCs for example.
I will cover here how to connect to then with a Mac as they do not have a serial port.

First you will need a USB serial console cable. These can be purchased cheaply from Amazon or ebay.
For example the “KUMEED FTDI RS232 USB to RJ45 Serial for Cisco Console Rollover Cable for Cisco Routers” costs £10.99 inc delivery and works with Windows and Macs

To connect to a console port you need a few bits of information, The port speed, in the case of Sangoma SBCs the is 115200. also you need the device address.

To get teh device address open a terminal window and type:

ls /dev/*usb*

you will be returned something like:

/dev/cu.usbserial-DN01YED6 /dev/tty.usbserial-DN01YED6

so now to connect to the console port you need to enter:

screen /dev/tty.usbserial-DN01YED6  115200

you should now be connected, and can interact as if on a ssh session.

to disconnect is not as simple as just closing the terminal window, as a screen session will still be running. to exit a screen session enter the following key combination.

ctrl a \ 

If you do close a terminal you can see if any sessions are active by opening a new terminal and entering:

screen -list

Something like below will be returned if a session is active.

There is a screen on:

5177.ttys000.Ians-MacBook-2 (Detached)

1 Socket in /var/folders/bl/7k0f_2695njbsqwx762kr_380000gn/T/.screen.

to reconnect type

screen -r

and you should reconnect.

then exit as normal with ctrl a \

Categories
Support

Zoiper Account and Server configuration

Enter details below to create Zoiper QR code that can be scanned from your smartphone
On entering you details you will be taken to a page with installation instructions.

Userrname

AuthName (Often same as username)

Password

Server hostname (host.domain.co.uk:5060)

 

You can download and purchase Zoiper Softphones from Here

Categories
Products Sangoma Software

Zulu UC – The Ultimate Desktop and Softphone integration for your Business

Zulu UC Desktop and softphone integration unifies the most popular business communication tools & applications enhancing user productivity and mobility. Designed specifically for FreePBX and PBXact phone systems, Zulu enables features such as:

  • Zulu Softphone enabling users to make/receive phone calls from their desktop or mobile stations, including Chat for team collaboration.
  • Faxing directly from the Zulu widget & softphone.
  • Click-to-call to make calls directly from your web browser and /or email client.
  • Call Pop for CRM and help desk integration.

Get Your Free Zulu 2 User FreePBX License, FreePBX Zulu UC 2 User Package is Free of Charge. Each User Package comes with 2 users good for 12 months.

Features

Click to Call

With Click-to-Call integration, users can instantly call any phone number that is seen on their web browser or MS Outlook client which a click!. Simply click on the phone number and Zulu will initiate an outbound call via the softphone client or your desk phone, whichever is with you at the time. Great for mobile users who come and go from their workstation.

Click-to-Call also recognizes extensions and phone number prefixes, so you never have to worry about having to modify the phone number or extension you wish to click to dial.

Call Pop

Ideal for CRM and Help Desk Integration, Call Pop will automatically open your desktop web browser on an inbound call with all the information of the caller. This feature helps users expedite phone calls and provide the caller with the best customer service experience.

* For additional CRM integration check out our CRM Link Module

Presence

Improve communication between staff members by allowing them to see each other’s presence via the Zulu Softphone. This feature can save your employees time by reaching out to members who they know are available to take their request.

Users can set their presence using a variety of pre-set statuses or create their own. And because Presence is server side, a user’s presence will be updated across all communication endpoints automatically too.

Zulu Softphone

At the Center of the Zulu is the all new softphone enabling users to take their office with them and never miss a call.

  • Make and receive phone calls using Desktop
  • Send and receive FAX*
  • Control Presence status which will update your status on all your devices
  • Flexible Calling Options- generate a phone call from either the client on your desktop or your desk phone. Great for mobile users who come and go from their workstation.

Faxing requires the Fax Pro Module

Chat

The Zulu UC Softphone features integrated Chat functionality so that staff members can communicate with each other more effectively. Features like 1-to-1 messaging, group chat, file transfer and auto-archiving will improve employee collaboration and improve business results. Finally, one tool to do it all!

*Compatibility
Operating System: Zulu UC is compatible with Windows, Mac and Linux operating Systems. Browsers: Click-to-call and Screen POP work with Firefox and Chrome (Safari coming soon).

Categories
Gateways Products

Vega VoIP digital gateways

The Most Resilient VoIP Digital Gateways in Their Class

Vega VoIP digital gateways are small appliances that seamlessly connect your legacy telephony infrastructure, made up of PRI (T1, E1) or BRI lines, to IP networks. They are great for businesses with legacy phone equipment (such as a TDM PBX) who want to connect to SIP trunking services without having to spend money altering their current network infrastructure. They are also great for businesses that are already VoIP enabled at the core (with an IP-PBX) that need PSTN connectivity and require a SIP-to-TDM converter. Simply place the Vega VoIP Digital Gateway at the edge of your network, plug in your existing internet cable for VoIP connectivity and E1,T1 or BRI cables from your phone system and let the Vega VoIP Digital Gateway automatically handle the SIP signalling and voice media conversion for seamless voice and T.38 Fax integration.

Advanced Web GUI
Features an intuitive Quick Wizard which does all the hard work for you for new deployments. Flexible dialplan to allow you to make your own routes, including automatic failure detection with failover routing.

Diagnostic Tools
Web GUI based PCAP tracing tool to capture full signaling and media, eliminating the need to connect equipment for line tracing, fully compatible with wireshark.

Low and High Density Models
The Vega 100G and Vega 200G are our low density models with a maximum capacity for 30 and 60 SIP-TDM simultaneous calls. The Vega 400G is our high density model and the most flexible field upgradable unit for a maximum capacity of 120 simultaneous SIP-TDM calls.

E1/T1 & BRI Interface
Each E1/T1 interface (for Vega 100G, 200G, 400G) and BRI interface (Vega 50 BRI) can be independently configured as network side or terminal side. The Vega gateway can therefore be connected to a PBX or the PSTN.

Built-in Local Survivability
In the event of a WAN failure, IP phones behind the Vega gateway can continue to call each other, be routed to a backup switch or connected directly to the PSTN.

Vega VoIP Digital Gateway Models

Vega VoIP Digital Gateways are one of the most reliable fault tolerant SIP-to-TDM media Gateways on the market, sized for your business needs. All Sangoma hardware carries a one year warranty with options to extend.

Vega 50 BRI

Sangoma’s Vega 50 BRI VoIP Digital Gateways are a 2-4-8 port BRI appliance for up to 16 simultaneous BRI calls

 

  • Web GUI for configuration and troubleshooting
  • Featuring Quick Wizard for rapid d
    eployment
  • Onboard DSP for media translation
  • Interoperable with most legacy and VoIP carriers worldwide
  • Advanced flexible call routing with automatic failover and bypass routing
  • Built in Local Suitability in the case of WAN failure

Vega 100G

Sangoma’s Vega 100G VoIP Digital Gateways are a single port T1/E1/PRI appliance supporting up to 30 simultaneous calls.

 

  • Web GUI for configuration and troubleshooting
  • Featuring Quick Wizard for rapid deployment
  • Onboard DSP for media translation
  • Interoperable with most legacy and VoIP carriers worldwide
  • Advanced flexible call routing with automatic failover and bypass routing
  • Built in Local Suitability in the case of WAN failure

Vega 200G

Sangoma’s Vega 200G VoIP Digital Gateways are a dual port T1/E1/PRI appliance supporting up to 60 simultaneous calls.

 

  • Web GUI for configuration and troubleshooting
  • Featuring Quick Wizard for rapid deployment
  • Onboard DSP for media translation
  • Interoperable with most legacy and VoIP carriers worldwide
  • Advanced flexible call routing with automatic failover and bypass routing
  • Built in Local Suitability in the case of WAN failure

Vega 400G

Sangoma’s Vega 400G VoIP Digital Gateways are a quad port T1/E1/PRI supporting up to 120 simultaneous calls.

 

  • Web GUI for configuration and troubleshooting
  • Field upgradable licensing
  • Dedicated bypass ports for High availability
  • Support for Private Wire or Point-to-Point applications
  • Onboard DSP for media translation
  • Interoperable with most legacy and VoIP carriers worldwide
  • Advanced flexible call routing with automatic failover and bypass routing

For me details see Here 

Categories
Peripherals

2N IP Door Intercoms comparison chart

2N offer a range of stylish solutions for door communication. The 2N® Helios IP door and security intercoms will ensure comfort for you and your visitors and with a range of different models and feature enhancing accessories there will be an option to suit your needs.

Due to the complexity and multiple configurations possible please call or email for pricing and options.

2N® Helios IP Uni 2N® Helios IP Vario 2N® Helios IP Verso 2N® Helios IP Force 2N® Helios IP Safety 2N® Helios IP Base
Uni Vario Verso Force Safety Base
Integrated camera No Optional Optional (HD) Optional

(Standard or HD)

 No yes
Buttons 1 or 2 up to 54 up to 146 1, 2 or 4 1 or 2 1 or 2
Keypad No Optional Optional Optional No No
Internal RFID card reader No Optional Optional Optional No Optional Card reader sold separately
NFC support No No Optional Optional No No NFC supported card reader and licence required
Display No Optional Optional No No No
Pictograms for visual signalling Optional No yes Optional No yes
Integrated electric switch yes yes yes yes yes yes
10W loud speaker No No No Optional Optional No
PoE yes yes yes yes yes yes
Tamper Switch yes No Optional Optional Optional yes Independent circuit control

Tamper switch sold separately
IP Rating IP54 IP53 with roof IP54 IP65/IP69K IP69K IP65 IP65 ~ on 1W speaker models

IP69 ~ on 10W speaker models
IK Rating IK10 IK07 IK08 IK10 IK10 IK07
Phone book entries 2 2000 2000 2000 2000 2000
Security Relay support yes yes yes yes yes yes Security relay sold separately
2N® Helios IP Eye & 2N® Mobile Video Support No yes yes yes No yes Only models equipped with camera
Categories
Asterisk Support Blog Design FreePBX Knowledge Base Software

G.729 Goes Royalty Free

G.729 – IMPORTANT INFORMATION

As of January 1, 2017 the patent terms of most Licensed Patents under the G.729 Consortium have expired.

With regard to the unexpired Licensed Copyrights and Licensed Patents of the G.729 Consortium Patent License Agreement, the Licensors of the G.729 Consortium, namely Orange SA, Nippon Telegraph and Telephone Corporation and Université de Sherbrooke (“Licensors”) have agreed to license the same under the existing terms on a royalty-free basis starting January 1, 2017.

For current Licensees of the G.729 Consortium Patent License Agreement, no reports and no payments will be due for Licensed Products Sold or otherwise distributed as of January 1, 2017.

For other companies selling G.729 compliant products and that are not current Licensees of the G.729 Consortium, there is no need to execute a G.729 Consortium Patent License Agreement since Licensors have agreed to license the unexpired Licensed Copyrights and Licensed Patents of the G.729 Consortium Patent License Agreement under the existing terms on a royalty-free basis starting January 1, 2017.

As soon as we hear how this is going to affect Digium Asterisk we will update here.

 

Categories
Blog Design FreePBX Knowledge Base

Voice recognition and Asterisk.

This is primarily about Googles new Cloud Speech API and Asterisk recordings.

Having worked on many Voice rec systems including Mitels attendant system, Oranges Wildfire virtual assistance and Lumenvox’s add on for Digium’s Asterisk system one thing none could do was transcribe speech such as voicemails and this is what people want. There was a startup in the UK called Spinvox  but as anyone knows this wasn’t all it seems and when I questioned them while working on a project they clammed up and withdrew our testing account and the rest is history as they say.

So now we are many years on and Google have their second API for this service. The first API was a little flaky to say the least and came up with some amusing translations. The cloud version is much better and does a good job with most voice and also can be localised.

So what have we done. Well we have mixed together some existing code we use and created a “mini voicemail” that records your message converts it to text saves it as a voicemail and emails the resultant Text and recording to you.  In the process we did find a few “gotchas” with the API for example a pause of more than a couple of seconds will result in the translation stopping there, also a big one is that the translation takes as long as the recording is, and the API has a 60 second limit. Both of these can be overcome by limiting the record time in Asterisk to 60 seconds and using sox to remove silence of more than a second.

exten => s,n,Record(catline/${UNIQUEID}.wav,3,60,kaq)

/usr/bin/sox /var/lib/asterisk/sounds/catline/${origdir}.wav ${PATH}${origmailbox}/INBOX/${FILENAME}.flac  lowpass -2 2500 silence -l 1 0.1 1% -1 0.8 1%

As you can see from these snippits of code above we have used variables where possible to that it can be incorporated easily with existing asterisk systems using GUIs such as Freepbx, We use the voicemail greetings that the user recorded and also use the email address thats linked with their mailbox for simplicity of management.

Now having Voicemails as text is nice but where it comes into its own is with structured mailboxes or simply put questionnaires where the caller is asked a number of predefined questions and these are recorded as one single voicemail. We already do this for some customers but they still have to have some one transcribe teh voicemail to text to input it. The quality of the Google translation means that soon they will be able to just copy the text over. Other applications are only limited by your imagination, Such as automated voice menus for Takeaways or Taxi firms.

To be Continued…HERE

Categories
Blog Knowledge Base

Do you hate having to use Module admin to update Freepbx

One of my pet hates is having to use module admin to update the Freepbx modules via the GUI. Its not a big deal but as we use SSH to connect to servers and then tunnels to connect to the GUI. Which is all fine unless you have multiple SSH sessions open and things get complicated..

So I have written a small “dirty” Bash script to prompt you through the fwconsole method of updating all or just one module of your choice.

#!/bin/bash
echo ssh freepbx update tool. 2016 cyber-cottage.eu
echo "Welcome"
echo "We will check for upgrades"

read -p "Do You want to check upgrade status of freepbx modules? (y/n) " RESP
if [ "$RESP" = "y" ]; then
  echo "Glad to hear it"
 fwconsole ma showupgrades
else
  exit
fi

echo "We will now apply all upgrades"

read -p "Do You want to upgrade all freepbx modules? (y/n) " RESP
if [ "$RESP" = "y" ]; then
  echo "Glad to hear it"
 fwconsole ma upgradeall
else
 echo "OK We will just upgrade the module you choose"
  read -p "Please enter the name of the module you want to upgrade " MODU
  echo "We Will Now Upgrade $MODU"
  fwconsole ma upgrade $MODU 
fi

read -p "Do You want to update permissions? (y/n) " RESP
if [ "$RESP" = "y" ]; then
 echo "Glad to hear it"
fwconsole chown
else
echo "Dont forget to apply changes on GUI then"
fi

read -p "Do You want to apply the changes? (y/n) " RESP
if [ "$RESP" = "y" ]; then
  echo "Glad to hear it"
 fwconsole reload
else
  echo "Dont forget to apply changes on GUI then"
  exit
fi

As I said it was quick and “dirty” but it does work and can save a bit of time.