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
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 Elastix Support

Elastix changes and what it means

This week, significant changes at Elastix were announced, including the involvement of 3CX and the removal of key Elastix versions for download. Since those announcements, many things have been written by many people, and this has left some folks wondering what happened. Sangoma would like to reinforce its commitment to open source, this open letter from Sangoma, will provide our own clarity about how these events affect or involve Sangoma. Sangoma are a professional, global, growing, profitable, engineering-focused, publicly traded company, and this is the only reliable source of information to understand how those recent events affect or involve Sangoma. Other commentary released by other third parties about Sangoma, is not to be relied upon.

Everyone comes to open source software for their own reasons: software developers to do what they love; some to earn a livelihood; manufacturers to augment the project and sell their wares; and most importantly community members to find flexible/cost effective/well-supported solutions to their ‘business problem’ (in our case, for UC/Telecom/PBX needs). In the end, the good projects build something bigger than themselves… a community, a solution, and an opportunity for end users to utilize the project to build their own businesses. Over the course of a project many people will enter and exit those communities as their needs change.

As the primary investor in and developers of FreePBX, Sangoma actively works with many different members of the Open Source Telephony (OST) community, including Asterisk Developers, other FreePBX-based distros (including Elastix!), and many third-party hardware/software developers and manufacturers. As just one example, we have a great relationship with Digium and talk with them on an almost weekly basis, even though many consider us competitors. This may seem surprising to some, as many folks would think we might be bitter enemies. In fact, the opposite is true…we encourage and help those products to compete in the marketplace on their own merits. And this is entirely consistent with the commitment Sangoma has demonstrated to open source for many, many years over the time when we worked hard to also make Asterisk better. When Sangoma took over stewardship of FreePBX, we reiterated this statement clearly and unequivocally.

So Sangoma continues to work very hard every day, and invests many millions of dollars each year, in order to build strong relationships and to benefit to the entire open source telephony community. There is a saying that ‘a rising tide lifts all boats.’ Thus, it is usually counter-productive for open source contributors to battle with each other. In other words, there is no reason for them to fight over the same slice of pie, when there is an entire cake that no one is touching.

Their approach was no different with Elastix. For over a decade, Sangoma has been a direct supporter of Elastix, in many, many different ways, visiting them in Ecuador many times. They supported the project financially, They attended/exhibited/supported/spoke at multiple ElastixWorld events over many years, They cooperated with their distribution partners who also supported Elastix, They invested in R&D to ensure their products (software and hardware) were compatible with Elastix, etc. The list goes on and on.They had (and hope, still have), excellent relationships between the companies, in all parts of the organizations right up to the CEO level of both companies.

With recent changes at Elastix, some people/blogs/websites have made comments which claim that the removal of Elastix downloads of version 4 or MT, was in some way caused by Sangoma/FreePBX, due to concerns about compliance with GPL conditions. That is not true and They wish to set the story straight.  Sangoma hold ourselves to high ethical standards, and as a publicly traded company as well, setting the record straight with facts and not rumours, is both important and required.

While it is indeed true that Sangoma pointed out to Elastix some time ago, that there was a copyright issue,They did so in a very friendly manner, with words carefully chosen to be respectful of the long term relationship between the companies, and critically, to ensure that this important relationship continued. It was a 2015 letter from CEO to CEO, and certainly did not suggest any legal action, since it was not that kind of letter at all…it was a positive, complementary letter seeking to deepen the relationship, not harm it. That letter was sent shortly after Sangoma acquired FreePBX, when they made it a priority to reach out to PaloSanto to reinforce that the Elastix Project was a valuable strategic partner to Sangoma. It was in no way threatening, did not ask for, was not intended to, and given it was 2015, did not cause any versions of Elastix to be withdrawn. Elastix decision this week to shutdown these versions is a business decision not a response to Sangoma. While it seems that these days, the number of open source projects that remain truly open source is definitely on the decline, Sangoma’s commitment to open source remains as true today, as always.

And while it is admittedly a little unusual for companies to do so, in this case, for full transparency to the open source communities that they respect so very much (and to dispel any untrue rumours or claims), the entire letter is available. They share it for those who need confirmation of the above statements, and to reassure the Elastix community that Sangoma continues to be committed to you as well as to the entire Latin America region (and would be honored to have you consider joining the family)

This page is a shorted and edited version of Sangoma’s announcement at https://www.freepbx.org/what-happened-to-elastix/  follow the link for the full version.

Categories
Knowledge Base Sangoma

Building FreePBX CallCenters

Leo D’Alessandro, Product Marketing Manager at Sangoma, and Frederic Dickey, VP of Product Management at Sangoma, will in this webinar explain how to build an efficient contact center cost-effectively with Sangoma’s FreePBX / PBXact UC.

In this webinar, you’ll learn how the many ways FreePBX / PBXact UC can solve your contact center requirements:

• How calls are best routed using call queues
• Maximizing Agent Productivity and Customer Satisfaction with automated Queue Callbacks
• Integration with desktop and CRM
• Monitoring live call metrics
• Reporting tools to analyze overall performance

WEBINAR: Building Your Contact Center with FreePBX / PBXact UC from Sangoma on Vimeo.

Categories
Asterisk Support Knowledge Base Security

Catching the IP of anonymous callers on Asterisk servers

Hi just sharing a simple bit of dialplan to catch anon callers ip addresses when using freepbx and Anonymous callers is set to yes, which is needed for some suppliers.

Normally I would say lock your firewall to only known IPs, but in some cases this isn’t possible

Im sure if you have a Asterisk server with a public IP you will have seen calls on the console screen where the call is to a destination but the callers are exten@yourserver . Well this little bit of dialplan at the end of you default sip context should catch them and log them with the ip of the originating server

In extensions_custom.conf add the dialplan below

[catchall]
exten => s,1,Noop(Dead calls rising)
exten => s,n,Set(uri=${SIPCHANINFO(uri)})
exten => s,n,Verbose(3,Unknown call from ${uri} to ${EXTEN})
exten => s,n,System(echo "[${STRFTIME(${EPOCH},,%b %d %H:%M:%S)}] SECURITY[] Unknown Call from ${CALLERIDNUM} to ${FROM_DID} IPdetails ${uri}" >> /var/log/asterisk/sipsec.log)
exten => s,n,Hangup()

Then in Custom Destinations add a destination as  catchall,s,1

so you now get in your logs

[May 1 00:11:06] SECURITY[] Unknown Call from  to 900441516014742 IPdetails sip:101@37.75.209.113:21896

 I hope this is some help to you, It allows other scripts to pick up this address and add it to your firewall.
Categories
Blog

Gigaset Pro Bundle Offer

gigsetpromo

Until further notice you can get a free N510IP PRO base when ordering three Gigaset pro handsets. To get this deal order one of the following bundles, If you want more handsets get in contact and we will put a budget beating price together for you.

Buy 3 Get the Base Free

 

  • N510IP PRO base and 3 x R650H PRO handsets
    • Normal RRP Price £322+vat     Our Price £190+vat
  • N510IP PRO base and 3 x S650H PRO handsets
    • Normal RRP Price £277+vat     Our Price £160+vat
  • N510IP PRO base and 3 x SL750H PRO handsets
    • Normal RRP Price £367+vat    Our Price £250+vat
  • All orders £10 UK p&p for NI please confirm p&p

Call 01225 580025 or email us to place an order.

Categories
IPPBXs Sangoma

The FreePBX Appliance

Introducing the FreePBX appliance! The FreePBX appliance is a purpose-built, high-performance PBX solution. Designed and rigorously tested for optimal performance, this is the only officially supported hardware solution for FreePBX. The appliance comes preloaded with the FreePBX Distro and includes a one-year warranty!

appliances-header

Featuring the FreePBX Distro, this appliance is an ideal fit for businesses looking to get more from a PBX. With millions of deployments throughout the world, FreePBX is relied upon daily by everyone from enterprises to startups. Leveraging the power of FreePBX has enabled businesses to grow while keeping communication expenses minimal. The FreePBX Distro has made deploying, configuring and using a PBX system easier than ever! With an easy-to-use GUI (Graphical User Interface), getting started is a breeze!

Current Models and System capacity:

The full datasheet can be downloaded from here

Categories
Knowledge Base Technical

Fortigate issues such as one way audio on Call Pickup With Hosted Asterisk and other problems.

We have noted that with some Fortigate routers and firewalls come with SIP helpers enabled by default.

The customer may initially not think that there is any issue and inbound and outbound calls work as expected, But we had noted on one customer site that when they did a call pickup on another phone that was ringing in the office they would not be able to hear the caller. The caller could hear them and if they put the call on and off hold they could speak normally.

On further  investigation with wireshark we noted that the RTP port changed when the pickup took place. We tested this on other sites not using the Fortigate hardware and did not have this issue.

Below are listed the commands to clear the SIP helper settings from the Fortigate hardware.

  1. Open the Fortigate CLI from the dashboard.
  2. Enter the following commands in FortiGate’s CLI:
    • config system settings
    • set sip-helper disable
    • set sip-nat-trace disable
    • reboot the device
  3. Reopen CLI and enter the following commands – do not enter the text after //:
    • config system session-helper
    • show    //locate the SIP entry, usually 12, but can vary.
    • delete 12     //or the number that you identified from the previous command.
  4. Disable RTP processing as follows:
    • config voip profile
    • edit default
    • config sip
    • set rtp disable
  5. And finally:
    • config system settings
    • set default-voip-alg-mode kernel-helper based
    • End

on a fortigate 200D the following is the method to use

Step 1) Removing the session helper.

Run the following commands:

config system session-helper
  show

Amongst the displayed settings will be one similar to the following example:

    edit 13
        set name sip
        set protocol 17
        set port 5060

In this example the next commands would be:

delete 13
end
Step 2) Change the default –voip –alg-mode.

Run the following commands:

config system settings
set default-voip-alg-mode kernel-helper based
end
Step 3) Either clear sessions or reboot to make sure changes take effect

a) To clear sessions

The command to clear sessions applies to ALL sessions unless a filter is applied, and therefore will interrupt traffic.

diagnose system session clear

Taken from

http://kb.fortinet.com/kb/documentLink.do?externalID=FD36405