Page 1 Multi-Studio IP Phone Interface System USER’S MANUAL Version 2.0.1, May, 2014...
Page 2: We Support You
Feedback We welcome feedback on any aspect of the Telos VX, or this manual. In the past, many good ideas from users have made their way into software revisions or new products. Please contact us with your comments.
Page 3 Telos Systems USA Telos Systems 1241 Superior Avenue E Cleveland, OH 44114 USA +1-216-241-7225 (phone) +1-216-241-4103 (fax) +1-216-622-0247 (24/7 Technical Support) Support@Telos-Systems.com Inquiry@Telos-Systems.com Notices and Cautions This symbol, wherever it appears, alerts you to the presence of uninsulated, dangerous voltage inside the enclosure –...
Page 4: Table Of Contents
What’s the Big idea? ......2 The VX Engine ......3 The VSet Phone Controller .
Page 5 SIP Configuration ......16 Regarding The VX and SIP ......16 Server Configuration.
Page 6 Telos VX mAnuAl | Line Status Icons ......35 Line Button Columns......36 Hold Button.
Page 7 Controllers ....... 63 VX Engine ....... 63 IP/Ethernet Connections .
Page 8 Outbound Calls ......89 Connect your VX to Asterisk ......89...
Page 9 Power Over Ethernet (PoE) ......91 VX FAQ ....... . . 92 Application Example - WKSU .
Page 11: Welcome
Digital audio workstations, automation systems, and As you discover what the VX can do, I trust you will mobile phones are all beneficiaries of this remark- appreciate how it can enhance the appeal of your able progress.
Page 12: What's The Big Idea
And, with two channels, you could simultaneously talk and look up recipes. YouTube had apparently not entered their imagination. In the modern networked world, The Telos VX gets with the program. It is the next generation multi-studio phone system from Telos, providing a powerful, simple, and cost-effective way to share phone lines across a number of studios using standard IP technology.
Page 13: The Vx Engine
The VX Engine A 2U rack-mount device with enormous processing power, the VX Engine provides all the call control and audio processing needed for the system. It supports dozens of telephone lines and many studios. Its two gigabit Ethernet ports provides a cost-...
Page 14: The Vset Phone Controller
But because the VX system has a hybrid per line, it is often desirable to spread multiple calls over a number of faders, using one for each call so that operators can control each line’s level individually.
Page 15: Console Controllers
DJs prefer to record calls without using headphones or when guests need to hear calls without headphones. The AEC in the Telos VX is a remarkable new development. Its performance is shockingly impressive, permitting very high loudspeaker volume with no noticeable feedback or return echo.
Page 17: Installation And Configuration
Telco gateway, they are not needed when the studio equipment can accept AoIP natively. For an example of this sort of installation, see Axia Element Console as VX Controller in Section 5. Connecting to PSTN Lines: Gateways & PBXs.
Page 18: Using Voip To Connect To The Telco Network
| section 2 Grandstream GXW4004 4-port FXO gateway The “gateway” could also be a full-up IP PBX, such as from Cisco, Digium (Asterisk), and many others. In this case, compatible VoIP phones can be used for general office locations. SIP lets you move calls between the office and studio systems with no audio degradation.
Page 19: Livewire For Audio I/O
There is a lot of information in the pages that follow. Here’s the (very) abridged version of what you need to do to get your VX system up and running. If you will not be using an existing network to support the VX, install the Ethernet switch, or switches, depending upon your needs.
Page 20: Network Connections
Livewire audio streams to all ports, potentially overwhelming devices like PCs and printers - So you don’t want to plug the VX LAN port into an office network after Livewire outputs are enabled! ! Multicast-capable switches will not propagate the Livewire traffic to ports that are not subscribed to a particular audio channel.
Page 21: Vset Installation
Do not enter anything in the “Vset Gateway” field unless directed to do so by Telos Support. The Vset will operate on the Livewire network and will not need to be routed anywhere unless you have an unusual or custom installation.
Page 22: System Configuration
System Configuration Selectable and Fixed Lines The VX supports the concept of both ‘Selectable’ and ‘Fixed’ lines. It is possible to have a mix of both. Selectable lines are used for an operation style like traditional Telos systems, where there was a line selector before the hybrid or hybrids.
Page 23: Configuration Web Pages
InsTAllATIon AnD ConFIGuRATIon | Configuration Web Pages To access the configuration Web pages, find or connect a PC to the network that the VX Engine is connected to. Start a web browser and open a connection to the VX Engine by entering http://192.168.2.200 into the browser’s address field (assuming the default IP number has not...
Page 24: Main Page - General Configuration
LWCP Password. LWCP stands for Livewire Control Protocol, the way controllers communicate with the VX Engine. A password can be used to secure the Engine from being controlled by unauthorized devices. The VXset, for example, uses this password. By default, this is set to...
Page 25: Default Username, Password, And Ip Settings
NTP server here An NTP server must be accessible on either the LAN or WAN port. The VX automatically finds the correct port. If the NTP server is outside of both the LAN and WAN local network segments, gateway must be set. You can use a domain name if a DNS server is configured and accessible.
Page 26: Sip Configuration
| section 2 SIP Configuration The SIP configuration page shows the global SIP settings, along with a list of all SIP servers VX is configured to communicate with. The form at the bottom of the page lets you to add another server;...
Page 27 SIP Invite message and procedure. SIP-connected g.722 audio is via IP, so the VX Engine needs a public IP address that is accessible to the calling side. G.722 does not pass via the PSTN and gateways (which means that ISDN g.722 codecs will not work with the VX).
Page 28: Server Configuration
SIP the “line” is the address (and associated configuration). To change the line from trunk to station, mark the Register checkbox. This will make the VX register it with the server (meaning, log in) whenever a show referencing this line is active,...
Page 29: Sip And Network Address Translation (Nat)
Many SIP providers use clever hacks to work around this limitation without any additional sup- port from the client. If you are connecting VX to a SIP provider that doesn’t have such service, don’t worry - all is not lost, as VX has basic NAT support built-in.
Page 30: Individual Studio Configuration Pages(Part1)
| section 2 Individual Studio Configuration Pages (part 1) The individual Studio pages are to configure each studio. For each, you can choose how many faders you want to devote to selectable and fixed lines and you assign the Livewire channel numbers to each.
Page 31: Individual Studio Configuration Pages (Part 2)
With native Livewire consoles such as the Axia Element, this is what you need. But if you want to connect the VX to another console via an interface Node, the Node can generate only a “From source”, not a backfeed. For this case, choose Advanced receive and enter the LW...
Page 32: Gpio
AEC is what you usually would feed back to phones. For configuration purposes, you should think of the AEC as a separate block outside of other VX functions. There is no internal connection from/to the AEC and other VX signal paths.
Page 33: Overall Show Configuration
InsTAllATIon AnD ConFIGuRATIon | Overall Show Configuration Individual Show Configuration Pages...
Page 34: Vx In The News And Production Rooms
The flexibility of the VX extends to the newsroom and production studios. It’s easy to create a ‘studio’ in the VX that can be dedicated to a workstation or special need. Most newsrooms these days are simply PC workstations with a simple mixer off to the side. You can set up a hybrid for each of your workstations or production rooms.
Page 35: Call Audio Processing Page - Audio Processing And Metering
Canceller. The Call Audio Processing button allows you to enable or disable all of the audio processing features of VX, such as AGC, EQ, ducking, etc., all at once. Enabling this option will let you adjust the individual options below.
Page 36: Assigning Sounds & Tones - The Tones Page
48kHz Livewire rate and allows the switching and mixing to be performed synchronously within the VX Engine. The SRC adapts a number of rates on the telephone side. The usual telephone rate is 8kHz, but newer wideband codecs making their way into telephony will have higher rates.
Page 37 InsTAllATIon AnD ConFIGuRATIon | SIP signaling is via a text message, not audio. This means the VX has to make its own sounds in response to various line status conditions. We provide default sounds, which are similar to the traditional tones generated by the PSTN, but you can upload your own to create a unique on-air ‘signature’.
Page 38: Sip And Dtmf
Tone MultiFrequency) audio signal that the PSTN uses to direct the call. A gateway to ISDN lines translates the SIP message to the equivalent ISDN call setup message. The VX does not generate DTMF audio. Likewise, the gateway translates call progress signals from analog tones or ISDN messages to SIP.
Page 39: Logging
In the Logging config page you determine where logs are stored and how much detail is recorded. You can choose to store logs inside the VX or to an external PC on the network that is set up to handle Linux standard syslogs - If you choose the internal option, log messages will be kept in the VX’s RAM, so are limited in...
Page 40 DEBUG level for the module you want to trace. For example, if you want to trace what VX Producer or VSet is sending to the Engine, you might chose to select LWCPBE module log level to DEBUG.
Page 41: Firmware Updates
New firmware for the VX Engine is first obtained from Telos via email, Internet download, etc. It is then uploaded to the VX Engine using this page. You can have as many as five stored in the Engine at one time. On this page, you choose which of them will be active. Reboot the Engine after activating new firmware.
Page 42: Backing/Restore Configuration Settings
Engines appear in iProbe, just like any Axia device, and can be backed up and restored in the same manner. When restoring, either from backup or to factory defaults, it is possible to keep the current network settings intact. Note that VX will automatically restart to apply the new configuration. VSet Phone Configuration...
Page 43: Operation
operation VSet Operation Select Studio and Show If these have not been already selected, this should be done as the first step before using the VSet. Press the Menu button to access the menu functions. The LCD will show the various items that can be changed.
Page 44: Set Talent Or Producer Mode
The yellow/orange rectangle near the status icon. Line name The first text line. This is defined in configuration using the VX Engine’s control center web GUI. Caller ID Caller ID will be displayed on incoming calls if it is available. Outgoing calls will show the num- ber dialed.
Page 45: Time
opeRATIon | Time The length of time the call has been ringing-in or on hold. Action ‘hint’ (None active in the graphic.) Shows what line will be affected by drop and hold. Choose the line you want to drop or hold after pressing either of those buttons. This is shown only when more than one is possible;...
Page 46: Line Button Columns
| section 3 Hold Ready Hold (Screened Hold) Used Elsewhere (in another studio) Line is blocked Line Button Columns Each line has 2 associated buttons to the left of the LCD. Pressing a left column line button puts a held or ringing caller on the handset. Press the right column button to put a held or ringing line on air, dropping any other unlocked calls.
Page 47: Drop Button
opeRATIon | Drop button Drops a call that is active on the handset, or on-air. If more than one line is in this category, there will be a small drop icon near the status icon on the lines that could be held. Press the line that you want to drop.
Page 48: Numeric Keypad
In our older on-air phone systems, we were limited by the cost of the day’s technology to one or two hybrids, so we only used one or two console faders. With the VX system, this limitation is removed. That means that you can assign a fader to each call if you want to do so, allowing independent control of each call’s volume.
Page 49: Take It Easy
Entering the VX Engine IP address connects all the needed logic functions - line selection is the main one, but muting for the VX sets is also handled. Any VX set assigned to a studio and show shared with an Element will mute its ringer when the Element tells it to. Nice, eh? Selectable lines use the Element’s Call Control Module as the line selector.
Page 50 Fader modules would give you “Euro-style” operation, where each line has its own hybrid/ fader and there is no line selector. There is no limit to fixed lines vs. selectable in the VX, so you can have as many as you have available console faders.
Page 51: The Vx Producer Application
It also can install the Audacity audio editor. Remember that since the VX uses Livewire audio to carry the “handset audio” to your VX producer PC, that the PC needs to be on a LAN subnet that carries Livewire traffic (the same subnet as the VX engine Network LAN port).
Page 52 | section 4...
Page 53: Set-Up: Studio And Show
Press the handset button to the left of any line to take the call to the ‘handset’ that is connected to the PC running the VX Producer application. (Often this will actually be a headset/mic.) If no call is ringing-in, the system assumes you want to dial out and opens a window to let you do that.
Page 54: On-Air
Line name. This is the name that was given to the line during show configuration. Caller info. Text you enter into this area will appear on any other VX Producer applications and VSets that are assigned to the same studio. It automatically replaces the line name text.
Page 55: Selectable Lines
In older on-air phone systems, we were limited to one or two hybrids (audio paths), so we only used one or two console faders. With the VX system, this limitation is removed. That means that you can assign a fader to each call if you want to do so, allowing independent control of each call’s volume.
Page 56: Next Button And Function
| section 4 Next Button and Function The Talent next priority is: longest waiting ready hold longest waiting hold longest ringing-in The producer priority is: longest ringing-in The producer can manually override these and assign priority as desired. Block All Pressing this button will cause all inactive and ringing lines to be dropped and blocked from accepting any calls.
Page 57: Playing
The file will appear in the tabbed list box determined by Settings. Normally, a VX Producer in producer mode will put recorded files under the Producer tab and in studio mode, under the Studio tab (naturally). You can move files among the tabbed lists by either dragging them to the destination tab or using the Send button and drop-down list.
Page 58: Host/Producer Text Chatting
Host/Producer Text Chatting The window at the lower right lets people within a Studio communicate with each other. Type text into the lower field and it will appear on any other VX Producer applications selected to your studio. The ‘Studio’ and ‘Producer’ identifiers are added automatically. The colors for them can be configured in settings.
Page 59: Notes, Resources, Additional Information
In the past, when analog Telco lines were the norm and mobile phones were not yet attached to everyone’s ears , feedback was the usual problem, not echo per se. In earlier Telos hybrids, we addressed that with a combination of ducking, frequency shifting, and a basic canceller.
Page 60 AECs are provided within the VX Engine, but they are not connected into its internal signal paths. This is because they need to be placed in the signal paths between the studio loud- speakers and microphones.
Page 61: Telco Services And Interfaces
14.4kbps codec. Passing this through g.711 within the PSTN and then yet another codec on the way to your studio over an IP link is asking for aural trouble. The VX natively supports the g.711 A-Law, g.711 µ-Law, and g.722 codecs. Almost all PBXs and gateways support these formats.
Page 62: Ip Centrex And Hosted Pbx Services
| section 5 IP Centrex and Hosted PBX Services Just as it shouldn’t matter whether the PSTN gateway is on your premises or not, it also shouldn’t matter where your IP PBX is located. This is the principle that allows IP Centrex services or hosted PBX services to happen.
Page 63: E&M Trunks
noTes, ResouRCes, ADDITIonAl InFoRmATIon | Actually, there are two types of off-hook signaling. On a loop-start line, when a phone goes off-hook, the circuit is closed, and the central office detects the change in current. This is the common residential format. Ground-start signaling is a small modification to the scheme to permit disconnect supervision and remove the possibility of glare, where a PBX mistakenly takes a ringing-in line for an outgoing call.
Page 64: Isdn Bri
(like adding or removing prefixes, or changing between internal and external num- bers) is done by a gateway or PBX. For VX to do that would require it to know a lot more about the phone network, and believe us, you don’t want to configure that.
Page 65: Installation
VSets to PoE (powered) ports. The Engine WAN port could have its own switch to connect the VoIP lines and gateways. Or the VX can be configured so that the LAN port could serve everything with no additional switch. The Powerstation could also link up with a central switch to pass audio to/from other locations in the facility.
Page 66: Configuring The Element To Control The Vx
Element configuration is via web. Do the VX configuration first, and then move on to Element configuration. Things you will need to know about your VX in order to configure an Element as a controller are: The VX Engine IP address ♦...
Page 67 “Studio 1.”If you wish to change a VX Show using an Element Show Profile, simply specify the name of the VX Show you wish to load in the Show Name field. If you leave it empty, no VX Show will be changed, and whichever show is currently loaded will continue to be used.
Page 68 New Source Profile and, using the Primary Source drop-down box, select the audio channel that you configured in VX Engine under Studio configuration. In the list, you should see a Livewire channel number and short description of the Hybrid; give it an Element source name, such as “VX-Hybrid1,”...
Page 69 Normally this is used with a 4-Phone Fader Module. If you want to use a Hybrid in this mode, specify one of your VX system’s “fixed” hybrids in the Primary Source field. In the VX Engine studio configuration, you configure both fixed and selectable hybrids;...
Page 70: What's Next
IP is a potent enabler that has already showed us plenty of surprising things; it’s inevitable that more is on the way. The Telos VX represents an exciting change in direction for studio telephone operations. The open IP nature of the system, along with its rich user interfaces and a powerful platform, offers ongoing opportunities for enhancement of its capabilities.
Page 71: Beyond Edison's Legacy
Some mobile phone applications, such as Apple’s ‘facetime’ are beginning to take advantage of this. The VX is ready for high-fidelity phone calls, and includes support for g.722, a codec rapidly gaining favor with VoIP providers and users. g.722 uses about the same amount of bandwidth as g.711, but samples audio at 16kHz - double that of g.711! It delivers...
Page 72: Specifications
| section 5 Specifications System Maximum number of phone lines: 48, when used with aLaw or uLaw codecs for VoIP lines. (Higher quailty codecs, such as G.722, consume more system resources and result in a de- creased number of total lines available.) Maximum number of SIP numbers: 250 Maximum active on-air calls: 30 Number of hybrids: one per each active line...
Page 73: Controllers
Analog Line Input CMRR: >60 dB, 20 Hz to 20 kHz ♦ Controllers VSet12 telephone VSet6 telephone VSet1 telephone VX Producer Windows application Axia Element studio console Axia IQ studio console Neosoft Neowinners Broadcast Bionics PhoneBox VX VX Engine IP/Ethernet connections...
Page 74: Studio Audio Connections
| section 5 Gated Receive AGC Receive filter Receive dynamic EQ Ducker Sample rate converter Line Echo Canceller (hybrid) Acoustic Echo Canceller (wideband) Power Supply AC Input Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse ♦...
Page 75: Packet "Sniffer
noTes, ResouRCes, ADDITIonAl InFoRmATIon | Packet “sniffer” www.wireshark.org/download.html (free) VoIP Soft phone” SIP PC clients X-Lite “Soft phone” SIP PC client (free) www.counterpath.com/x-lite.html Ekiga, free open source SoftPhone - ekiga.org/ Linphone open source VOIP phone - www.linphone.org/ Open Source PBX Distributions www.digium.com pbxinaflash.net/ www.trixbox.org...
Page 76: Telephony Discussion And Voip News Sites
| section 5 Telephony Discussion and VoIP news sites www.sipforum.org www.telos-systems.com www.voipforums.com/ www.voipuser.org/forum_index.html www.broadbandreports.com/forum/voip forum.voxilla.com/ SIP Information and suggested reading A Request for Comments (RFC) is a memorandum published by the Internet Engineering Task Force (IETF) describing methods, behaviors, research, or innovations applicable to the work- ing of the Internet and Internet-connected systems.
Page 77: Sip Providers
Internet connectivity in your area. Visit the Telos Website for the latest information. The short list below was created in December 2010 and the marketplace is changing rapidly. These are companies we or our clients have had direct experience with and report positively on their experiences.
Page 78: Local Number Portability In The Usa And Canada
The following is a peek inside of SIP. There is no reason you need to know any of this to use the VX. It’s an excerpt from Steve and Skip’s book Audio over IP. SIP is fast rising to be the big-daddy buzz-acronym among telecom technology acolytes. SIP is how calls are set up over IP connections, so it is actually pretty important.
Page 79: The Parts Of A Sip System
noTes, ResouRCes, ADDITIonAl InFoRmATIon | The SIP message protocol is similar to the Web’s HTTP (Hypertext Transfer Protocol), and shares some of its design principles: It is human readable and request-response structured. SIP even shares many HTTP status codes, including the familiar “404 not found.” Here is a typical SIP message: INVITE sip:skip@there.com SIP/2.0 Via: SIP/2.0/UDP 4.3.2.1:5060...
Page 80 Gateway as well, making a one-box solution that includes everything needed for a small office installation. An example a SIP Server being used in the broadcast world comes from the Telos Z/IP codec family, which uses a Telos-developed, enhanced SIP Server, called (naturally) Z/IP Server. The server is provided as a service on the Internet, but may also be installed by users who prefer to maintain their own.
Page 81: Addressing
SIP Request (INVITE, REGISTER, etc.). URLs (Universal Resource Locator) describes the location of a resource available on the Internet. For example, http://www.telos-systems.com is the URL for a Web home page. It is resolved by DNS to a concrete IP address.
Page 82: How Sip Works
| section 5 (E.164 NUmber Mapping) is the Internet service used to look up the URI associated with a particular E.164 telephone number. It’s part of the DNS system. SIP can use ENUM to locate the VoIP system associated with a telephone number that accepts incoming calls. How SIP Works As we’ve seen, SIP is a simple, text-based protocol.
Page 83 Within a PBX system, designers usually choose one codec as a standard for the system and stick with it for all connections. For example, the Telos VX studio system uses 8kHz sampling- rate, 16-bit uncompressed PCM internally for all calls that connect to the PSTN.
Page 84: The State Of Sip, And Its Future
Asterisk can also be used as a gateway. For example, you may need to peel a few channels off of a PRI to send them over to the VX, while the others feed a PBX. An ISDN PRI card in an...
Page 85: Asterisk Software: Distributions
SIP signaling are interfaced over a single Ethernet link. Telos has people on-staff and access to consultants who are experienced with Asterisk and who can help you with regard to integrated Asterisk/VX installations. Feel free to call us with your questions and suggestions.
Page 86: Hardware
We often put an extra Ethernet card in the machine or buy a motherboard with two Ethernets on the board. One is configured one for the “inside” (LAN)” and one for the “public” (WAN), similar to the VX. 2-4G of RAM is fine, but more is better. Hard disks should be server-grade and kept well-cooled for longest life.
Page 87: Expansion Cards
noTes, ResouRCes, ADDITIonAl InFoRmATIon | Keep any filters clear and inspect the machine periodically to see if ‘dust bunnies’ are col- lecting anywhere. Do a full backup using the Linux ‘DD’ (full disk image bit copy, or ‘clone’) command once in a while (quarterly/annually), or after you are satisfied that any major change you’ve made should be retained (after a trial period).
Page 88: Get Your 'Distro
| section 5 1 Get your ‘distro’ The most popular are at: www.digium.com ♦ pbxinaflash.net ♦ www.trixbox.org ♦ www.freepbx.org ♦ Download your distribution. We’ll proceed assuming that you’re using PBX in a Flash, hence- forth ‘PIAF’. It’s a few hundred megabytes. Once you have it saved on your hard drive, you’ll need to burn it to a CD.
Page 89 noTes, ResouRCes, ADDITIonAl InFoRmATIon | Select the US keyboard (or the keyboard of your choice) when prompted, then press the <TAB> key to advance to the “OK” prompt. After this selection, the GUI starts up. Move your mouse to verify that it works. The system asks how you want to use the disk.
Page 90: Log In To The Console For The First Time
| section 5 then Asterisk and the PBX in a Flash files. Enjoy a sip of your favorite beverage as the process continues over the next few minutes, Eventually, you’ll see a message congratulating you and that “Installation is complete”. Not so fast...
Page 91 noTes, ResouRCes, ADDITIonAl InFoRmATIon | Type ‘help-pbx’ and <enter>, you’ll get a list of commands. First, Let’s set up your ethernet IP address and get this box on your network so that you can get it going fully. At the ‘$’ prompt, type ‘netconfig’ and <enter>. You’ll get a blue screen (not like in windows, don’t worry!) asking if you’d like to set up networking.
Page 92: Confirgure Asterisk
VX because they are more flexible. For example, station SIP signaling conveys caller ID, which the VX can display and use. SIP trunks, on the other hand, do not pass CID. The VX supports the required SIP registration when you enter the authentication password on the show configuration pages.
Page 93 Create an extension that we’ll later use with the VX. ♦ Set up a trunk to accept incoming calls and to be used to make outgoing calls. ♦ Set up an ‘inbound route’ to tell incoming calls how to get to our extension on the VX. ♦...
Page 94 Set up an outbound route so that you can make calls out ♦ Then we’ll use the VX web interface to use the line that we created and then make a test call. Finally, we’ll create a trunk from the PSTN and route calls from that trunk to our extension, all by using the web GUI.
Page 95 noTes, ResouRCes, ADDITIonAl InFoRmATIon | We see Recording Options and Voice Mail settings (You can record all calls or the extension user can dial a code while on the line to initiate recording of that call. Recording is optional, of course.
Page 96: Trunks
| section 5 We enable the Vmx Locater. This is useful as it allows callers to be transferred to number inside or even outside of the system if 0, 1 or 2 is pressed. Here ‘0’ is programmed to send callers to the station operator, ‘1’...
Page 97 noTes, ResouRCes, ADDITIonAl InFoRmATIon | In our example here, we only use Vitelity for incoming calls, so the ‘Outgoing Dial Rules’ are irrelevant. We could insert a period in the Dial Rules box that would simply pass whatever digits were dialed down the trunk to Vitelity. Keep in mind that this is just the trunk setup, and that there still has to be inbound and outbound routes to get calls into and out of the switch.
Page 98 FreePBX GUI. Remember that if the show that uses extension 2222 is not selected on the VX (through the VSet or VX Producer software) calls to 2222 or the DID number routed to it will receive an “All...
Page 99: Outbound Calls
We’re going to assume that you’ve done the basic setup of your VX. That means you’ve got audio going and a VSet phone connected and talking to your VX engine, and it’s on the same network as your Asterisk box.
Page 100: Make A Test Call On Your Vset
Extension 2222 has been verified as present. Now we’ll go to the VX Shows page and look at the show that we built earlier. It has a 6 line ‘hunt group’ for extension 2222 and a couple of “back lines” or hot lines. It’s a ‘hunt group’...
Page 101: Default Ip Addresses
ResouRCes, ADDITIonAl InFoRmATIon | Default IP Addresses: VX Engine ‘Network’ interface = 192.168.2.200 (front panel settable) VX Engine ‘WAN’ interface IP address: OFF by default, no IP set. SIP Parameters Codecs supported as of early 2011: g.711 alaw and ulaw. g.722.
Page 102: Vx Faq
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 1MB POTS lines? How? Can I use ground start lines for incoming calls only? You can do that, and it’s not difficult.
Page 103 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.
Page 104 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’...
Page 105 MPEG-ELD in Apple’s ‘facetime app. Note that VX’s call audio processing is bypassed when using g.722. 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.
Page 106 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...
Page 107: Application Example - Wksu
When a caller chooses the option “go on the air,” the call is passed to the VX, where it may either be directly answered on-air or be fielded by a producer. This IVR function...
Page 108 Calls are routed from Asterisk to the VX via SIP/VoIP over the LAN. The Asterisk is configured to provide a SIP extension for each line that connects to the VX. (SIP trunks could have been used, but extensions are more capable - for example, they can pass Caller ID.
Page 109 ResouRCes, ADDITIonAl InFoRmATIon | Incoming Call flow to VX via Asterisk Open Source PBX Asterisk Switch DID numbers from SIP Provider Asterisk SIP Trunk (311) 555-2368 Digits sent include Asterisk Extensions (311) 555-9467 Caller ID and 10 May be SIP or ZAPTEL...
Page 111 Product must be delivered, shipping prepaid, to: Telos Systems, 1241 Superior Ave., Cleveland, Ohio 44114. Telos Systems at its option will either repair or replace the Product and such action shall be the full extent of Telos Systems’ obligation under this Warranty. After the Product is repaired or replaced, Telos Systems will return it to the party that sent the Product and Telos Systems will pay for the cost of shipping.

Telos VX User Manual

Welcome

Installation and Configuration

Operation

The VX Producer Application

Notes, Resources, Additional Information

Quick Links

Summary of Contents for Telos VX

Table of Contents