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CHAPTER 2 |

37

INSTALLATION AND CONFIGURATION

SIP and DTMF

The VX uses standard SIP techniques for dialing. A SIP call setup message containing the number is sent to the IP network. The
gateway to the PSTN generates the DTMF (Dual Tone Multi-Frequency) audio signal that the PSTN uses to direct the call.
The VX generates DTMF audio just as sound effects really. Listeners expect phones to sound like phones. DTMF tones “played”
for the audience are not the actual digits dialed and some are not even “real” DTMF digits, but sound like them. This is so that
if a number is dialed “live, on the air” nobody can get the telephone number being dialed by recording the program audio or
“decoding” the tones heard on the air.

There is sometimes the need to send DTMF to the PSTN after a call is connected, such as for automated attendant systems or
“dial up” remote control systems. In this case the VX sends a special SIP message that tells the PSTN gateway to generate the
corresponding audio DTMF tones, according to the standard specified in the IETF (Internet Engineering task force) RFC2833.

Information Pages

These pages provide some insight to the inner-workings of the VX for troubleshooting and management purposes.

Studios Information Menu and Pages

The studio Information Menu provides a list of Studios, Fixed and
Selectable Channels handled by that Studio, devices (VSets and other
controllers) connected to the Studio and the Show that is logged into the
Studio. Clicking on a Studio Name will take you to the Information page
for the studio.

The Studio’s information page will repeat the details provided by the
Studio Information list and provides more information about the Fixed
and/or Selectable channels, POH, the IP address of the Devices connected
how many calls are currently active or on hold, and the status of the Line
positions. This is really helpful if you are having trouble with lines on the
VSet. You can also navigate to the Studio or Show configuration from here
by clicking the blue links at the top of the information page.

The status of the lines in Studio A (Show 1) say, “OK (Registered)”
meaning they are successfully registered to the SIP server and are ready to
make or take calls.

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Содержание VX Enterprise

Страница 1: ...The Whole Plant Broadcast Talkshow System USER S MANUAL Version 1 4 April 2019 TelosAlliance com...

Страница 2: ...an equipment rack an easily accessible safety disconnect device should be included in the rack design To reduce the risk of electrical shock do not expose this product to rain or moisture This unit is...

Страница 3: ...Telos Alliance which are mentioned in this manual or accompanying material are informational only The Telos Alliance makes no endorsement of any particular product for any purpose nor claims any resp...

Страница 4: ...authorization number which must be written on the exterior of your shipping container Please do not include cables or accessories un less specifically requested by the Technical Support Engineer Be su...

Страница 5: ...eening software applications 6 The Livewire AOIP Network 7 The Acoustic Echo Canceller 7 2 InstallationandConfiguration 8 Connecting to the PSTN PBXs and Providers 8 What s today s equivalent of a Lin...

Страница 6: ...tion 36 SIP and DTMF 37 Information Pages 37 Studios Information Menu and Pages 37 Calls Information Page 38 Devices Information Page 38 Stream Statistics Information Page 38 System Status Page 38 Lic...

Страница 7: ...9 GPIO PortWiring 50 Setting up a Mute Ringer GPI 50 Setting up a Ringing Indication GPO 51 Single Mix Minus Routing with xNode 52 Operation with Fusion Consoles 56 Setting it up 57 Ringer Mute 58 Ope...

Страница 8: ...nputs withTelos Alliance xNode 71 Analog Outputs withTelos Alliance xNode 71 Digital Audio Inputs and Outputs 71 Frequency Response 72 Dynamic Range 72 Total Harmonic Distortion Noise 72 IP Ethernet C...

Страница 9: ...thatinspireourcustomerstocreatethemostexcitingandengagingaudioexperiencesimaginable We re grateful that you have chosen audio tools from Telos Systems Omnia Audio Axia Audio Linear Acoustic 25 Seven S...

Страница 10: ...ions Engineer from Telos who s installed hundreds of VX systems Both will be well worth your while Thanks for considering and purchasing the Telos VX conceived by a great Broadcast Engineer and our fo...

Страница 11: ...phone numbers with you Since this is the next generation of VX Engines this is the second revision of the VX manual The first was created back in 2012 and was co written by Steve Church Founder of Tel...

Страница 12: ...S67 environment in addition to our own Axia Livewire network With plug and play connectivity you can network multiple channels of audio with any manufacturer s AES67 compliant hardware Beyond AES67 Li...

Страница 13: ...ion 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 Fraunho...

Страница 14: ...mode only Fusion and iQ Console Controllers The Fusion Controller Module and iQ 6 Fader Expansion provides the ideal way to integrate broadcast phones into the on air console the control center of ev...

Страница 15: ...ng software applications PC based applications take studio phone operations to a higher plane With many packages to choose from most provide the usual call screening functions for phone active broadca...

Страница 16: ...a Approved It is important to understand that not all switches are created equal or are interchangeable A qualified Livewire or AES67 capable switch is required to use Livewire or AES67 The switch mus...

Страница 17: ...s PSTN is an acronym for Public Switched Telephone Network the traditional telephone network is now made up of many types of service providers including the Legacy Utility Telco providers often called...

Страница 18: ...ecret We are pretending to be old POTS lines on an old key system for the user s benefit at least as far as the workflow and feature set goes We designed the system to look and feel familiar Talent ha...

Страница 19: ...his could involve separate LANS VLANs Quality of Service QOS settings IP routing and even firewalls Codecs we now have some control over what codecs are used The PSTN as it exists in 2018 uses the g 7...

Страница 20: ...er That number is recognized by Telco and is connected to a PRI trunk to your facility The telco connects to your PBX and sends the DID number to the PBX on the far end of the Trunk The PBX uses those...

Страница 21: ...e PBX on the far end of the Trunk Your provider s PBX uses those digits to route the call further to the correct extension in most cases The PBX rings the desired extension and causes ringback tone to...

Страница 22: ...should be returned charging the calling party of the call When the call has ended the stream is torn down and the customer billed If there was more than one appearance of the called extension on any p...

Страница 23: ...onvenient either in individual studios or in a central rack room Select the right mix and quantity of inputs and outputs that is most useful foryour plant When doing this keep in mind that the outputs...

Страница 24: ...a damage claim with the carrier who delivered it The VX Enterprise and VX Prime Plus are packed into a single rack space using modern server chassis technology There is a special method to racking the...

Страница 25: ...ill fit a rack between 26 and 33 deep The fixed inner portion of the glides with their locking tabs are already installed onto the chassis so the assembly of the outer portion is all that s needed The...

Страница 26: ...k 5 Align the square pegs on the rear end of the glide to the holes on the rack and push the glide rail into the rack until the quick release bracket snaps into place securing the rail to the rack or...

Страница 27: ...Getting theVX on the Network s The Engine front panel display gives at a glance status information The main status screen is pictured here It shows the IP number for the LAN Livewire port its connect...

Страница 28: ...J 45 Ethernet jacks on the rear panel Both are 100 1000 auto speed and duplex setting The WAN port is to be connected to the business network or the network containing the SIP server The LAN port is f...

Страница 29: ...propagate the Livewire traffic to ports that are not subscribed to a particular audio channel Thus blocking the high volume traffic from places where it isn t wanted An example of a Livewire AES67 sw...

Страница 30: ...nd will not let you change it 1 Enter the IP address that you are assigning to the phone and press the green Go button 2 Do the same for the netmask value Most often this may be kept at the default 25...

Страница 31: ...on your key phone s handset This term has faded away in the wake of modern phone systems but we ve held on to this term along with dial hyrbid and line to keep the workflow easy to comprehend by user...

Страница 32: ...rnet or sometimes over a dedicated connection Its IP address goes here and should be within the WAN port s subnet WARNING Do not assign the gateway address to the Livewire network as Livewire is not r...

Страница 33: ...Config are the settings for your location A green check mark will appear after you enter and accept your NTP server setting and the server connection is successful If there is an Asterisk server pres...

Страница 34: ...er name is user Default password is ___________ blank there is no password Passwords are case sensitive If you change any of these you might want to write the new values into this manual here or elsew...

Страница 35: ...service This will not be required for all SIP server configurations The External IP field is available just in case port forwarding is needed at a router or Gateway If port forwarding is required at t...

Страница 36: ...commend 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 tick the Register checkb...

Страница 37: ...works fine Be aware that there are several other G 722 codecs that are proprietary for use with certain brands of SIP business telephone sets or mobile devices These are not currently supported by th...

Страница 38: ...d new ones And it lists the Shows that studios are using A Studio is a collection of controllers GPIO and audio I O while a Show is a collection of lines to be used in the studio Individual Studio Con...

Страница 39: ...box From source should be selected at the dropdown box and enter the LW channel number of the desired xNode source This is the audio that goes back to the caller it should be a Mix Minus or clean feed...

Страница 40: ...d days this acoustic coupling would probably cause feedback howl instead of echo but today s mobile phone and VoIP delays have made echo the trouble du jour The canceler needs two inputs and produces...

Страница 41: ...TYPE to TO GPIO Available indications include LINE RINGING NEXT CALL AVAILABLE BLOCK ALL ACTIVE RINGER MUTED AUTO HOLD AND AUTO ANSWER ENABLED among others Since each GPIO port has 5 input and 5 outp...

Страница 42: ...Studio s configuration profile From there we can select NO SHOW for the CURRENT SHOW and click CHANGE Then we can go back to the Show profile and edit away The name of the Show can be edited as can t...

Страница 43: ...ources Full details are in the Using VX for couplers and Creating Dummy Studios for call screening and couplers App note toward the back of this manual Call Audio Processing Page The VX has dynamics p...

Страница 44: ...ones page SIP signaling is via digital message not by tones in the audio like it used to be This means the VX has to make its own sounds in response to various line status conditions for human consump...

Страница 45: ...eck the Enable checkbox to disable it Caller Alert tone Sent to caller when call is answered or when caller is taken to air from hold to inform the caller that they are now on air The Caller Alert sou...

Страница 46: ...to the standard specified in the IETF Internet Engineering task force RFC2833 Information Pages These pages provide some insight to the inner workings of the VX for troubleshooting and management pur...

Страница 47: ...e any troubleshooting in a Livewire environment will be familiar with these statistics These problems are usually caused by an unstable clock on the AoIP network or an improperly configured Ethernet s...

Страница 48: ...le to understand the log messages after some effort we don t expect you to do so These are mostly intended for our support people who would advise you as what settings to use for the capture and then...

Страница 49: ...omatically restart to apply the new configuration DO NOT attempt to load a backup configuration from another type of VX system For example you can not load an older VX or VX Prime configuration into a...

Страница 50: ...ugh the Engine Setup menu press and hold to access while the show setting can be done by any user from the Main Menu The studio option is not expected to be change often or casually but probably only...

Страница 51: ...olume calling lines will not provide caller ID information though most do Time The length of time the call has been ringing in or on hold Action Prompt None active in the graphic Shows what line will...

Страница 52: ...n the two states Not to worry we ve made it difficult to accidentally drop a call There is also a Two Fader Operation mode that will make the VSet operate more like the legacy Desktop Director and a 2...

Страница 53: ...n 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 Lock Normally taking a call to air causes any o...

Страница 54: ...inging rather than a busy tone Numeric Keypad and GO Button For dialing out either on the handset or on air The keypad works during an active call to generate tones for voice mail and other services W...

Страница 55: ...utton to take calls in the sequence determined by your producer and you press the Drop button when you are finished with your last call and you want to stop taking more One step up would have you pres...

Страница 56: ...I O for Analog AES EBU and GPIO First thing to do is to set up the xNode sources In the xNode s Simple Setup page there are three modes Stereo 4x4 Mono 8x8 and Custom Because caller channels are mono...

Страница 57: ...VX Studio s channel numbers into the appropriate DESTINATION output fields in the xNode config Once these changes are applied the VX Studio is ready to receive and send audio to callers via the xNode...

Страница 58: ...io has two sections for GPIO GPIO Actions are inputs to the system to remotely engage a feature or option while GPIO Indications are outputs to drive indicators as the name suggests We ve described th...

Страница 59: ...at the port s output pin normally high connect the Output Common Return pin 7 to the Ground pin pin 8 At the same time if the input pins are to be normally high device creates closures to ground then...

Страница 60: ...number 21509 When this is finished and the show reloaded into the studio a closure at this pin will mute the ringers on the VSets logged into that studio This is normally tied to the mic talley Setti...

Страница 61: ...manual available here https www telosalliance com Axia xNodes The instructions found above will apply to Legacy GPIO Node Element PowerStation and QOR GPIO Single Mix Minus Routing with xNode While a...

Страница 62: ...and SRC 8 to be our New Mix Minus channels Notice that we are making these all Live Stereo streams and that the IP address of this xNode is 192 168 2 215 so our channel numbers will begin with 215xx...

Страница 63: ...MIX 7 23 CLEAR MIX 17 1 CLEAR MIX 19 3 CLEAR MIX 21 5 CLEAR MIX 23 7 CLEAR Once the mixer is cleared we can begin setting up the I O and the mix minus routing First we need to route the physical input...

Страница 64: ...lect 0dB Make sure to click the ENABLE CUSTOM MODE link if you see it in the upper right hand corner of the matrix This is just an example The xNode s mixer can be setup using a virtually endless numb...

Страница 65: ...h activates flashing and displays PRESS LINE on the alpha display Selecting a line assigns it to the right fader To cancel the mode before selecting a line press the right drop button or the right cha...

Страница 66: ...enter the hybrid number in the HYBRID field and check MASHING ALLOWED to allow traditional conferencing Enter a 1 to allow the hybrid to follow the left hand column of line buttons on the module Enter...

Страница 67: ...controller similar to the one found on the Fusion 2 Fader Phone control module It works hand in hand with the VSet even though a VSet isn t needed for talent control with one of these The setup is sim...

Страница 68: ...on pin 1 so we will set PIN to PIN 1 and TYPE to FROM GPIO When the iQ s show and the VX s Show are both reloaded turning on the mic will mute the ringers on the VSets in addition to the speakers in t...

Страница 69: ...nd indications to this studio This allows us to do things like hook up indicators to GPIO xNodes to alert someone that ANY phone line in any studio is ringing among others See the GPIO section of this...

Страница 70: ...ow Now we need to create a new Show and assign these same extensions it Notice that we ve labelled both the Studio and the Show the same to keep confusion at bay In our example here we created a show...

Страница 71: ...igured in the dummy Studio The VSet will operate as a producer VSet only an can t control the lines on the air regardless of the Talent or Producer mode settings Creating an Asterisk Extension forVX B...

Страница 72: ...ndow will take you to the ADD SIP EXTENSION page USER EXTENSION Enter the extension number for the system In our example the extension is 5002 Then in the DISPLAY NAME field enter the name you would l...

Страница 73: ...hat ever you want but remember this because we will use it later We don t recommend using the default unless you can remember it DTMF MODE Set this to RFC2833 This is a standard DTMF protocol for SIP...

Страница 74: ...AME field and click the APPLY button directly beneath the General Settings fields Next find the ADD NEW section of the page Enter the extension number into the EXTENSION and AUTH USER fields check the...

Страница 75: ...ess the power button inside the front door 3 After Asterisk has booted the screen will show the Linux login Asterisk will process calls whether logged in or not To log in a User name is root b Passwor...

Страница 76: ...with another of the same model 4 Slide the new power supply module into the power supply bay until it clicks into the locked position WARNING The bulk capacitors in all switch mode power supplies can...

Страница 77: ...oIP 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 Sour...

Страница 78: ...D and further transmission will not be as loud clear or natural sounding as with VOIP SIP The companies below make great products but they are limited by POTS technology for performance and dependabil...

Страница 79: ...ne Sets Bell System Technical Journal 17 1938 358 380 Just for fun Waaay back Mr Inglis proposed high fidelity phones noting that cost was the only barrier to better audio quality in the telephone net...

Страница 80: ...vel Range Selectable 4 dBu or 10dBv Input Headroom 20 dB above nominal input Analog Outputs withTelos Alliance xNode Output Source Impedance 50 Ohms balanced Output Load Impedance 600 Ohms minimum Nom...

Страница 81: ...3 1 kHz 20 dBFS Digital Input to Analog Output 0 005 1 kHz 6 dBFS input 18 dBu output Crosstalk Isolation Stereo Separation and CMRR Analog Line channel to channel isolation 90 dB isolation minimum 20...

Страница 82: ...may take and supply audio directly to from the network Telos Alliance xNodes are available for pro analog and AES3 AES EBU breakout VX Enterprise supports AES67 connectivity Telco Connections Audio st...

Страница 83: ...rts and protocols used by theVX System 80 HTTP TCP LAN Port only 843 TCP Server for Flash socket policies it is required for the meters applet to make an LWCP connection to the VX 22 SSH TCP LAN port...

Страница 84: ...P Livewire Control protocol port 40000 40050 UDP Vset Audio Power Over Ethernet PoE The VSet uses standard 802 3af Power Over Ethernet POE and consumes 15 4 watts Please consider this relatively high...

Страница 85: ...C Telos Alliance Warranty Telos Alliance Limited Warranty For the latest Telos Alliance warranty visit telosalliance com warranty...

Страница 86: ...1241 Superior Ave Cleveland Ohio 44114 USA 1 216 241 7225 TelosAlliance com 2019 TLS Corp All Rights Reserved C19 17050 P N 1490 00210 001 USER MANUAL VX Enterprise VX Prime...

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