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Telos VXs is a phone solution specifically designed for broadcast. Paired with an appropriate PBX or hosted VoIP provider, VXs can replace multiple standalone studio phone systems.
VXs allows you to buy just the number of hybrids needed for your application.
VXs is modular and scalable, capable of handling 60 or more studios, each with its own program-on-hold audio.
VXs is available both as a one-time buyout and annual license subscriptions.
Lines may be shared between studios, and in the case of cloud deployments, between different studio facilities.
VXs integrates seamlessly with all Axia consoles, providing automatically generated Livewire mix-minus channels and native GPIO commands and status indications.
VXs connects to any broadcast console using available Telos Alliance analog, AES/EBU, and mixed signal xNodes. Indications and call control are available using GPIO xNodes, and the GPIO ports of the mixed signal xNode.
VXs includes support for AES67, including SMPTE ST 2110-30 and SMPTE ST 2022-7 redundancy.
VXs works with the full range of VSet call controllers.
In addition to G.711, VXs supports the G.722 codec, which offers wider than normal caller bandwidth of 50-7000 Hz.
Fifth-generation Telos Adaptive Telephone Technology, including dynamic EQ, AGC, adjustable caller ducking, and audio dynamics processing by Omnia.
Wideband acoustic echo cancellation eliminates open-speaker feedback.
XScreen call-screening software from Broadcast Bionics is provided at no cost.
While you can control the VXs with call screening software and Axia consoles, most systems will include one or more Telos VSets. These are phone-like controllers that have handsets for off-air conversations. VSets are available in 6 or 12 line models.
VSets work like a traditional Telos call controller, for instant familiarity with operators.
The Telos VSet Desktop Controller provides visual line status indicators, and provides control for up to 12 lines in an attractive desktop enclosure.
The enclosureless version, called the VSet Console Controller, is typically integrated into legacy broadcast consoles using custom metalwork. A drawing package is available for custom applications. Many console manufacturers already have metalwork designed for the VSet Console Controller available for purchase.
Given the enduring popularity of the Telos VX platform, several different companies have developed call screening software packages that support it. You can learn more about them here: https://www.telosalliance.com/call-management-software-guide.
We’ve partnered with Broadcast Bionics to offer XScreen, a locally deployed call screening system with optional cloud integration. The free version is fully capable when compared to the call screening software supplied with legacy POTS based phone systems, but the paid version offers a number of new features many find useful. You can find more information here: https://www.bionics.co.uk/xscreen.cshtml.
Fraunhofer’s acoustic echo canceler algorithm solves the problem of callers hearing an echo of themselves when talent is interacting with them on a monitor speaker, rather than headphones. The AEC in the Telos VX is impressive, allowing very high loudspeaker volume with no noticeable echo transmitted to callers.
One thing to keep in mind is that VXs is not a complete phone system in and of itself. It is a collection of SIP endpoints that have to be registered to a local or cloud PBX, a compatible hosted VoIP provider, or a gateway device.
This diagram shows a generic installation that could be used standalone with one studio and an analog audio console. Two Ethernet ports are used on the VXs engine to isolate the networks that are present in this example. Earlier versions of the Telos VX platform had two Ethernet ports designated LAN and WAN. With VXs, you have complete control over how your interfaces are assigned. In this example, eno1 is assigned to SIP and eno2 is assigned to AoIP traffic, though you can route SIP and AoIP over the same interface if desired.
These days, calls can arrive at your VXs system multiple different ways. Each delivery method has its own set of positives and negatives. If VoIP is relatively new to you, we recommend that you consult with our Telos ProService Field Application Engineers or one of our worldwide partners to help you find a solution that will work well for your application.
To illustrate the point that VoIP telephony involves more than one component, this diagram shows a very simple, 4 line phone system. Here, the connection to the outside world is ordinary POTS lines from a traditional phone company. They are fed into a device called an FXO gateway, which converts them to SIP channels that this 4 line VoIP phone can peer to on a one-to-one basis.
You could do the same with VXs in place of the VoIP phone to create a very simple broadcast phone system without a PBX. (In this example, that system would be limited to 4 lines, because the depicted POTS gateway only supports 4 lines.)
While it’s unlikely you would want to do this in 2023, as inexpensive, high quality VoIP options abound, it demonstrates the point we discussed above - unlike the POTS based broadcast phone systems of the past, VXs operates like a modern VoIP phone and requires something more than a simple handoff from the local phone company to function.
While this may sound complicated, it also gives us tremendous flexibility. After calls from the PSTN (Public Switched Telephone Network) are converted to IP, either by a VoIP provider, or a gateway like the one above, they can be routed anywhere in the world, just like any other IP traffic.
Quality VoIP providers leverage this technology to deliver superior service with innovative features at a fraction of the cost of most traditional phone companies.
As of this writing, SIP is a mature technology and there are options at virtually every price point. It would be impossible to cover every potential provider’s capabilities, partly because they are constantly evolving.
That said, both Telos and our worldwide partners have deep experience in making VoIP work for broadcast. We can offer suggestions that will help you avoid challenges based on our experience with several different providers. Contact Telos or one of our worldwide partners for details.
Telos VXs accepts and generates many audio channels. Using Livewire+, it’s possible to make them all available through a single Ethernet interface, along with all GPIO commands and status indicators.
For facilities with Axia consoles or consoles that utilize AES67, this approach saves money and simplifies installation. When needed, analog and AES/EBU I/O can be provided via Telos Alliance xNodes.
Telos Alliance xNodes feature stereo I/O, but each stereo input or output can be configured for dual mono operation. For example, for a studio with two phone pots on a board, you can use a single L/R in and L/R out of an analog or AES/EBU xNode to drive both pots. (For AES, keep in mind that the L and R signals will enter and exit the xNode on a single cable pair for input and another single cable pair for output, so you will need some means of splitting then routing the audio post xNode.)
Make sure you account for a program on hold audio input for each studio when planning an xNode purchase.
Note: VXs is highly scalable and can generate many more channels of audio I/O than most Livewire devices. If you plan to run more than 43 channels of I/O (including program on hold and any channels used by Acoustic Echo Canceller) make sure you use a gigabit ethernet port on your Livewire switch.
Telos VXs can support multiple studios and can share lines between studios.
A “studio” includes all the I/O resources needed to support operation for one physical studio. This includes AoIP inputs/outputs, program on hold audio, GPIO actions, GPIO indications, and the acoustic echo canceller.
A “show” is a profile that assigns extensions to line buttons. With VXs, you can assign any “show” to any “studio.” Within a show profile, you can also define whether a particular line button is on a selectable channel, or assigned to a fixed channel. You can also determine whether each line is blocked or not when the “block all” feature is engaged on a VSet, and assign custom ringers to each line.
VXs supports what we call “selectable” and “fixed” lines. It’s possible to have a mix of each type in any studio.
If you’re in North America, you’ll most likely want to use “selectable” lines. Fixed lines are more common in European deployments. With fixed lines, a specific extension is always associated with a particular button on the VSet phone and a specific fader on the console. It’s as if each line has its own hybrid. This allows VIP numbers and hotlines to have fixed and dedicated console faders. Auto answer is also an option for fixed lines.
Multiple calls assigned to a single fader have individual hybrids and are actively conferenced within the VXs engine using an internal mix-minus. Calls assigned to different faders would normally be conferenced via a mix-minus of some kind within the mixing console.
Selectable lines use an operating style like most previous Telos phone systems, where there was a line selector before the hybrids, and any line could be assigned to any hybrid.
Most operators are used to these older systems, where you have column of buttons and pressing one takes a line and drops the previous caller. We have kept this operating style as the default for VXs. We have also kept the “lock” function that allows the operator to keep VIP callers on-air while answering and dropping additional calls.
Keep in mind that all faders (fixed or selectable) count against the number of channels you’ve licensed for VXs.
There is full documentation for the VSet series of controllers available on our website. This section is meant to be a quick reference on VSet operation. If you would like more information about VSet operation and setup, please refer to the product manual for the controller you have.
This is the first step to using VSet to control VXs. Press the VSet MENU button. The LCD will show various items. Select the studio and show you want, then exit by pressing the MENU button again.
Installation setup options like Setup and User Setup are protected by a delay. Whenever you want to access one of these advanced configuration menus, you will need to press and hold the button to enter those menus for at least 5 seconds before releasing. Upon releasing, you will be in the menu you selected.
Selecting the studio is done through the Engine Setup menu. While the show setting can be changed by any user from the main menu, the studio option is not expected to change often, typically only when a new VSet is installed.
VSet offers two modes of operation - talent and producer. Talent mode is used whenever the operator needs to put calls on the air. Producer mode is used by a producer to screen calls. In producer mode, calls cannot be put on-air and are protected from being dropped.
Press MENU then MODE to toggle between talent and producer mode. Press MENU again to exit the menu.
Each line has two associated buttons to the left of the LCD.
Pressing a left column button (the column with a phone handset in the middle) will put a ringing or held caller on the handset.
Press the right column button to put a held or ringing line on-air. This will drop any other calls, unless they are locked. To lock a particular caller and prevent them from being hung up while answering other calls, simply press that caller’s right column button after answering the line. A lock icon will appear on top of the on-air indicator on the display. To unlock that caller, simply press the right column button again to remove the lock.
Status indicator showing line is on-air.
The hold button places any answered call on hold.
The drop button drops any call that is active on the handset or on-air, unless that call is locked. To drop a locked call, press the associated line column button prior to pressing the drop button.
This button takes calls to air in this level of priority:
Longest waiting ready hold
Longest waiting hold
Longest ringing in
The producer can manually override these and assign priority as desired.
Pressing this button will cause all inactive lines to be dropped and blocked from accepting any calls. Pressing the button again releases the block and allows calls through.
When dialing out, the VSet will wait 3 seconds after the last digit is dialed before sending the call. (Keep in mind, the tones you hear aren’t actually dialing the call in a VoIP environment like VXs, they are just there for the sake of familiarity.) The number being dialed is transmitted digitally.
Pressing GO bypasses the 3 second timeout and sends the number being dialed immediately.
The 3 second delay can be adjusted by selecting MENU -> USER PREFERENCES -> AUTO DIAL TIMEOUT. It can be increased to 5 seconds, reduced to ½ second, or turned off entirely, which would require the GO button to be pressed every time you make an outbound call.
After selecting a line, press the GO button before any digits are dialed. The last dialed number appears in the line info field. If you want to continue with that number being dialed, simply wait for the auto dial timeout to elapse, or press the GO button again to dial immediately. If you do not want to dial that number, press the drop button before the auto dial timeout elapses.
Normally, when a call is taken, it goes to fader 1, but you can assign it to any of six configured console faders (as defined in the Studio configuration.)
To change the fader a particular call is assigned to, simply press the fader assignment button for the fader you want along the right side of the screen after answering a call, or before (or after) dialing a call. Note: The buttons along the right side of the screen change function depending on the state of the VSet. You will need to look at the screen to view your options.
You can upgrade the firmware on your VSet using a web browser and firmware update files from Telos. Firmware updates are available on the VSet product page at the Telos Alliance website. Log into the VSet you wish to upgrade by typing its IP address into your web browser. The username is "user" and there is no password, so leave it blank. In the firmware upgrade section of the page, press CHOOSE FILE and navigate to the file on your local PC. After selecting the file, press the UPGRADE button. Your VSet has two banks for storing firmware. The upgrade will be stored in whichever bank is currently inactive. To activate the new firmware, select the inactive bank and press CHANGE BANK. The VSet will reboot using the new firmware.
Status indicator showing line is on-air and locked.
Status indicator showing line is on hold.
Status indicator showing that the line has been screened and is on hold.
Status indicator showing line has been blocked by block all function.
Status indicator showing line is currently in use by another studio.
One of the many advantages of the Telos Alliance Axia line of broadcast mixing consoles is how quickly they integrate with other Livewire+ enabled products. Things like mix-minus feeds are generated automatically.
In this example, we have configured a studio with two selectable channels, and assigned Livewire channel numbers 2000 and 2001 to them. We left the Manual Backfeed / Input setting at their default option, which is “auto.” This generates the automatically configured backfeed for use with Axia consoles.
The program buss of this studio’s console is Livewire channel 2201, so we’ve entered that in the Input Channel field under Program On Hold. (We did not enter / 2,1 afterward. This was generated automatically by VXs after we pushed SAVE to commit these changes.)
For this example, we are using an Axia Altus virtual mixing console, but the process is very similar for all Axia consoles.
Inside the Sources section of your Axia console, create a new Phone type source.
Set the source name to whatever you wish.
Leave the source input type set to the default of Livewire.
All of the other parameters can be left at their default settings, or modified to fit your needs.
Press APPLY and you’re done! You’ve virtually “wired” your phone system to your console, including an automatically generated mix-minus with just a few easy steps.
Here’s a screenshot of the Axia Altus console we’ve just configured with this VXs. As you can see, a 6 line call controller was automatically generated by Altus for perfect integration with VXs. If you are using an Altus virtual console, make a special note of the last setting on the page - Hybrid for Telos Phone. Your options here are None, Fixed, and Selectable.
When set to "None" no call controller is added to the virtual console surface. This setting is ideal if you are using a standalone call controller such as a Telos VSet.
When set to "Selectable" a virtual call controller is added adjacent to the corresponding phone channel on the console.
When set to "Fixed" a RING button appears on that console fader while the fixed line/channel is ringing. Pressing the RING button answers the call. The button will then be labeled AIR. Pressing AIR will terminate the call.
You can integrate VXs with any type of broadcast audio console using Telos Alliance xNodes for audio I/O. In this example, we will be using a Mixed Signal xNode, which contains I/O for analog, AES/EBU, and GPIO.
These instructions assume you are configuring a new xNode for this application, which isn’t currently in use for anything else.
In this example, we have configured a studio with two selectable channels, and assigned Livewire channel numbers 15801 and 15802 to them.
We’re generating Livewire channels 21501 and 21502 on the xNode for send to caller audio, so we configure those channels here like this:
Note that we have changed the drop down menu to the right of the Address field from its default of To Source and made it From Source instead.
We’re using Livewire channel 21503 on the xNode for Program on Hold audio, so we’ve entered that in the Input Channel field under Program On Hold. (We did not enter / 2,1 afterward. This was generated automatically by VXs after we pushed SAVE to commit these changes.)
On the xNode’s Simple Setup page, select the Mono 8x8 operating mode and press APPLY.
Here, we have configured the sources that were defined on the VXs Studio page above.
We've used Stereo 1ms AES67 streams for interaction with callers where cumulative latency can be an issue, and a Standard Stereo (5ms) stream for the program on hold audio, where an extra 4ms of latency is not a concern.
We have configured the destinations that were defined on the VXs Studio page above.
Everything should now be ready to receive audio from and send audio to callers via the xNode. In this example, caller I/O is on analog output 1 of the xNode. The left channel is used for caller 1 I/O, and the right channel is used for caller 2.
If GPIO indications and actions are needed, go to the Simple Setup page of the Mixed Signal xNode and assign the ports available Livewire channels. Note: GPIO LW channels do not advertise to the network, so you will not see them in the browse window. In this example, we’ll continue with the numbering scheme we started earlier, by manually typing those channel numbers in here.
We left off at Livewire audio channel 21508, so we’re starting with 21509 for GPIO. It is possible to utilize Livewire audio channel numbers for GPIO as well, but keep in mind that Livewire audio channels have certain GPIO functions tied to them in certain applications inside Axia consoles, which may interfere with VXs GPIO. Therefore, we highly recommend using completely different Livewire channel numbers for VXs GPIO.
You can see that we have configured one GPIO Action and two GPIO Indications on Livewire channel 21509. In this example, anytime GPI pin 1 is pulled to ground, it will mute the VSet ringer for this studio. (A common application would be when a mic is on or a call is being recorded.)
In this example, GPO pin 1 goes low to indicate that a non-Busy All line is ringing, and GPIO pin 2 does when Block All is enabled for the studio.
Please consult the xNode manual for proper wiring of the GPIO ports: https://www.telosalliance.com/Axia/xNodes
While all Axia consoles and some other manufacturers’ consoles create an individual mix-minus for each caller channel, there are some which share a common mix-minus among all phone faders. In this circumstance, it’s necessary to create a separate mix-minus for each fader so the caller on fader 1 can hear the caller on fader 2 and vice-versa.
The Telos Alliance xNode has a matrix mixer that can be used to overcome this obstacle. One thing to keep in mind is the internal mixing will use additional sources in the xNode.
We’ve gone to the Sources page in the xNode and set up our input and “New Mix Minus” channels. In this example, Line 1-L is the main mix-minus input and Line 1-R is the Program on Hold input for the studio. Sources 7 and 8 are configured as the “New Mix Minus” channels.
After doing this, and pressing APPLY, we go to the Mixer page of the xNode. The xNode mixer routing is configured as shown here:
Afterward, return to the appropriate Studio page in VXs and make the necessary changes. In this example, the manual backfeed setting is 21507 for caller 1, and 21508 for caller 2. Make sure to save your changes afterward.
You can type the Livewire channel into the field manually, or bring up a list of all Livewire sources on the network by clicking the icon just to the right of the source selection field.
Telos 5th-generation Adaptive Digital Hybrids
Maximum number of hybrids: Hardware dependent
Maximum number of SIP numbers: Unlimited
Maximum active on-air calls: Hardware dependent
Maximum on-air calls on one fader: 12
All processing is performed at 32-bit floating-point resolution
Send AGC/limiter
Send filter
Gated receive AGC
Receive filter
Receive dynamic EQ
Ducker
Sample rate converter
Line echo canceller (hybrid)
Acoustic Echo Canceller
Via Livewire IP/Ethernet. Each selectable group and fixed line has a send and receive input/output
Each studio has a program-on-hold input
Each Acoustic Echo Canceller has two inputs (signal and reference) and one output
Livewire+™ AES67 equipped studios may take and supply audio directly to/from the network. Telos Alliance xNodes are available for analog and AES3 (AES/EBU) breakout.
Audio: Standard RTP
Codecs: G.711 u-Law and a-Law, and G.722
Control: Standard SIP endpoint
© 2023 TLS Corp. Telos Alliance® All Rights Reserved
Telos®, Axia, the Telos Systems logo, and Telos Alliance®, are trademarks of TLS Corp. All other trademarks are the property of their respective holders.
All versions, claims of compatibility, trademarks, etc. of hardware and software products not made by Telos Alliance which are mentioned in this manual or accompanying material are informational only. Telos Alliance makes no endorsement of any particular product for any purpose, nor claims any responsibility for operation or accuracy. We reserve the right to make improvements or changes in the products described in this manual that may affect the product specifications or to revise the manual without notice.
Telos VXs is a containerized software product. We routinely release new versions to add features and fix bugs. Check the Telos Alliance website at www.TelosAlliance.com for the latest. We encourage you to sign-up for the email notification service offered on the site.
We welcome feedback on any aspect of Telos VXs, or this manual. Many good ideas from users have made their way into software revisions or new products. Please contact us with your comments.
You may reach our 24/7 Support team anytime around the clock by calling +1-216-622-0247.
For billing questions or other non-emergency technical questions, call +1-216-241-7225 between 9:30 am to 5:00 PM, USA Eastern time, Monday through Friday.
Technical support is available at support@TelosAlliance.com.
All other questions, please email inquiry@TelosAllinace.com.
The Telos website has a variety of information that may be useful for product selection and support. The URL is www.TelosAlliance.com.
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+1-216-241-7225 (phone)
+1-216-241-4103 (fax)
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Telos VXs is a containerized broadcast phone system that allows you to deploy your system with maximum flexibility on-premises or in the cloud.
Telos VXs is a containerized product. It will require some initial configuration from a Telos ProServices Field Application Engineer or one of our worldwide partners. They will assist with deployment of the Docker container for your VXs engine.
When your VXs engine is up and running, log into it using the IP address provided by the container deployment team.
Because VXs is designed to be deployed on-prem or in the cloud, the login procedure is slightly different from other Telos Alliance devices with dedicated hardware. You will need to log in using your TelosCare ID. If you don’t yet have a TelosCare ID, you can register for one by clicking “Click here to register for your TelosCare ID” and filling out a simple form.
Once submitted, you will receive a verification code via the email address you used to register. You’ll need to enter the code to confirm your registration.
After you enter the correct verification code, you will be registered and allowed to log in.
Before provisioning the VXs networks, it’s important to make sure that you have the right kind of network switch for the AoIP interfaces. The switch you choose must be capable of handling multicast audio. If you use a switch that isn’t capable of properly routing multicast traffic (most unmanaged consumer grade switches) you will send all AoIP traffic to all ports on the switch, flooding your network, diminishing performance, and likely interrupting or crashing other devices that might be plugged into your network.
During deployment of the VXs container, your Telos ProService Field Application Engineer will have configured one or more network interfaces and set their IP addresses. You will now select which interface(s) handle each function - SIP or AoIP.
To set your AoIP interfaces, select the MAIN tab at the left of your browser window, and select at least one interface for use with Livewire+ or AES67 audio via the dropdown menus. In this example, our AoIP network is eno2 and has an IP address of 192.168.2.20. You will see the NIC’s MAC address after the IP address.
If needed, you can configure a second AoIP interface to enable SMPTE 2022-7 redundancy. If you don't have SMPTE 2022-7 requirements, do not specify a NIC for Interface B.
After making your selections, press the SAVE button in the Network Interface area. Confirm your selection in the box that appears, then press the RESTART VXS button in the System Control part of the screen to ensure that all network changes have been applied. Once VXs restarts, you’ll need to log back in using your TelosCare credentials.
When you’ve logged back in, select the SIP tab and choose a network interface for SIP traffic from the dropdown menu. In this example, our SIP network is eno1 and has an IP address of 192.168.1.20.
After making your selection, press SAVE and reboot VXs once again by going to MAIN and pressing the RESTART VXS button under System Control.
While it is most common for VSets to be connected to one of the AoIP networks defined on the MAIN tab, they can also be connected to the VoIP network defined on the SIP tab.
As of this writing, VXs supports the G.711 (alaw or ulaw) and G.722 codecs. G.711 is the codec used by the PSTN and features bandwidth of 300-3400 Hz.
G.722 is the same codec known to broadcasters from the pre-MPEG days of ISDN remotes. It has a wider bandwidth of 50-7000 Hz, offering much better than usual speech quality.
Not all VoIP providers support G.722, and those that do are limited to using it for calls internal to their own network, as the PSTN does not support G.722. G.722 can be useful in some circumstances though. For example, G.722 calls may be made from a mobile softphone client on the same provider as your VXs system to allow for easy remote broadcasts with better than PSTN quality audio from virtually any smartphone.
SIP always negotiates a codec supported by both ends, dropping to G.711 if a better option isn’t available at both ends of a call.
The SIP configuration page shows the global SIP settings, along with a list of all SIP servers that your VXs is configured to use. You can configure one or many SIP servers.
The Use SRV Lookups feature is not widely used and should be left turned off unless Telos support asks you to enable it.
Port is the TCP or UDP port used for SIP signaling between VXs and your PBX, or hosted provider, or gateway. Leave it at the default of 5060 unless a non-standard port is required by your provider.
Add your first SIP server by clicking the ADD button in the servers section of the page.
The SIP Server field is where you enter the IP address or URL of the SIP server. In this example, the SIP server is a local FreePBX server with an IP address of 192.168.1.11.
If desired, you can add a more descriptive name for each SIP server using the Name field. In this example, the name we’ve entered is FreePBX. (If left blank, the name field will automatically populate with whatever value you entered into the SIP Server field.)
The External IP field is available if port forwarding is needed at a router or gateway. If port forwarding is required at the NAT router of the SIP network, this is where the public facing IP address of the router should be entered.
If your server is part of a Local Domain, enter that here.
As you can see from the example, it’s possible that one or more fields will be left empty.
After populating all the necessary fields, press SAVE.
Press the ADD button to add your first SIP extension.
In our example, we’re registering extensions to a FreePBX server. Both Extension and Auth User are the extension number configured in FreePBX. We’ve toggled the Register switch to ON and left the Expires field blank. For Auth Password, we copied the “Secret” field from each extension’s configuration page in FreePBX.
As you can see, not all fields are populated in our example configuration. This is OK.
We recommend configuring SIP extensions as endpoints that use registration. This makes network troubleshooting easier and tests the entire IP connection to the SIP server. Be sure to toggle the Register switch to ON. This will make VXs register with the PBX, essentially logging in, whenever a show referencing that extension is made active.
The Expires field lets you change the interval (in seconds) that VXs will refresh registration for a particular extension. As a general rule, don’t populate the Expires field unless Telos support recommends it to solve a specific problem.
As SIP messages list IP addresses and ports used to transmit audio on (via RTP) it doesn’t work well if the client is in a private LAN but needs to communicate with a SIP provider outside of the LAN. As messages pass through the router, it translates addresses in IP headers, but not the SIP message itself, giving the provider wrong connection info.
Many SIP providers use clever hacks to work around this limitation without any additional support from the client. If you are connecting VXs to a SIP provider that doesn’t provide such a service, don’t worry - VXs has basic NAT support built-in. Contact Telos support for details.
Before going any further in your configuration, you will need to add a license to your system. Copy and paste the license code supplied by your dealer into the License field on the Licenses page and press the ADD button. You should then see the license key you activated, as well as what features it enables in the Licenses section of the page.
If you are unable to activate your license, or if the license count is not correct, please contact Telos support.
The Studios page lists all the studios that are configured for your VXs. It lets you add new ones, and lists the show that each studio is using.
To add a new show, press the ADD button.
The Studio Name field will populate automatically with a suggested name for this studio. You can leave it at the default or change the name to whatever meets your needs.
You’ll then need to set a number of lines that are available to this show. Typically 6 for a VSet 6 and 12 for a VSet 12, but it can be any number, even beyond the capacity of the call controller you will be using with this studio.
For example, if you have a VSet 6, you could configure a show with more than 6 lines, as long as lines 7 and above were configured as fixed lines with auto answer fixed lines enabled. This could be useful if you wish to have an extension dedicated to IFB or intercom, or configured as a listen line. (Note: You would not have control over this line to force a disconnect, and would depend on the calling party to terminate the call.)
Keep in mind that lines are a licensed feature, and the number of lines assigned to each studio counts against the number of line licenses you’ve purchased.
Next, you need to add either Fixed or Selectable channels.
Keep in mind that fixed Channels have a 1-to-1 relationship with a particular line in a show, while selectable channels are allowed to choose any line that isn’t tied to a fixed channel. Selectable channels function the same way previous generations of Telos talkshow systems always have.
To add a fixed or selectable channel, press the ADD button in either the Fixed Channels or Selectable Channels area.
The Display Name field shows how this channel will be advertised alphanumerically on your Livewire network. (This is the same info you would enter into the Source Name field on an xNode.) A default value will populate when you add the channel, but you can change it to whatever you like.
The Output field is where you define which Livewire channel number or AES67 multicast address you want this channel to have on your AoIP network. The adjoining drop down menu allows you to select between 1 ms AES67 Multicast streams or 5 ms Standard Stereo streams. This is the caller audio.
The Manual Backfeed / Input field defines the input audio path from the studio. This is the audio that will be sent to the caller. Depending on your selections, this field will populate in various ways.
If you set the Output field to a Livewire channel number and select Standard Stereo, it will default to an auto backfeed, useful for interfacing with Axia consoles. The Manual Backfeed toggle switch will remain off, and the field will display “auto.”
If you set the Output field to a Livewire channel number and select AES67 Multicast, it will create an auto backfeed for you, which will also work with an Axia console, however, the Manual Backfeed switch will automatically toggle to on, and the field will be populated the information for a Livewire backfeed automatically. If you are using VXs with a phone type source on an Axia console, there is no need to make any changes.
If you set the Output field to an AES67 multicast address, the Manual Backfeed switch will toggle to on, but the Input field will be blank, and will require configuration.
Anytime you click inside the Input field, this box will appear, allowing you to configure everything as needed for your application.
If you are utilizing xNodes to interface to another brand of console, click the dropdown box that currently says To Source and select From Source instead. Enter the Livewire channel number you wish to receive in the Address field and press SET. In this example, we are taking Livewire channel 5302 as the send to caller audio for this selectable channel
The same principle applies if you are configuring VXs for use in an AES67 environment, only you won’t see the To Source / From Source options. Simply put the appropriate AES67 multicast address in the Address field and make the appropriate selections indicating the number of channels in this AES67 stream and which channel to use.
Program On Hold is the audio a caller will hear when placed on hold. Configuration for this works the same way as Manual Backfeed / Input configuration. In most situations, the program on hold audio is fed from the console’s main program buss. This audio should normally be pre-delay to allow callers to interact with hosts in a natural way when their call is taken to air.
Note: You must configure a Program on Hold feed for each studio. It could be a generic program audio or air feed, but if no working channel is assigned, calls on hold may drop after 30-60 seconds. This is because some PBXs, trunks, and some endpoints will hear silence and disconnect, thinking the call was lost.
AEC is an optional, licensed feature which helps when you monitor calls via a loudspeaker in the same room as the microphone feeding the phone. Without a canceller, the caller’s own voice would be sent back to them as an annoying echo.
Prior to configuration, you will need to toggle the Enabled switch to on.
The canceller needs two inputs and produces one output. The Mic Input is fed from the studio microphone. This can also be the entire mix-minus signal to include multiple microphones.
The Reference (CRMON) input is the audio that needs to be canceled. This is the audio that is going to the monitor or preview loudspeaker that caller audio is played out of.
The Output (Backfeed) of the canceler goes to the VXs phone feed input(s.)
As with fixed and selectable channel configuration, these can be either Livewire channels or AES67 multicast addresses.
VXs supports GPIO (General Purpose Input/Output.) It is a useful way to control VXs functions, or get indications of VXs conditions or modes.
Electrical connections are made via Livewire GPIO xNodes. You can also use Livewire GPIO to signal several different commercially available Livewire enabled studio notification systems, which display GPIO indications on a video display, or create your own using Pathfinder Core Pro.
For GPIO Actions, press the ADD button, select the specific action you wish to perform from the dropdown menu, specify a Livewire Channel number and Pin. Since GPIO Actions are inputs, leave Type at the default setting - From GPIO.
The available actions are:
Take next call
Take next ringing line
Hold all calls
Drop all calls
Enable Block All
Disable Block All
Toggle Block All
Toggle Auto Answer & Hold
Mute Ringer
For GPIO Indications, press the ADD button, select the specific indication (labeled Action here as well) then assign a Livewire Channel number and Pin. Since GPIO Indications are outputs, select To GPIO as the type.
The available indications are:
Next call available
Line ringing
Line ringing (Busy All)
Line ringing (non-Busy All)
Call can be held
Call can be dropped
Block All enabled
Auto Answer & Hold enabled
Ringer muted
Delay Dump
VSet phones have access to an address book stored in VXs. There are actually two sets of address books - one tied to each studio, and one tied to each show.
To edit the address book for a particular studio, go to STUDIOS -> ADDRESS BOOK then press the EDIT button associated with the Studio Name you wish to edit. Once inside, type a name to be displayed in the address book in the Name field, then the number or SIP address you wish to store in the Number / SIP Address field. Finish by pressing the SAVE button.
The procedure for editing the address book for a particular show is very similar to the one used for studios. Go to SHOWS -> ADDRESS BOOK and perform the same actions you would in editing a studio address book.
If a number is added to both the studio and show address books, that number will appear in the address book twice whenever that particular combination of studio and show are selected.
In VXs, a show is a collection of lines that are available to a studio. A studio can log out of one show and into another, bringing up a different set of lines on the VSets and other controllers. In the main shows page, you will see a list of shows that are configured and the studios they are currently assigned to.
To add a new show, press the ADD button.
On the show configuration page, we’ve set up a show with 12 lines, 8 of which are labeled Listener Line, 3 of which are labeled Warmline, and one is labeled Hotline.
The Extension and Server fields reference the SIP server and the extensions configured earlier. You can assign the same extension to multiple buttons as we’ve done here. Calls will ring in on the first available line, acting like a hunt group.
The Channel field allows you to designate whether each line will be Selectable (a number of lines switched to a single or small set of faders) or if it will be tied to a particular Fixed channel (a one-to-one pairing of lines and faders.) The order of the fixed channels in the dropdown menu references the order in which they’re listed in the Studio profile.
The Block All toggle specifies whether a particular line is busied when the Block All button is pressed on a VSet.
Finally, the Ringer dropdown can be used if you have uploaded custom ringer tones to the system. This feature is documented later in the manual.
It’s important to note that shows cannot be edited or deleted while assigned to a studio. You will see a notification like this one if you attempt to edit a show that is in use by a studio.
If you see such a notice, go into that studio’s configuration page and deselect it using the Change Show dropdown menu.
VXs has dynamics processing on both the send (from studio to caller) and receive (from caller to studio) audio paths. While we explain how these controls work, the vast majority of users find the default settings work very well and do not need to change them.
The send (to caller) audio processing is fixed and consists of a protection limiter and some EQ. The purpose of the send limiter is to protect the caller from clipping distortion.
The receive (from caller) processing is adjustable and includes ducking level, an AGC, noise gate, and dynamic EQ.
Receive AGC - This helps level calls out, to make each caller sound similar to the last. We recommend leaving this control at 16, which provides the most consistent levels.
Noise Gate - This removes background noise by gating (turning off) the output when audio drops below a certain threshold. We recommend leaving this setting off because it could mistake a soft-spoken caller for noise and gate unexpectedly. If most of the callers to your VXs are professional announcers calling from noisy environments, you may find this feature helpful.
Receive EQ Mode and Additional EQ - This control changes how the EQ algorithm functions. Setting the control to off bypasses all equalization of the caller’s voice. Setting the mode to Fixed will allow VXs to only apply the EQ specified in the Additional High and Low EQ fields. In Adaptive mode, the VX will dynamically adjust EQ levels for callers in an attempt to make all callers sound similar. The Additional High and Low EQ fields will be added to this automatic EQ.
Caller Ducking Level - This control reduces caller level when audio is being sent to the caller. This helps with intelligibility of host voices when the caller and host are speaking simultaneously, and during contentious interactions, the higher the setting the more the host can “shout down” a caller without anyone having to ride gain on the caller’s fader.
SIP signaling is via digital messages, not tones in the audio like it used to be. Like most of today’s VoIP devices, VXs has sounds loaded into it to mimic the sounds everyone was used to with the PSTN.
You can customize these sounds if desired, either disabling them, or by uploading new sound files to replace them.
If you wish to customize any of these tones, the files you upload must be in the following format:
AU file extension
Linear PCM
8, 16, 24 or 32 bit
Ringtones MUST be 8 kHz, 16 bit, mono with 48 kHz sampling.
The easiest way to get audio files into this format is with the open source audio editor Audacity. Open your file, then export it using these settings:
On the File menu, choose Export, then Export Audio…
Use file type “Other uncompressed files”
Under format options, select AU (Sun/NeXT) with Signed 16-bit PCM encoding.
If you are making ringtones for your VSet phones, convert your audio to 8 kHz before exporting by:
On the Tracks menu, choose Resample…
In the “New sample rate (Hz)” box, select 8000.
The various types of tones loaded into VXs are described below.
Call progress tones:
Dial tone - The sound heard when you pick up the handset before a call is dialed.
Ringback tone - Heard when dialing is complete and the called phone is ringing.
Busy tone - Heard when the called phone is busy.
Reorder tone - A fast busy tone. Signals that there are no call paths available.
Error tone - Usually caused by an incorrectly entered number, but can be from other problems during call setup.
Call disposition tones:
Call answered - A clicking sound played whenever a call is put on air.
Caller hang up - A brief, low tone that is heard whenever a caller on-air disconnects.
Line switch - Mimics the sound of a line button being pressed on an old 1A2 key phone.
Caller alert tone - Sent only to the caller when a call is picked up (not played over the air.) It is a brief ding played at the same point in a call as the white noise bursts played by earlier phone systems to help with line equalization.
Choose the DTMF sub-menu to set these tones. VXs is loaded with the same DTMF tones used traditionally with the PSTN, but they are scrambled on-air so listeners can’t easily decode the number being dialed. (The correct DTMF tones are played to the VSet handset and headset jacks.)
One thing to note about SIP and DTMF. The DTMF tones generated by VXs are merely played for the audience and operators to mimic traditional phone service. It is sometimes necessary to send DTMF to the PSTN after a call is connected, such as for dial-up remote control systems. In this case, VXs sends a SIP message to the PBX or gateway to generate the corresponding DTMF tones. The scrambled DTMF tones used by VXs for privacy do not impact this process at all.
Ringtones play through the VSet speaker when a line is ringing (if not muted via GPIO or VSet settings.) Should you wish to upload custom ringtones, please note the audio file requirements listed above.
These pages provide some insight to the inner workings of VXs for troubleshooting.
The studio information menu provides a list of studios, the number of lines in use by that studio, the number of fixed and selectable channels handled by that studio, and the number of devices (VSets and other controllers) logged into the studio. Pressing the SELECT button for a particular studio will provide more information about that studio.
This page shows the SIP registration status for each line currently in use by the studio.
In this example, we intentionally used an incorrect password for line 5 to show the difference between when a line/extension is registered properly, and when registration fails.
The devices page shows us what devices (VSets, Consoles, PCs and other controllers) are currently connected to VXs. The IP address and ports are listed, as well as the studio that owns the device.
This page displays statistics about the streams handled by the AoIP network interface(s.)
This page displays all of the hardware and health statistics for VXs.
Before using VXs, you will need to add a license code to enable the functionality you have purchased. To add a license, copy the code from the software license certificate provided by your dealer into the License field and press ADD. Afterward, you will see the license code under Licenses, below, as well as the activated features.
If a particular license code has an expiration date, a countdown to expiration will appear. If a license is perpetual, instead of a countdown, the word “Never” will appear in the Expires field.
As of this writing, the Node name and Activation Refresh/Release buttons serve no purpose and can be ignored. They will support future functionality.
The User Management page shows all of the users currently registered to the VXs with their associated TelosCare ID email address, or local account.
Though a TelosCare ID is required for the first login to a VXs after container deployment, you can create one or more local user accounts. To create a "user" account that functions like the user account on (for example) a Telos Alliance xNode, press the ADD button. On the configuration page that appears, leave the Username field set at its default of "user." Press ADD again to create the generic "user" account. You will now be able to log into VXs by clicking LOG IN at the main login page.
To change the password of a user, press EDIT.
To remove access for a user from this VXs, press DELETE.
Click BACKUP SETTINGS to download a backup configuration file. If a login dialog appears, please type user in the user name field, nothing in the password field, and press OK. After doing this, your backup will download as any other file would in your browser.
To restore your settings from a backup, click the RESTORE SETTINGS button, then navigate to and select the backup file on your computer.
When this dialog box appears, press OK to proceed. Any existing settings on your VXs will be erased and replaced with the backup file you loaded. The system will then reboot with the new settings in place.
This tab does exactly what you think it would - it logs you out of VXs. Be aware that you will be logged out instantly. No confirmation dialog box will appear. Should you wish to log back in, you will need to use your TelosCare credentials to do so.
For help with selecting a suitable switch, please see .
There are separate manuals covering VSet configuration. You can find them on our website here:
For a list of all ports and protocols used by VXs, please visit .
The Outbound Proxy field is for use when your SIP provider has specified an outbound proxy IP address for you to use for your service. This will not be required for all SIP server configurations. More info is available via .
Getting callers to air has always been a bit of a challenge.
Telos was founded in 1985, and our very first product used DSP to resolve call quality issues that Bell Labs deemed impossible. The Telos 10 hybrid did what no other had done up to that point - it enabled interactions between hosts and callers that were truly broadcast quality.
In the years that followed, Telos phone systems made their way into thousands of studios worldwide, connecting listeners to their favorite stations in ways that were never before possible. These digitally processed analog phone systems made caller audio sound natural.
As time and technology progressed, it became clear that calls delivered to stations digitally was the future, and Telos delivered the TWOx12 - an ISDN BRI (or POTS) based 12 line broadcast phone system. Outfitted with ISDN line cards, the connection between the telco central office and studios became digital, eliminating many of the call quality issues earlier hybrids had to resolve. Not long after, the Telos 2101 system brought this technology to scale, and made it possible for a single phone system to serve a facility with up to 4 ISDN PRIs and 32 studios.
Over time, Session Initiation Protocol (SIP) became the standard on which most commercial phone systems were based, ushering in the VoIP era in telecom. Expensive circuits gave way to solutions that are more cost effective and flexible, routable anywhere IP networks exist. With SIP, your phone system isn’t tied to a telco demarc. Calls can be routed via wired or wireless networks, even over the public internet. SIP allows us to create phone systems for broadcast with levels of backup and redundancy never before possible, and the same (or better) call quality than all technologies that preceded it.
SIP is now a mature technology. Developed in 1995, 25% of all calls were SIP by 2003. Telos saw the quickly emerging trend and unveiled the world’s first SIP based broadcast phone system, Telos VX, in 2012.
VXs is our 3rd generation SIP phone system, offering more flexibility than previous generations. While the original VX, VX Prime, VX Enterprise, and VX Prime+ required custom hardware based engines, VXs is container based and can be deployed:
On-premises, sharing the same server hardware as other Telos containerized solutions.
In the cloud.
Via a hardware appliance, much like VX Enterprise and VX Prime+.
With VXs, you have the flexibility to choose the solution that best suits your needs, and the support you desire. We’re here to help with:
Telos ProService Field Application Engineers who can work with you to take your project from concept to installed, working product.
Worldwide partners who thoroughly understand VXs, and the entire Telos Alliance product line.
Expert support through our TelosCare PLUS customer care program.
