VSet (v1.1.0-10148) VX Producer (v1.0.1) Engine (v1.1.1-10110)
Meet Telos VX: the future of broadcast talkshow systems.
VX is the world’s first VoIP (Voice over IP) talkshow system. It’s incredibly powerful, incredibly flexible, and highly scalable -- a powerful whole-plant broadcast phone system that’s also economical enough for stations with just two or three studios. VX connects to traditional POTS and ISDN telephone lines via standard Telco gateways. But it can also connect to VoIP-based PBX systems and modern SIP Trunking services to take advantage of low-cost Internet-delivered phone services. Using standard Ethernet as its data backbone, VX significantly eases the cost of phone system installation, maintenance and cabling, while making it easier than ever for talent to take control of their phone system. VX is truly the future of broadcast phones.
With VX, we’ve wedded the capability of modern networking to the remarkable power of today’s digital processing to bring the benefits of the resulting synergy to broadcast facilities. With VX, you can move and share lines between studios at the touch of a button. VX is naturally scalable, capable of serving even the largest of facilities -- while remaining surprisingly cost-effective for even single stations with more modest needs. To make the most of this networked environment, we’ve built VX around the VoIP standard.
VX uses Ethernet as its network backbone, a powerful yet simple way to share phone lines among studios and connect system components. VX plugs right into Axia IP-Audio networks, connecting multiple channels of audio and control via a single Ethernet RJ-45. If you don't have an IP-Audio network yet, VX works with Axia nodes to provide analog or AES audio and GPIO connections.
Why VoIP for broadcast?
VoIP has taken the business world by storm, increasing the flexibility of office phone systems and PBXs while simultaneously lowering maintenance and equipment costs. In fact, most Fortune 500 companies have replaced their older PBX systems with VoIP for just these reasons.
VoIP is a natural for broadcasters, interconnecting the phone system CPU with audio interfaces, phone sets, console controllers, and PCs running screening software by way of efficient, low-cost Ethernet. Using VoIP, you can finally share phone lines among multiple studios and route caller audio anywhere in your facility, easily and instantly. Got a hot talkshow that suddenly needs more lines in a certain studio? Just a few keystrokes at a computer and you’re ready -- no delays, and no cables to pull. VX can even connect with your business office’s VoIP PBX to allow easy call transfers.
But it’s not just VoIP — It’s VoIP from Telos. Every incoming line has its own fifth-generation Telos Adaptive Digital Hybrid, our most advanced ever — packed full of technology engineered to extract the cleanest, clearest caller audio from just about any phone line, even notoriously noisy cellular calls. Multiple lines can be conferenced with superior clarity and fidelity. Smart AGC ensures consistent caller audio levels. New Acoustic Echo Cancellation from FhG removes feedback and echo in open-speaker studio situations. And should you choose to use SIP Trunking telco services, calls from mobile handsets with SIP clients will benefit from VX’s native support of the G.722 codec, instantly improving caller speech quality.
The VX Engine is the heart of the system. A fan-free 2RU rack-mount device with enormous processing power, the Vx Engine provides all the call control and audio processing needed for the system. It supports up to 30 active calls on-air simultaneously, across as many as 20 studios. VX is Web-based, so remote control and configuration are a snap – you can work with it from any place you can get online.
The VX Engine’s DSP platform is so powerful, it provides a hybrid for every line, allowing multiple calls to be conferenced and aired simultaneously with excellent quality. Incredibly advanced DSP hybrids make caller audio sound its best, no matter what kind of line or phone the caller uses. Smart AGC coupled with Telos three-band adaptive Digital Dynamic EQ and a three-band adaptive spectral processor are part of the toolkit; send audio gets a frequency shifter, AGC/limiter and FhG’s Advanced Echo Cancellation technology to eliminate open-mic feedback. Call ducking and host override round out the package.
With VX, choice comes standard. Want to use traditional phone services, like T1/E1, ISDN, and POTS? The Vx Engine works with standard telco gateways from Patton, Cisco, Grandstream and others. Want to use a VoIP-based PBX or SIP Trunking telco service? Vx Engine uses standard SIP (Session Initiation Protocol) and RTP (Real-time Transport Protocol) to direct up to 30 simultaneous on-air phone calls.
The VSet12 phone controller is an IP-based phoneset with two large, high-contrast color LCD panels that provide line status and caller information using easy-to-understand Status Symbol displays. Use them like a traditional Telos controller –select, hold and drop calls as normal. Or, use them to map phone lines to individual faders for greater control – even assign a group of lines to a single fader. Additional controls lock calls on-air, start external recording devices, and queue calls for sequential airing. There’s a built-in address book and call history log, and the display screens deliver detailed line status, caller information, caller ID, time ringing-in or on-hold, and comments entered in the VX Producer screening software application – perfect for error-free production of fast-paced shows.
The VSet6 phone controller is a six-line version of the VSet12. Like its big brother, it has a large, friendly color screen with animated Status Symbol icons, and controls for 6 phone lines. With all the control functions of the VSet12, it's great for secondary studios or other locations where only six lines of control are desired.
The VSet1 phone controller provides convenient single-line access to your VX system in news booths, voiceover stations, etc., where control of multiple phone lines is not necessary. Its bright display screen and intuitive controls let operators easily hold, drop and step through queued calls.
Specs At A Glance
- The first VoIP telephone system designed and built specifically for broadcasting.
- Works with POTS, T1/E1, ISDN and SIP Trunking telco services for maximum flexibility and cost-savings.
- Standards-based SIP/IP interface integrates with most VoIP-based PBX systems to allow transfers, line sharing and common telco services for business and studio phones.
- Standard Ethernet backbone provides a common transport path for both studio audio and telecom needs, resulting in cost savings and a simplified studio infrastructure. Connection of up to 100 control devices (software or hardware) is possible.
- Modular, scalable system can be easily expanded to manage a network of up to 20 studios, each with a dedicated Program-On-Hold input – truly a “whole-plant” solution for on-air phones.
- System capacity of up to 48 standard phone lines; supports up to 250 SIP numbers.
- Up to 16 hybrids, with as many as 48 active calls (up to 4 per hybrid), may be placed on-air concurrently.
- Each call receives a dedicated hybrid for unmatched clarity and superior conferencing.
- Native Livewire integration: one connection integrates caller audio, program-on-hold, mix-minus and logic directly into Axia AoIP consoles and networks.
- Connect VX to any radio console or other broadcast equipment using available Analog, AES/EBU and GPIO interfaces. Audio interfaces feature 48 kHz sampling rate and studio-grade 24-bit A/D converters with 256x oversampling.
- Powerful dynamic line management enables instant reallocation of call-in lines to studios requiring increased capacity.
- VSet phone controllers with full-color LCD displays and Telos Status Symbols present producers and talent with a rich graphical information display. Each VSet features its own address book and call log.
- Drop-in modules can integrate VX phone control directly into your mixing consoles.
- Included VX Producer screening software with built-in soft-phone allows a “phone” connection on any networked PC. Integrated recorder/editor simplifies recording of off-air conversations.
- Clear, clean caller audio from fifth-generation Telos Adaptive Hybrid technology, including Digital Dynamic EQ, AGC, adjustable caller ducking, and send- and receive-audio dynamics processing by Omnia. Wideband acoustic echo cancellation from Fraunhofer IIS completely eliminates open-speaker feedback.
- Support for G.722 codec enables high-fidelity phone calls from SIP clients.
Console Controllers: Live calls or pre-recorded, interviews or audience participation, one thing’s certain: phone segments are an integral part of today’s fast-paced radio. Wouldn’t it be great if talent could take control of phones without ever having to take their focus from the board? They can: IP-Audio networking technology provides the ideal way to integrate broadcast phones into the on-air console -- the control center of every studio. VX connects directly to Axia Element, iQ and Radius mixing consoles using Livewire IP-Audio to eliminate the cost and complexity of old-style inputs, outputs, and mix-minuses. IP-Audio consoles with built-in phone controllers don’t need any additional wiring: control signaling, caller audio and backfeeds ride on the network connection that’s already there. Easy!
Integration helps shows run smoother, too, since phone controls are right in front of them (instead of on some outboard phone control panel).Talent enjoys phone controls right on the board to dial, answer, screen, and drop calls without ever diverting attention from the console. And since the console now communicates directly with the phone hybrid, mundane tasks such as mix-minus generation, recording device activation and playback of pre-recorded conversations can all be automated, allowing talent to focus on doing what they do best — their show.
VX Audio & Logic Interfaces let you connect VX to any non-networked radio console or other broadcast equipment, using standard analog or AES/EBU interfaces. A GPIO Logic interface provides control logic where needed.
The Telos VX is a “facility wide” on-air telephone system. That means multiple studios, multiple stations, multiple shows and with minimal hardware requirements. With VX, there’s no need for the maze of discrete cables once required by a multi-line talkshow system. All VX components are linked with standard Ethernet, so a single CAT-5 cable provides connection to the telco interface, line switching commands, data communication between the Vx Engine and VSet phones, transport of caller audio to mixing consoles, return of mix-minus and program-on-hold audio to the caller, data messages (such as call notes and IM) between producer and talent, Livewire audio call recording, and transfer of recorded call files from the producer to the studio. Telco is delivered via IP through a POTS, ISDN or T1 gateway device, a SIP PBX, or a dedicated IP circuit using SIP Trunking. Got an Axia Livewire AoIP studio network? Telos VX will plug right in. Audio inputs and outputs are Livewire real-time audio channels and travel over your existing Axia system just like the rest of your audio. Axia console GPIO ports can be used for “phone ringing” tallies or remote control of profanity delay units. It’s the seamless integration of studio phones, mixing consoles and routing network you’ve dreamed about! This diagram shows just how easy it is to combine VX with your Axia network.
Don’t have IP-Audio networking yet? Not to worry… VX will work with all console brands, networked or not, via VX Audio and Logic interfaces – compact 1RU breakouts that put multiple I/O channels right where you need them. This diagram shows a typical studio with an analog mixer, using VX Analog and GPIO logic interfaces to connect to the console and other broadcast equipment.
VX comes complete with XScreen from Broadcast Bionics. When they asked if they could use these products as a platform for their new XScreen product, it took us about a millisecond to say "yes!" Partly because we believe in open standards and the benefits of partnerships, but also because we think XScreen is very cool. XScreen’s interface gives screeners and hosts tons of information and control using sophisticated visual talkback, including a drag and drop database of all calls for your show as well as a phonebook and visual warnings for persistent or nuisance callers. A fully-functional copy of XScreen Lite is provided to all VX customers, but an upgrade to the full XScreen client software adds even more features, including extended call history, an enhanced phonebook, prize management, powerful GPIO functionality plus more. XScreen, deployed as part of a Livewire network, also enables call recording, editing and console integration directly over the network.
The Telos VX uses VoIP. What does that mean to me?
Let’s define what “uses VoIP” means. The VX uses it in two distinct ways: One: it can connect to Telco services using standard SIP VoIP. You benefit from having options –connecting to PBXs digitally, to ISDN and analog lines via gateways, etc. With VX you can finally integrate your on-air phones with office phone systems from a variety of vendors. Getting Telco service from VoIP dial tone providers means that your audio quality and hybrid null will be much better as VoIP dial tone is delivered “4 wire” without hum, noise, and loop loss. Building on ubiquitous VoIP standards means a variety of third party hardware can offer flexibility. And you might save a lot of money getting service this way. Two: VX system components connect to each other over standard IP/Ethernet networks with all the advantages that brings. For example, in Livewire-equipped facilities, one RJ-45 jack connects dozens of audio channels and rich control to phones-like controllers, PC applications, integrated console controllers, etc.
So I can use the VX with my regular POTS lines? How? Can I use ground start lines for incoming calls only?
You can do that, and it’s not difficult. You only need a POTS gateway device. However, we encourage using digital delivery for the best sound quality.
What about ISDN BRI and PRI lines?
If you have ISDN now and want to keep it, again there are gateways available. However, it is often cheaper to port these numbers to a VoIP dial tone provider. All of these options are worth considering.
But you must have old-fashioned analog and ISDN circuit switched connections covered somehow, eh?
Yes, VX works with these by means of “gateways” made by many third-party providers such as Grandstream, Patton, and others. See the “Gateway Products and Suppliers” section of the VX User Manual for some suggestions.
How do I know if my existing PBX is VoIP capable?
Since VoIP depends on SIP, and SIP is a defined standard, the only question is if your PBX can provide a trunk between itself and your VX system. If so, your PBX will work with VX – but if you have any doubts, call Telos Support and we’ll help you figure it out.
My PBX is not VoIP capable. What else do I need to order, and from whom?
If your existing PBX cannot provide a SIP trunk line, it may be too old to offer such service, or might need upgrading. Contact your PBX provider and ask if your PBX can be upgraded for use with VoIP systems.
If you do not need to connect VX to your office PBX, however, you might opt for pure SIP Trunking service. We have experience with several SIP Trunking providers, such as EndStream Communications and Bandwidth.com, but you can obtain service from any VoIP provider that will give you a “my own device” plan. Be aware, though, that a simple Vonage account will not work – they limit the number of calls you can take based on the number of trunks you have (usually, in this case, just one). This type service is a home phone replacement — not a true SIP provider.
So I can experiment with SIP VoIP trunks? Why would I want to do that?
Actually, we recommend it! We think that you’ll find that they work better than you may have expected, as many of the VoIP problems we have seen are caused by limitations of analog terminal adaptors (IP to POTs gateway devices)! Since these are not needed with a VoIP-based system such as the VX, that class of problem is eliminated. There are a number of inexpensive ways to try VoIP without risk. You can also get VoIP-delivered numbers from distant area codes and exchanges. If you’re paying mileage for foreign exchange lines or have national toll free numbers, you’ll definitely want to consider this option.
Let’s say I decide to use SIP trunking. What sort of line provisioning do I need to order from my VoIP provider?
Unlike ISDN, SIP trunks are actually pretty simple. Your VoIP provider will give you the IP address and registration information you will need. Once you have this, enter it into your Vx Engine using the SIP Configuration web page in the VX Control Center.
Will a SIP trunk ordered from, say, AT&T work directly with VX as long as it has QoS?
Sure! AT&T (or any SIP Trunking provider) will provide you with the authentication info necessary for the connection between VX and your VoIP provider.
I can’t put a flasher across a VoIP line, so how can I flash a light when the hotline rings?
This same issue arises with ISDN, so beginning with our TWOx12 we included a GPIO output for this function. Not to worry – The VX has this capability. In fact, it has multiple outputs which can be assigned to any of your VoIP lines.
I have been watching VoIP with interest. But reports I have heard about services such as Vonage are that sometimes they work well, but other times not. Frankly, I am surprised to see Telos advocating it.
We get this concern often and understand why you ask. The term Voice over Internet Protocol (VoIP) does not distinguish between “VoIP over the Internet” versus “Voice over other (managed) IP networks”. If you have been keeping up with the transition to IP Codecs, you probably have noticed the same terminology issue there. IP-based STLs over an IP T1 are just as reliable as traditional STLs over traditional TDM T1 circuits. You are completely right to be concerned about a VoIP trunk (or an STL) over the Internet, as that is not at all the same thing, and performance in that case could be variable. Inside the facility, on a LAN, all problems disappear, since you have plenty of bandwidth and full control over the network.
So shared bandwidth is the only problem with VoIP?
Well, there’s audio quality. In the early days VoIP used a lot of compression, with bit rates being as low as 6kbps. Needless to say, the resulting audio was not impressive. These low-rate codecs have mostly fallen by the wayside. The lowest-grade codec the VX supports is g.711, the standard for digital audio in the telephone network pre-IP. And you will eventually benefit from higher-fidelity codecs as these proliferate in the VoIP world.
How does AoIP relate to VoIP?
Despite the similar names and underlying technologies, they are very different with regard to performance and application. An analog phone line and a balanced 600Ω studio audio circuit are pretty much the same tech, but the applications and performance are very different. “AoIP” has come to mean professional studio-grade audio networking - full-fidelity and usually with no compression. Low-delay and synchronized channels are other distinguishing characteristics.
Another way the two differ is that AoIP uses an advertising/discovery protocol for receivers to find sources instead of the Session Initiation Protocol (SIP) that VoIP employs. AoIP uses a system-wide clock mechanism to support low-delay and tightly synchronized channels.
Finally, AoIP often takes advantage of IP’s multicast capability to permit multiple receivers to listen to an audio source efficiently. AoIP is intended for managed, guaranteed-bandwidth networks, such as with an Ethernet switch as the core of a local area network.
And what about “IP codecs?”
Now you are very close to VoIP! These use SIP for call setup and various codecs for compression, so are similar to VoIP telephones. In fact, they actually are VoIP telephones and can sometimes interoperate with them. They have better codecs than VoIP phones, though. AACELD, in particular. Advanced IP codecs, such as our Z/IP, employ sophisticated technologies to overcome the Internet’s deficiencies. Dynamic buffering, error concealment and more clever stuff.
I have read that some VoIP PBXs use IAX trunking. Can I use the VX with these?
IAX is a protocol invented by the Asterisk people. It provides functions similar to SIP, but with more bandwidth efficiency. The VX doesn’t support IAX trunking at this time. But you can connect the VX to an Asterisk with SIP trunking or as multiple SIP extensions. There’s plenty of bandwidth on a LAN, so this works fine, while staying with a standards-based approach. We like Asterisk as a VX adjunct. It can add voice mail, automated attendant, blocking callers from caller ID, off-premise SIP extensions, and more, to a VX installation. Asterisk is free Linux-based PBX software that runs on a PC. The VX and Asterisk PBX are an attractive combo we expect will become popular within the broadcast industry.
I would like to use an Asterisk PBX with my VX system. Does Telos offer a pre-configured Asterisk installation?
A pre-configured Asterisk box just isn’t practical due to the variety of configuration options needed to tailor such a system to your station’s specific equipment and Telco service. We’ll be happy to provide resources that can help you get up and running, though.
I see that Asterisk can host POTS and ISDN interface cards. Should I use those to interface my old-tech lines, or should I use a stand-alone gateway?
This is a matter of taste, but a general guideline is that analog lines are best interfaced with an external gateway to optimize audio quality, while ISDN BRI and PRI can use Asterisk cards to have a lower-cost solution.
How can I get reliable VoIP trunk lines? What is involved? Can you recommend a vendor?
Yes, we can assist. We have been working closely with the VX beta sites and other early adopters of VoIP, so we have plenty of experience to share. There are several types of VoIP dialtone providers. You’ll want to consider how the service will be delivered to you; via the Internet (like Vonage), or via a dedicated IP circuit from the provider that includes a Service Level Agreement and guaranteed Quality of Service (as offered by a number of vendors including most of the traditional Telcos). For discussion of this and other matters, check the Telos web site on a regular basis as we continue post material on this and related topics.
I know that SIP is supported by the new IP codecs. Will the VX be able to connect to my Zephyr/IP in the field? Other codecs?
As we hinted above, Yes! The VX supports g.722 (7khz, ‘wideband audio’ and g.711 (3.4khz, “phone quality”).
What about SIP, SDP, RTP, ENUM and UDP?
We know that engineers are lifetime learners and encourage that. However, just as you probably don’t know much about “SS7” or “IUP” in the telephone network, understanding these details is optional. We do have White Papers on our web site to educate you on these, and other, terms. Start with the one below. You could also read Steve and Skip’s AoIP book, too, for a fun and comprehensive coverage of this stuff.
Here’s a paper on the adaptive IP codec: Advanced Tech for IP Remotes by Steve Church
Does Livewire technology come in to the VX picture?
Yup. The VX takes VoIP on the Telco side and Livewire AoIP on the studio side. This makes integration with Axia consoles and networks easy and efficient. If you don’t already have a Livewire network, you would use Axia analog or AES audio ‘nodes’ to provide I/O in either format. Each node provides eight stereo inputs and eight stereo outputs to and from the system. Each Axia GPIO node provides 8 “groups” of 5 inputs and 5 outputs, covering the needs of 8 studios. Telos Support is always available to help you specify exactly what you need. If you are new to Livewire Technology you may wish to skim through our Primer at http://www.AxiaAudio.com/manuals/files/IntroToLivewire2.1.pdf
So the two can live together side-by-side on the same LAN?
Yes they can.
So let’s talk caller audio quality. What does VX offer compared to the Nx series and your legacy products?
Advanced audio processing, and the fact that you never have to overcome Telco loop losses or extra two- to four-wire conversions means that the voice quality is as good as it can be. Calls from mobile phone calls will be less than perfect at times, but VX extracts the best possible from them. Caller audio is maintained digital and four-wire end-to-end, and VX does a clean sample-rate conversion from the Telco rate to Livewire’s 48khz. There is also a sophisticated dynamics processing section and automatic EQ designed with help from our colleagues at Omnia.
How are callers on the VoIP trunks going to sound? What about echo? Won’t cell phones sound even worse than usual?
The VX has enough processing horsepower to deal with even extreme echo situations, and four-wire Telco delivery means that the only external echo path is from the caller’s line, when it’s analog, and the caller’s handset. The VX uses Telos’ latest hybrid technology (5th generation), enhanced with the latest state of the art acoustic echo cancellation. Even when using open speakers, and changing levels during a call, the new algorithm makes feedback nearly impossible.
I heard that amazing Acoustic Echo Canceller on your Axia intercom system. No feedback at all with the speakers blasting and the mic a few inches away. Why don’t you put that in the VX so that DJs can answer calls on the cue speaker without feedback?
Done! The VX has it.
Steve Church once told me that IP cell phones can sound better than usual 3 kHz circuit-switched phone technology - something about G.722 dot something. Is this true?
Right. Current Cisco VoIP phones, for example, support the g.722 codec. The VX supports this, as well. However, Steve was probably referring to “AMR Wide Band” also known as G.722.2., sometimes called “HD Audio”. It’s 7khz and doesn’t sound at all like “phone audio” - in fact, it sounds better than regular G.722! AMR-WB is part of the new ITU standard for mobiles, so should grow over time. Meanwhile, some IP-based apps for mobiles are starting to use wideband codecs, such as MPEG-ELD in Apple’s ‘facetime app.
OK, I am starting to see the light. Cool stuff, but where’s the catch? Is VX hard to install and configure?
Setup is via web. It may be little different than what you’re used to (or not) but it’s not difficult, and some customers never crack open the book to set it up. Power and flexibility do come with a little complexity, but we’ll always be at your side should you need us.
There is no such thing as a free lunch - it must be hard to use then. I know there’s a catch... I really don’t have time to explain a new system to the air staff.
We know! Rest assured it’s easier for your talent, not harder! We recognize that any time you change anything in a studio, there can be some transition time. While there are a lot of new features in the VX, your staff can use the basic stuff immediately because it works just like familiar and comfortable Telos gear. The color, hi-rez LCDs and seamless console integration (to Axia Element and iQ) enhance the user experience. As you read this, systems around the world are screening calls and putting them on the air without drama. Jocks and Talk hosts alike praise the VX! Operators familiar with our longstanding two-column line selection will be right at home.
Is there any support for VIP lines?
Yes. We call these “fixed” lines, and you can have as many as you want. They are used for callers who stay on-air while other callers are coming and going.
What about conferencing?
This is one of the strengths of the VX. Since VoIP calls are four-wire, multiple lines can be conferenced with very high quality. The user interface lets operators assign selectable lines to multiple faders.
If I use an Axia console, it gets even better?
Yes – that’s the ultimate. You start with the most flexible console/audio-platform and then add smoothly integrated phones with the IP network powering it all. Sweet! The network delivers any of your Telco lines to any of your studios, in any combination. Any line is available in any studio at any time.
I notice the Vx Engine has both a LAN and WAN connector; why is that?
It’s a built-in firewall, isolating the VoIP connection from your studio network. We use this same approach in the iPort Livewire-WAN MPEG gateway.
What about call screener and database functions?
A basic Call Screening app, VX Producer, comes with the system. Other networked, PC-based apps, such as Broadcast Bionics’ PhoneBOX VX or NeoGroupe’s applications put information about your callers in front of your producers without the need for caller ID boxes, serial cables or other hassles.
What about SMS messages and chat - Can they be integrated into my phone system?
Using Broadcast Bionics Phone Box VX, yes! Telos has always built open systems to allow others to create their own visions around our gear.
OK, so the catch has to be the price?
The VX lets you leverage cheap networking to serve your entire facility. Since you don’t need hardware boxes for each studio, cost is surprisingly reasonable. You’ll use the VX in your on-air studios to replace older multi-line systems, and you’ll use it to replace hybrids in newsrooms and production studios. You might also decide to eliminate walls full of “couplers” for pre-delay IFB dial-up lines, and transitioning your Telco service might save you a lot of scratch. We seen stations saving thousands of dollars a month (no kidding) by eliminating POTS lines, with their taxes and fees.
Anything else cool about the VX?
Did we mention the color LCD user interface on the new VSet phone/control surface? Producers and talent love it!
Fancy gear like this has to be trouble, no?
VX is simpler than a multiple-box approach. With fewer components, it’s more reliable. At the time of this writing, no Vx Engine at a radio station has crashed, ever. We’ll probably have a contest to see who has the longest “uptime”. Right now, it’s 6 months - but that system was installed 6 months ago…
How do I enter an IP Address into the VSet phone?
Press the Menu button and then navigate to Setup. Select the IP Address field and enter the IP number you wish to assign. (If you need to correct an entry error, use the large X assigned to one of the 8 soft keys on the right side of the LCD display.)
The following Patton Gateways have been approved for use with Telos VX systems. The list below consists of orderable SKU numbers and Web links to product info pages for each unit.
For more information, please see Using Patton Gateways With Telos VX and the Patton Gateway Configurator (courtesy Broadcast Bionics)
- SN4940/1E24V/EUI (1-port T1/E1/PRI)
Connect your VX to digital PBXs or data networks for up to 30 simultaneous calls using SIP, T1, E1 or PRI signaling.
- SN4112/JO/EUI (2-port analog FXO)
Connects to your VX and provides 2 analog ports for use with POTS phones and devices.
- SN4634/3BIS/EUI (3-port ISDN BRI
Connect up to 3 ISDN BRI lines to your VX.
- SN4912/JO/RUI (12 FXS VoIP IAD)
Converts up to 12 incoming POTS lines to packet-based VoIP lines for use with your VX.
Telos VX systems use the Livewire standard for audio networking, and use the same Ethernet switches approved for Axia AoIP networks. Please visit AxiaAudio.com/switches for the list of approved switches.