Promotional offer: Gigaset pro multi-cell DECT solution

We have a great bundle offer available for our reseller partners giving you a FREE DECT Manager when you buy 4 of more Gigaset N720 IP DECT base stations. To take advantage of this bundle you will need to contact the ProVu sales team.

The Gigaset N720 IP DECT multi-cell solution offers its users seamless handover for uninterrupted calls – perfect for large areas and multi storey offices. Gigaset’s multi-cell systems are expandable for up to 30 base stations, enabling use for up to 100 handsets. Plus with a great choice of affordable handsets this solution is very flexible, allowing you to meet both the needs and personal preferences of your customers.

Sangoma PBXact UC – Enabling Phone Apps

Sangoma’s PBXact UC systems enable you to remotely manage deployed phones from your computer, but did you know there’s also a range of Apps available, designed to enhance the capabilities of Sangoma IP phones? We’ve put the following step-by-step guide together to enable you to discover how you can start utilising the great features available.

Please note that this guide shows you how to enable Phone Apps, it doesn’t show the process of creating an extension. If you would like to find out how to create an extension, please see our earlier blog post on how to create an extension using PBXact

Step 1:
You will need to start by web browsing to the IP address of your Sangoma PBXact UC system. If you don’t already know the IP address of the PBXact UC system, you may need to log into your router to view a list of active network devices.

Step 2:
Once you have browsed to the IP address of your PBXact UC system, you should see the following options:


You will need to go ahead and select the ‘PBX Admini
stration’ option. You should then be prompted to enter a username and password in order to login to the main interface of the system.

If you don’t know the username and password, you may need to consult an administrator or relevant person to help you with this.

Step 3:
Providing you have entered the correct username and password, you should now see the homepage showing the ‘System Overview’, ‘PBXact Statistics’, ‘Uptime’ and options at the top of the page.

You can either hover over ‘Modules’ located at the top of the screen in ‘Advanced Mode’ and select ‘User Management’, or use the magnifying glass on the right hand side and search for ‘User Management’ and select the option when it appears below the search field.

Step 4:
You should now see the ‘User Manager’ screen which may list users below if you already have users registered on the system.
Your screen should look like the screenshot to the right.

Step 5:
Please note that this step will involve creating a new user. You will need to click on the ‘Add’ button as shown in the screenshot above.


This should then take you to the screenshot shown above. You will then need to enter a ‘Login Name’, ‘Password’ and if you are linking this to an extension that already exists on the system, you will need to select the ‘Primary Linked Extension’ which should display a drop down list of extensions. Please note that extensions already assigned to other users will not be displayed in the drop down list.

Step 6:
Using the navigation bar under ‘Add User’, you will need to navigate to the ‘Phone Apps’ section, as can be seen below.

When you navigate to ‘Phone Apps’, you should see the ‘General’ section underneath and a section called ‘Allow Access’. You will need to change this option to ‘Yes’. This allows access to the Phone Apps for the user you are editing.

Step 7:
Depending on which Phone Apps you would like to set up for the user, you will need to navigate across the different tabs and enable the applications that you would like to assign to that particular user.

If you want to enable access to call queues, you will need to navigate to ‘Queues’ and change ‘Enable Queue Access’ to ‘Yes’. If you would also like to setup queue agent access, you will need to change ‘Enable Queue Agent Access’ to ‘Yes’. You will then also need to select the queues from a list under ‘Allowed Queues’. Please see the following screenshot below that shows this:


As you can see from the screenshot above, there is ‘TestQueue <6000>’ is shown in the list under ‘Allowed Queues’. Please note that you must have created a queue before hand in order for it to be shown in the list.

Step 8:
You may want to enable the login and logout feature for that user so that they can log out of the Sangoma IP phone once they have finished using it. In order to do this, you will need to navigate to ‘Other Apps’ and change ‘Enabled Login/Logout Access’ to ‘Yes’. Please see below a screenshot that shows this:

You may also want to allow Do Not Disturb (DND) depending if this is required or not. You can simply change the options that you would like to enable to ‘Yes’. Please note that additional configuration may be required when enabling certain Phone Apps.

Step 9:
Once you have finished configuring the user with Phone Apps, you can go ahead and submit your changes by pressing the ‘Submit’ button located near the bottom right hand side of the page.

You should then be taken back to the ‘User Manager’ section which shows you a list of users. You will then need to click ‘Apply Config’ which should be displayed near the top right hand side of the page. This writes the changes made to the PBXact UC system.

That’s the end of the configuration process.

Providing you have a valid extension and user set up, you should now be able to see the Phone Apps on the phone once provisioned/configured correctly.

Exclusive money saving Snom 3CX bundles!


We’re delighted to bring our 3CX partners a brand new, exclusive bundle promotion!

Available until the end of May 2017 and whilst stocks last, you can now save up to £150 when you purchase a 3CX licence with any mix of Snom desk phones from ProVu – saving you £10.00 per phone!

3CX Bundle 1 Buy 5 Snom desk phones with a 3CX Licence and save £50.00

3CX Bundle 2 Buy 10 Snom desk phones with a 3CX Licence and save £100.00

3CX Bundle 3 Buy 15 Snom desk phones with a 3CX Licence and save £150.00

Simply call the ProVu sales team on 01484 840048 or email contact@provu.co.uk and quote the bundle code you require.

Please note: bundle pricing is only applicable to new 3CX purchases and is limited to a maximum of 15 desk phones per order. ProVu will require end user details for each bundle purchase.

SIP-TLS with the Panasonic TGP600

The Panasonic TGP-600 DECT phone supports encryption of SIP signalling and audio (RTP) using the common SIP-TLS and SRTP methods supported by many VoIP platforms.

Configuration is very simple.  In the SIP Settings page:

Important settings are:

  • Proxy Server Port, Registrar Server Port, Presence Server Port.  The standard port for encrypted SIP is 5061 (rather than 5060 for normal plain-text SIP).  This depends on your SIP platform.
  • Transport Protocol. Set this to TLS
  • TLS Mode.  Depends on your platform but SIP-TLS is what I am using with an Asterisk PBX

All other settings on that page are the as normal.  You might need to alter some of the SRTP settings for voice encryption, on the VoIP Settings page:

  • SRTP Mode. This also depends on your SIP platform but Asterisk doesn’t handle negotiation of encryption so if it is being used at all, you need to get the phone to always use it, not attempt to negotiate.  In that case, this setting is set to “SRTP”

Certificates

By default the Panasonic phone is set to accept all certificates (meaning that self-signed certificates will work OK).  You can provision the phones to verify the certificate if you want to using the setting SIP_TLS_VERIFY_1_=”1″.  You need to ensure that you have loaded the necessary root certificate beforehand.

Why use TLS & SRTP?

Security:  If you are able to sniff the traffic on someone’s network (e.g. using Wireshark or tcpdump) then you will capture any VoIP calls going on.  A tool such as Wireshark can be used to extract the audio from the RTP packets on the network.  The SIP packets can be read in plain-English and can be used to ascertain certain things such as what extension numbers there are, who is phoning different numbers etc…

If the SIP traffic is encrypted then no-one can see it other than the telephone and the SIP server at the other end (much like HTTPS used by secure websites).

If the RTP stream is encrypted then the audio cannot be extracted from the network without access to the SRTP keys generated on each call.  If you try this using Wireshark, the audio file you’ll get out of it will contain only white-noise.  Because the encryption keys for SRTP are generated on each call and send within the SIP packets, it would make no sense to use SRTP without encrypting the SIP packets as well.

Hiding SIP from Application Layer Gateways:  Routers with SIP-ALGs built into them are the biggest single cause of issues with SIP, things such as one-way audio, calls cutting out, calls failing to connect etc…. can all be caused by a SIP-ALG on a router.  The job of the ALG is to keep an eye out for SIP packets going past and then to modify them in an attempt to fix them up to work through NAT.  But they nearly always cause more problems than they solve.  A less obvious attraction to SIP-TLS is that if the SIP traffic is encrypted, then a SIP-ALG cannot possibly see any SIP traffic going through it and much less, make any modifications to it.  This can be very useful for remote phones talking to a hosted PBX or a central office PBX.

The latter advantage is the main reason I am seeing people interested in SIP-TLS or already using it, rather than it’s intended use which is for secure calling.

Heading to a location near you – Cisco’s Multiplatform SIP Phone Launch

Following the success of our Manchester and London Cisco Multiplatform SIP phone launch events, we are delighted to announce that we will be extending this event to an additional two locations. Join us in either Glasgow or Birmingham next month to discover more about this popular, upcoming range in our 7800 and 8800 Multiplatform phone launch event. Registration is free, simply select your preferred location to be there!

When & Where?

IET Glasgow
Tuesday 4th April, 13:00 – 16:30
IET Birmingham
Thursday 6th April, 13:00 – 16:30
IET Glasgow IET Birmingham
IET Glasgow – William’s Room,
14 St Enoch Square,
Glasgow,
G1 4DB
IET Birmingham – Boulton/Faraday Room,
80 Cambridge Street,
Birmingham,
B1 2NP

What will I learn?

Designed to give you an insightful introduction to the range, this event is the ideal opportunity to develop your knowledge on this increasingly popular range. Throughout the course of the afternoon you will discover more about Cisco’s new series of Multiplatform phones from Cisco’s Product Manager, Simon Brough plus other guest speakers.

Attendees will also receive a free* phone from the new range for testing and evaluation.

 

Register Now

 

*Free phones are limited to one per company who attend, companies who have previously received a free multiplatform phone from Cisco are not eligible. Free phone will be selected by Cisco and may be any phone from the 7800 or 8800 series.

Sangoma SBCs Webinar – Register now to join us!

Join us on Tuesday 21st March at 11 am when we will be teaming up with Sangoma to host a live webinar on how you can add security to your SIP networks and save costs through flexible routing with Sangoma SBCs.

Sangoma SBCs don’t only add crucial security to SIP networks, they also allow for flexible call routing to enable cost savings without network churn. With features such as TDM functionality, audio transcoding and NAT transversal, Sangoma SBCs can act as an all in one gateway for VoIP networks.

Register now for your opportunity to learn how SBCs allow you to securely attach any network device as well as how to utilise SBCs to provide flexible interconnects and save on call costs.

Register for your place now

Webinar: How to make 2N intercoms secure

Did you know 2N offer some of the most secure intercoms on the market? With a wide variety of additional security add-ons, you have the flexibility to create the features you require. We invite you to join us on Tuesday 14th March at 11:00 am when we will be taking a closer look at 2N’s security features, in our free webinar; how to make 2N intercoms secure.

Throughout the webinar we will be taking an in-depth look at 2N’s security range and will cover the following topics:

2N Access Commander Application

  • Introduction to the 2N range
  • The capabilities of 2N’s security products
  • 2N software products – features and benefits
  • The uses of Access Commander software
  • 2N bundle offer
  • Questions and answers

Register for your place now

Configure Door Release Button on Grandstream GXV3275 to work with 2N Helios IP Force

Did you know that you can configure a programmable key on the Grandstream GXV3275 that will allow you to unlock the door via a peer-to-peer call from the 2N Helios IP Force to the Grandstream GXV3275 from the press of a button? Please see the following guide below in order to find out how to do this:

Step 1:
The first step is to start by web browsing to the IP address of the Grandstream GXV3275 using a web browser of your choice e.g. Google Chrome, Firefox.

Step 2:
Providing you have entered the correct IP address, you should see the following page prompting you to login to the web interface of the phone:

If you don’t know the username and password to login, you may need to consult an IT administrator.

Step 3:
Providing you have entered the correct username and password, you should see a few different settings to choose from. You will need to navigate to ‘Account’ using the top navigation bar, then ‘Call Settings’ using the navigation bar to the left hand side.

Step 4:
You will need to scroll down to the ‘Programmable Keys’ section which can be found near the bottom of the interface. You will need to set ‘Key Mode’ to ‘Dial DTMF’ using the drop down list.

Step 5:
For ‘Name’, you will need to enter a suitable name that will be displayed on the door release button on the phone.

Step 6:
For ‘DTMF Content’, you will need to enter the code that is set under ‘Switch Codes’ on the web interface of the 2N Helios IP Force. The default switch code that is set on the IP Force is ’00’.

Step 7:
Because the Grandstream GXV3275 and IP Force are peer-to-peer calling, you won’t need to worry about the ‘Dial DTMF Condition’. The default option should be set to ‘Incoming/Outgoing Call’. You can go ahead and leave that as the selected option.

Step 8:
Once you have configured all of the above options, you can go ahead and hit the ‘Save’ button at the bottom of the page. You should then be taken to the top of the page prompted to apply the configuration changes.

Firmware/System Version Used:

Grandstream GXV3275 Android Phone: System Version: 1.0.3.144

2N Helios IP Force: Firmware: 2.18.1.27.8

ProVu’s Exclusive 2N Bundle Offer

From now until the end of April 2017, we’re giving our resellers the opportunity to purchase any Verso or Force intercom with a 2N tamper switch and receive a free gold licence, saving you £126.00!

Available with a variety of options, there’s a 2N intercom to suit you. The Force offers an exceptionally sturdy option, whereas the Verso offers a more stylish, modular option.

For increased security, 2N’s tamper switch can be configured to send out an alert to a security system if the intercom’s front plate is removed.

Gold licences enable you to unlock a range of enhanced video, audio, security, integration and NFC features, adding increased functionality to intercom units.

To take advantage of this great money-saving offer, simply call the ProVu sales team on 01484 840048 or email contact@provu.co.uk and quote ‘2N bundle offer’.

Yealink’s new T27G is now available to order

Yealink’s new T27G IP desk phone is now in stock and available to order from ProVu. This new phone offers an upgraded version of the T27P and will eventually become its replacement. We believe this will be a welcomed update with a new range of additional features offered at the same price.

What’s New?

  • Gigabit pass through
  • USB2.0 port (pending compatibility with Yealink’s BT40 and WF40 dongles)
  • Opus codes support for better audio quality
  • Device performance enhancement, faster response on the phone’s user interface

Up to date pricing and stock levels can be viewed on our reseller portal ProSys. If you do not have access to ProSys and would like to register for an account, please complete our ProSys account request form.