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1. From the perspective of telephone network routing each channel appears to be a separate line with it s own number and independent dial out capabilities Since each has to be dialed or answered separately they appear to be lines to users also We refer to a B channel as a line on the TWOx12 s menus and LEDs Details Details Names We ve seen Basic Rate Interface ISDN called by a number of names The term S0 is sometimes used as a technical term for ISDN BRI Of course the term BRI is also used mostly by Telecom technicians The most common term we ve seen in Europe is ISDN 2 Multiple Subscriber Numbers MSN These are simply your phone numbers The number of digits will vary depending on your area Normally each ISDN BRI circuit will have three MSNs You will be using only two of them for each ISDN BRI circuit one for each B channel The LDN need not necessarily be entered in the unit although it would be if it is also one of your MSN s 124 Appendix 4 CSD and CSV Recall that each ISDN BRI has two possible B channels It is possible to order a line with one or both of the B channels enabled and each may be enabled for voice and or data use Phone terminology for the class of service is CSV for Circuit Switched Voice and CSD for Circuit Switched Data In contrast to PSD Packet Switched Data which is possible but irrelevant to our needs CSD is not required for Telos Talkshow
2. 42 Section 5 5 2 Status Displays Telco line and Desktop Director status information is available on both the Web interface and the LCD On the LCD pressing any menu button twice displays the status screens These status screens display information about the state of the Talkshow System but do not allow you to make any changes The A amp V buttons cycle through the following screens Slot A through B Line Status Desktop Director Connection Status 1 8 Line Status This shows the status of the Telco channels helping you to confirm that the Nx Talkshow system is successfully configured for each In addition information about call status is indicated One of the following words will be shown for each Telco channel present Deactivated No line connected or connected to wrong jack on the Talkshow System Or ETS 300 permits the Telco to put a line to sleep Deactivated will be displayed when this occurs Or system is currently testing for the presence of a POTS line e g looking for loop current SPID Pend SPID has been sent to network DMS Cust And Natl 1 1 only Network has not approved SPID SPID is incorrect SPID Error The SPID has been sent and rejected by the CO switch Ready The unit sees a live ISDN S interface i e external NT1 is present or internal NT1 has complet ed startup Or loop current has been detected on a POTS line and the line has been declared in service by the system Pending
3. Name Channel Source Mode 1 Calleri 1 Disabled 2 Caller2 2 Disabled Y 3 Caller3 3 Disabled Y 4 Caller4 4 Disabled Entering a channel number configures both Livewire Source and Destination for the hybrid Destinations will be configured To source To configure Livewire Sources and Destinations separately use Expert Mode Program on hold Name Channel Type 5 POH Showi 0 lt Backpanel XLR input gt E From source Y 6 POH Show2 0 lt Backpanel XLR input gt E From source Y Apply There are four hybrid channels You can enter the Livewire text name and channel number or accept the defaults Livewire channels are two way so this applies to both the send and receive audio A Livewire enabled mixing console such as the Axia Element when taking one of these as a source will automatically connect both directions and provide the mix minus Cool eh The two Program on Hold connections are Livewire destinations meaning that they are from the network to the Talkshow System Type should normally remain at the default From source setting IMPORTANT If you are not using a Livewire ready network do not enable Livewire streams or you will flood your Ethernet network If you lose your Web access to the Talkshow System after enabling the streams this is probably what happened You can switch the streams back off by disabling Livewire from the main unit LCD Audio menu Only when a properly configured Livewire
4. Trunk This is not the same as a Two way DID trunk See DID trunk Hunt Group and Trunk Common Channel Interoffice Signaling See CCIS Competitive Local Exchange Carrier See CLEC Concentration The basic premise is to share facilities wherever possible For instance while there may be thousands of custom ers served by a given Central Office there will be substantially less than that number of calls which can be handled simultaneously And even fewer long distance calls can be made simultaneously The art of Traffic Engineering is to have enough capacity that calls are rarely blocked but no more than that number See also Choke Exchange and Blocking CPC Calling Party Control Sometimes referred to as CPC Wink or disconnect supervision A call supervision feature on an analog loop start line that provides the ability for a CO Central Office to signal the called party when the calling party hangs up CPC allows the PBX key system or telephone answering device to reset the line so that it is ready to accept or initiate another call CPC is accomplished by either a loop current drop or reversal With some CO equipment it is also provided if the called party drops the call See also MCLD CPE Customer Premise Equipment Customer owned equipment located at his her facility such as a CSU or terminal In the USA and Canada the NT1 is part of the CPE CPN Called Party Number European Term The number that has bee
5. Ol Please put all Directory Numbers for all BRI s in a single hunt group C Please create a hunt group with all Directory Numbers except Specify USA ISDN ORDERING GUIDE 121 CL No hunt groups You may call the manufacturer of the Nx6 Nx12 Telos Systems at 1 216 241 7225 for any additional required information about ISDN compatibility Ask for ISDN Customer Support PROTOCOL National ISDN 1 From AT amp T 5ESS Northern Telecom DMS100 and Siemens EWSD If you use IOC Capability Packages please use Capability Package S unless otherwise specified CO Values Line Type National ISDN 1 Bearer Service CSD CSV on both channels CSV is required CSD may be omitted if requested by user TEI One dynamic per number Terminal Type Terminal Type A Voice Terminal Mode 10XXXXX Long Distance Access Yes Turn off features such as packet mode data multiple call appearances Electronic Key Telephone Sets EKTS shared directory numbers accept special type of number intercom groups network resource selector modem pools message waiting hunting interLata competition etc Note for AT amp T Lucent 5ESS on loop extended lines REX test must be disabled for this line using one of the following ASDFI266 Prohibit REX on BRI lines per office or ASDF1267 Prohibit REX on BRI lines per line Give us Two SPID numbers depending upon number of active B channels Two Directory Numbers PROTOCOL Northern Tele
6. Default is Off When using open speakers it is best not to change the speaker or mic positioning while the system is operating as this will require the acoustic canceller to re adapt and temporary feedback may occur Enable Adapt Burst Web Only Normally when a hybrid is started it outputs burst of white noise to the caller This helps improve sound quality by calibrating the echo canceller Default is ON It is highly recommended to leave it at factory default at all times DETAILED CONFIGURATION amp REFERENCE 49 Mix Minus Mode On the Web interface the Setup Wizard can be used to set the Mix Minus routing mode This option determines how many mix minus feeds you will need see section 2 7 for more on mix minus a must read for those not familiar with this essential configuration for telephone hybrid systems and the number of fader channels required on the mixing console The effect of the mix minus mode depends upon the Show mode which can be One Show 4 Column or one of the Two Show modes See section 5 7 for more about Show modes The mix minus mode options are Separate default This option provides fully independent inputs and outputs for each hybrid Thus you will feed the number of faders on your console equal to the number of hybrids your show has and your console will have to create the same number of mix minus feeds back to the Talkshow System In this mode callers cannot hear each other ex
7. Each ISDN circuit has one protocol and the protocol has no effect on where or whom one may call See ETS 300 National ISDN and Custom ISDN ISDN 2 A term used in Europe for ISDN BRI Also called SA Not to be confused with National ISDN 2 See BRI ISDN 30 A term used in Europe for ISDN PRI Also called S2M See PRI ISG Incoming Service Grouping A Hunt Group See Hunt Group IXC IntereXchange Carrier See IEC kbps KiloBits Per Second Measure of digital channel capacity Key Telephone System A system that allows multiple telephones to share multiple pre determined telephone lines The system provides indicators to allow the user s to understand the status of each line available on a given phone In its most basic form it is up to the user to provide the intelligence to select an unused line or answer a ringing line for example The Telos 10 Direct Inter face Module One x Six TWOx12 Talkshow System and Series 2101can be classified as Key Telephone Systems See also PBX LATA Local Access and Transport Area The area within which calls are routed by your Local Exchange Carrier LEC Under the divestiture of the Bell System calls going outside of this area must be handled by an Interexchange carrier IEC With the latest round of de regulation the usual IEC companies are being allowed to compete in the IntraLATA long distance market and LECs are beginning to be permitted to handle InterLATA calls LDN Listed Di
8. Firmware This code is analogous to a bootstrap loader and contains certain critical lowlevel software This information can only be updated or changed at the factory It is stored in a partition of the non volatile memory on the motherboard The software version and software build date can be determined from the start up screen The software version and software build date are shown The build date is not the date your Talkshow System was made 6 3 General Troubleshooting Thinking about Problem Solving Despite best intentions something could always go wrong Sometimes troubleshooting a balky set up can make even the toughest engineer a Maalox and Rogaine addict You cant fix any system without the right world view a zeitgeist of suspicion tempered by trust in the laws of physics curiosity dulled only by the determination to stay focused on a single problem and a zealot s regard for the scientific method Perhaps these are characteristics of all who successfully pursue the truth In a world where we are surrounded by complexity where we deal daily with equipment and systems only half understood it seems wise to follow under standing by an iterative loop of focus hypothesis and experiment 80 Section 6 The notions here apply whether you are solving problems at the system level or at the compo nent level At the system level the actions you might take would be very different checking cables trying different menu set
9. ISDN lines come in two varieties Basic Rate Interface called BRI S1 or ISDN 2 in various parts of the world and Primary Rate Interface PRI 2 or ISDN 30 BRI lines are the kind we normally see in broadcast stations as these are what are used with MPEG codecs such as the Telos Zephyr and Zephyr Xstream BRIs have a capability of one or two active 64 kbps channels Since Telos Talkshow Systems are used with BRI lines we will only consider that type here INSTALLATION 7 Alphabet Soup U S and NT1 An ISDN line from the central office is a single copper pair identical to a normal unloaded analog line When it arrives at the subscriber it is called the U interface It is a two wire con nection usually via an RJ 11 style modular jack The S interface is at the user side of the Network Termination Type 1 NT1 device The NT1 is sometimes generically called a NCTE Network Channel Terminating Equipment or in some countries an NTBA It is a four wire connection via an eight pin RJ 45 style modular jack Sometimes the S interface is called S T There is a subtle distinction between the two but it is not relevant for our purposes here and the two may be considered to be the same In the USA amp Canada NT1 functionality is usually included in the terminal equipment and indeed Talkshow Systems in these countries supports the U interface In other parts of the world the telephone company provides the NT1 Only one NT1 may be co
10. If you have difficulties with incoming calls try this 7 Click Apply then Reboot Now to activate the ISDN circuits Tip Each ISDN Line has a certain ISDN protocol or switch type which the phone company should have given you By far the most common protocols in the USA amp Canada are National ISDN 1 amp National ISDN 2 in which case you should set the Telco setting to Natl I 1 If you are in the USA or Canada and you do not know what protocol your line is configured for try Natl I 1 first If that does not work see Section 5 5 for hints on how to guess your protocol More Tips Incorrectly entered SPIDs are the most common problem users in the USA amp Canada have placing ISDN calls Compare the SPIDs you have entered with the SPIDs provided to you by the phone company to be certain that you have entered them correctly Do not add anything to your SPIDs Note that there are no hyphens dashes or spaces in SPIDs so the system will not allow you to enter any non numeric characters If your installer has included them in the SPID ignore them For a list of known working USA SPIDs by telephone company see Appendix A If you have PTP AT amp T Point to Point or ETS300 Euro ISDN ISDN protocol you should not enter anything in the SPID fields Hot Tip If you experience problems only with incoming calls check your DNs Is the ISDN working Successful initialization of a liSuccessful initialization of a line w
11. SPID ignore them 3 Ifyou have any questions about your SPIDs call your phone company 4 Ifyou have AT amp T Point to Point or ETS300 Euro ISDN Protocol mode you should not enter anything in the SPID fields MSN Multiple Subscriber Number This item if required must be configured from the Web interface Used only when you are using the ETS 300 Euro ISDN Telco mode and another ISDN device is sharing the line with the Talkshow System MSN1 corresponds to the first B channel This is simply one of your assigned telephone numbers Normally for ETS 300 these should be left blank DN Directory Number This item if required must be configured from the Web interface When you are using an ISDN protocol other than ETS 300 or AT amp T Cust PTP you must enter the Directory Number DN here This is simply the telephone number associated with the channel Tip Each directory number will correspond to one SPID You must enter them to reflect this relationship The DN corresponding to SPID1 must be entered as DN1 while the DN corre sponding to SPID2 must be entered as DN2 If you experience problems only with incoming calls check your DNs Normally 7 digits should be entered but you might need to enter 10 digits In some cases lines in a hunt group will require that the same primary Listed Directory Number be entered for all lines that are part of the hunt group Hot Tip Some versions of the Telos Zephyr and Zeph
12. There is no need to use an Extended Director for Producer mode Call Controller The call controller along with a POTS telephone emulates the functionality of the Desktop Director It is suitable for any case where a reduced footprint reduced complexity or reduced cost is required Console Director Telos provides a kit that can be used to make a drop in module for many types of mixing consoles Note The Talkshow System can also be controlled over Ethernet using Telos Assistant Producer or other software applications and by the Axia Element Console s Phone Fader or Call Control modules Nx6 Nx12 Desktop Director Ports RJ 45 PIN DESCRIPTION 1 N C N C Transmit to Nx6 Nx12 Receive from Nx6 Nx12 Receive from Nx6 Nx12 Transmit to Nx6 Nx12 Power 48VDC Power N A UU Bb WN 20 76 meters Max 37 meters 338 meters Ma Max Power Supply Section 2 Connecting Desktop Directors Direct Connection Up to four Desktop Directors can be plugged directly into the Talkshow System AEM 40 feet 12 meters max x4 Extended Distance with Power Supply If the Desktop Director must be located more than 250 feet 76 Meters from the Talkshow System you will need to use an external power supply Telos Desktop Director Power Supply Kit part 2091 00001 In that case your set up will look like this 250 feet 2050 feet 625 meters M Power m MEA Supply The Desktop
13. This is no problem if you have a set up that permits selective assignment to the hybrid mix minus PGM _ Mix Minus Bus Busses Send Bus Oye an n on on os oe gt ISDN or IP Codec Headphones Tel Line 1 Bus z A problem could result from a hybrid with too much leakage and the codec delay If the hybrid isnt doing a good job of preventing the send audio from leaking to its output the remote mix minus is corrupted Remember if any of the announcer audio from the remote site is returned it will be delayed by the digital link causing an echo effect in the announcer s headphones PGM out The Telos Talkshow System really shows its stuff in this situation Because it has such good trans hybrid loss leakage is not at all likely to be a problem And should there ever be a prob lem you can solve it increasing the amount of ducking See Sections 5 4 and Input To Caller The Talkshow System can be ordered with either analog or AES3 audio connections A label near the XLR connectors will tell you with which your unit is equipped In the case of analog there are two inputs In the AES case there are two physical inputs corresponding to four signal inputs Analog or AES o INPUT TO CALLER OUTPUT CALLER 1 2 1 2 Analog Inputs The analog inputs have the following characteristics Active balanced Line level 10dBv to 8dBu nominal level Maximum input level cli
14. but it may not work for all areas The CID Alert Signal setting can be used to change the signaling expected if you are not receiving CID information on your Nx12 Ring in Before Caller ID Web only Normally the ringing indication is passed to the Desktop Director and other devices after the Caller ID has been received This delays the ring indication until after the 2nd ring You may want to have a faster ring indication and choosing yes here lets you have it But you risk losing the CID if an operator answers the call quickly Some lines don t have CID and in this case this should be set to Yes since there is no reason to wait for the 2nd ring Hookflash Affects analog lines only It adjusts the duration of the hook flash loop current interruption when the Desktop Director transfer key is pressed The default value is 720 milliseconds The following table indicates typical values for various countries USA 720 ms Belgium Netherlands Switzerland 120 ms France Luxemburg Portugal Germany 270 ms Rest of Europe 90 ms DETAILED CONFIGURATION amp REFERENCE 57 The hook flash is sometimes used as a signal to initiate transfer or other features with the specific effect being dependent on the Telco or PBX capabilities and configuration CPC loop drop threshold ms Web only Sets the threshold time value for loop drop disconnect Loop Current Checking Web only Affects analog lines only The Talkshow Sy
15. diagnostic field If a network does not support this cause cause no 1 unallocated unassigned number shall be used Cause No 26 Non selected user clearing This cause indicates the user has not been awarded the incoming call Cause No 27 Destination out of order This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly The term not functioning correctly indicates that a signal message was unable to be delivered to the remote party e g a physical layer or data link layer failure at the remote party or user equipment off line Cause No 28 Incorrect number invalid number format address incomplete Special intercept announcement This cause indicates that the called party cannot be reached because the called party number is not in a valid format or is not complete or This cause indicates the user should be returned a Special Intercept Announcement Cause No 29 Facility rejected Special Intercept announcement undefined code This cause indicates a facility requested cannot be provided by the network or This cause indicates that a user in a special business group i e Centrex dialed an undefined code Cause No 30 Result of a STATus ENQuiry Special intercept announcement number unassigned This cause is included in the Status Message when the reason for sending the Status Message was previous receipt of a Status
16. the former Bell Operating Companies The Bell Operating Com panies are the traditional local phone companies pre 1984 except where one s service is from an Independent non bell telephone company or a CLEC Due to their former association with the Bell System RBOCs are regulated by the FCC differently than are independent Telcos or CLECs In many cases the Bell Operating Company structure is no longer used For instance here in Ohio we now deal directly with the RBOC Ameritech while the old Bell Operating Company Ohio Bell Telephone no longer exists Another trend is mergers among the RBOCs and in some cases the independents as well See CLEC and LEC RD Receive Data Data coming from the network or DCE towards the DTE Also a light on a modem or CSU DSU that lights to indicate presence of this signal Red Alarm An alarm state on a T carrier circuit that indicates that the incoming signal at the network interface has lost frame for more than a few seconds Normally a Yellow alarm is then returned i e sent back if a Red alarm is present A Red Alarm indicates a loss of inbound signal a Yellow alarm indicates indirectly a loss of outbound signal See also Yellow alarm Blue alarm and LOS Regional Bell Operating Company See RBOC Repeater A device intended to extend ISDN telephone service to sites further from the central office than could normally be served i e beyond 18 000 feet ISDN repeater tec
17. using one of the following ASDFI266 Prohibit REX on BRI lines per office or ASDFI267 Prohibit REX on BRI lines per line Give us One Directory Number 123 A4 Euro ISDN Ordering Guide This section applies to users in Europe and other countries where the Euro ISDN ETS 300 standard is supported much of South America Mexico etc Since there are virtually no options when ordering Euro ISDN this section is probably not needed however we thought we could help out by summarizing what you will require If you live in a country outside of North America which does not support Euro ISDN your lo cal Telos Systems representative should be able to provide you with information about ordering and using ISDN in your area We assume you have read the description of ISDN in section 2 3 and are familiar with ISDN s basic concepts If you have not read that section doing so now will help you better understand the information that follows On one ISDN BRI circuit there are three simultaneous bi directional channels two 64kbps bearer or B channels for the transmission of user information and one 16kbps data or D channel for call set up and status communication That is why this is sometimes called 2B D service You can use the B channels but only the phone companies and equipment manufactur ers can usually use the D channel ISDN comes into your facility on two wires just like plain old telephone service POTS
18. 4 ISDN Technology 101 2 ee ee 6 Alphabet Soup U S andNT1 0 0 0 0 0 0 0 eee eee eee 7 Data NOVO ih cig ee a edie ok RE Pek Wa dct e 7 MUNUGIOUDS Coso ade ad oe he eae eee ee ee 7 Special Considerations for Euro ISDN ETS 300 users 8 Ordering Helpi 244 42 0056 oo oa 44444 aa a ob deed 8 2 5 Connecting the Talkshow System to ISDN 8 ISDN in USA amp Canada Using the U Interface 8 ISDN Worldwide outside the USA amp Canada The S T Interface 9 2 6 Integration witha PBX o o o o ooo o 10 Using ISDN is 20 rice aa ee oR oh e 10 USINGIPOMS sete a a a ek Ra aa e a 11 2 7 Studio Audio Connections 2 ee 11 MIXEMINUS i e Sy ae eck are eee bo we eK ak ee eee ts 11 Using a modern broadcast console s mix minus capability 12 Making do with an older console o o 12 Using asmall mixer o ooo o 14 Using aproduction style console o o o o ooo o 14 Phones and Remotes o o o o o o o o o ooo o 14 lpi To Caller e e ae iaa a ete EY ELS 15 Output Gallet ewe kG awa ca ad aA 17 2 8 Desktop Director Ports o o o oa sa olen ooo o 18 Connecting Desktop Directors 2 ee 20 Desktop Director Assembly o o o o o o ooo ee 21 LEE ia a aa e 22 2 10 Parallel GPIO Remote Control o o o o o ooo 2 2 11 AC mains Power corae e e a aa 24 Basic C
19. A communications path between two switching systems Note that many trunks may be on a single circuit if that circuit has multiple channels The trunks most users will deal with are between the Telco switch and a PBX However a Tie Trunk can connect two PBXs See also Tie Trunk and Trunk Group Trunk Group A number of telephone channels which are func tionally related Most common is the Hunt Group Other com mon types include Incoming Trunk Groups and Outgoing Trunk Groups See also Combination Trunks and Two way DID Trunks Trunk Side 1 The side of a central office that faces the network Historically many CO switches could not make trunk to trunk connections as opposed to tandem and long distance switches that are always used to connect trunk together Hence the need to dis tinguish between the line side and the trunk side of the switch See also Line Side 2 The side of a PBX that connects to the Telco Historically many PBXs could not make trunk to trunk connections Hence the need to distinguish between the line side and the trunk side of the switch Since a trunk is a switch to switch circuit these circuits can be called trunks Beware even though you have a PBX the Telco may still call these lines even though your PBX considers them trunks See also Trunk and Line Side Two way DID trunk An ISDN PRI equipped for direct in ward dialing Most PBX trunks are related to a given phone num
20. An optional AES interface module is available Up to eight Desktop Directors may be connected Computer control such as with Telos As sistant Producer call screening application is supported over an Ethernet LAN connection Processing Functions General Telos 3rd generation adaptive digital hybrids Telos exclusive feedback reduction functions including Acoustic Echo Cancellation Send to caller Processing Sample rate conversion High pass filter Frequency shifter AGC limiter Pgm on Hold AGC imiter Sample rate conversion Receive from caller Processing High pass hum filter Smart AGC platform leveler Noise gate Telos DDEQ Digital Dynamic Equalization 3 band adaptive spectral control processor Sample rate conversion Audio Analog Inputs 2x send analog inputs 2x Program on Hold analog inputs 96 Section 7 XLR female pin 2 high Active balanced with RF protection Input level Adjustable 10dBV to 8dBu nominal level Analog clip point 24dBu Impedance Bridging gt 10K Analog to digital converter resolution 20 bits Analog Outputs 2x receive analog outputs XLR male pin 3 high Output level adjustable from 10dBV to 8dBu nominal level Output impedance lt 60 Ohms Digital to analog converter resolution 24 bits Headroom before clipping 20 dB from 4dBu nominal level AES3 Digital Inputs Outputs option The AES input output option module substitut
21. Desktop Director interface Nevertheless if you wish to make changes from the Desktop Director these controls allow you configure each line individually with regard DETAILED CONFIGURATION amp REFERENCE 65 to studio assignment and priority thus dividing the lines between the studios according to your specific requirements If custom line configuration is performed via the Desktop Director Show Mode will change to 2 Show Custom automatically To assign lines woe Ow UN oS 6 Press the MENU softkey on the Desktop Director Press lt V gt repeatedly until Configure Shows is highlighted and then press lt SELECT gt Enter the password and then press the OK softkey Note The default password is 1234 Press lt V gt to highlight Set Show Lines and then press lt SELECT gt Use lt V gt and lt A gt to highlight each line For each line Use the lt gt or lt gt Softkeys to select from the following options NONE Line is not available for either show SHOW 1 Line is available for the show that uses hybrid 1 only SHOW 2 Line is available for the show that uses hybrid 2 only SHOW 182 Line is available for both shows shared line Press the SELECT button to accept your changes To configure lines as Priority or Normal lines N oP YM SS 6 Press the MENU softkey Press lt V gt repeatedly until Configure Shows is highlighted and then press lt SELECT gt Enter the password and then press the OK
22. Enable Auto Call Drop Web only o 57 Ring Timeout web only o o o o o o o o oo o 57 5 6 The System Menu LCD Panel and Web page options 57 TOPAP Setup ec eh eG rr RE eee Sw a 57 Contrast LCD Only oscar Be eae OES 58 Show Mode LCDOnly ee ee 58 AudioLoop LCDOnly o ooo eoe 59 Factory Defaults LCD Only aaa a 59 Reboot LED ONIY s 6 444246 044 58444445084 a aa AA 59 Firmware Update Web Only a aaa 59 5 7 The Setup Wizard Menu Web Only 2 2 2 a 60 Show Configuration 2 oo e mm 60 Show Line Configuration o oo eaea ee 62 Audio 1 0 Configuration 2 oo ooo o 62 5 8 Phone Book Menu Web and Desktop Director options 63 5 9 Show Modes i aie oie a a ae es 63 Single studio Modes 1 a 63 Sharing the Talkshow System Two studio Modes 64 Selecting a Show on the Desktop Director and other controllers 66 5 10 Livewite s soe 244 ee eG EE a E ee ea 67 Livewire GPIO sre koe a a a ae AW we ee a 68 QUAlIYOFSEIVICO exe ee aA a ee 70 AKIA CONSOlES ioa e vaa Ad Bh ee A A ee 70 5 11 Firmware Update os o e eo o o e 71 5 12 Backup and Restore Web Only o o o aaa ee 71 TELOS NX TALKSHOW SYSTEMS MANUAL vii 5 13 Other Uses of the IP Ethernet Interface 72 Call control Software 2 ee ee 72 Jeria rada A 7 5 14 System Signal Flow and Audio Proce
23. Enquiry message or This value indicates that a user from outside a basic business group i e Centrex has violated an access restriction feature Cause No 31 Network disconnect Normal unspecified Special intercept announcement Call blocked because of group restricitons This cause is used to report a normal event only when no other cause in the normal class applies or This value indicates that a user from outside a basic business group i e Centrex violated an access restriction feature CLASS 1 2 resource unavailable Cause No 34 No circuit available circuit channel congestion This cause indicates that there is no appropriate circuit channel presently available to handle the call Cause No 35 Call Queued 5ESS Custom only This indicates the call has been queued Cause No 38 Net out of order This cause indicates that the network is not functioning correctly and that the condition is likely to last a relatively long period of time e g immediately re attempting the call is not likely to be successful Cause No 39 Permanent frame mode connection is out of service Q 850 only This cause indicates a calls has been preempted due to the circuit being reserved for reuse TECHNICAL DATA amp TROUBLESHOOTING 89 Cause No 40 Permanent frame mode connection is operational Q 850 only This cause indicates a calls has been preempted due to the circuit being reserved for reuse Cause No 41
24. IMPORTANT Remember that the U interface connects to a long telephone line and can convey lightning surges into the system Surge protectors intended for analog lines work to protect U interfaces We recommend that you install one for each line U interface Status LEDs The small green LED on the ISDN interface card near each U jack indicate the status of the corresponding ISDN circuit Rapid flashing about five times per second indicates a loss of the ISDN at the lowest physical level If the Talkshow System can contact the central office or an active U repeater the LED will blink slowly about once per second The LED will come on solid when handshaking is completed and all is OK If these LEDs do not light continuously once you have connected your ISDN circuits you may wish to skip ahead to Section 6 4 for help with troubleshooting the problem Nx6 Nx12 U interface RJ 11 PIN a uu A UN Note that the polarity of the line connections doesn t matter DESCRIPTION N C N C Line R LineT N C N C Hot Tip INSTALLATION Normally your telephone company should supply an RJ 11 style jack for the U interface This is the same 4 or 6 pin plug used for analog lines However sometimes you may find an 8 pin RJ 45 style jack and sometimes you may even be charged extra for it If you have this situation an RJ 11 plug can be inserted into the center of an RJ 45 jack to make things work If you discover that the Telco charged
25. Interface part 1701 00141 4 AES option card part 1701 00149 Fuse Replacement A fuse is located on the power supply PCB If fuse replacement is required note For continued protection against fire replace fuse only with same type and value Access to the Main PCB The main PCB closest to the bottom of the enclosure is mounted component side down You 86 Section 6 can gain access to the component side of the PCB without taking it out by removing the bottom plate of the enclosure box 6 7 ISDN Cause Phrases Values When there is an ISDN problem in the network a phrase appears in the ISDN protocol log that communicates information about the cause of the trouble See Section 7 10 for how to enable ISDN logging through the Telnet port These Cause Values are numbers generated by the network When possible we use those suggested by the Bellcore standard Note that in some cases there may be more than one meaning This can frequently be evaluated by whether the message has been received by the calling party or the called party CLASS 1 1 normal Cause No 1 Check number redial unallocated number This cause indicates that the called party cannot be reached because although the called party number is in a valid format it is not currently allocated assigned Cause No 2 No route to network Prefix 0 dialed in error This cause indicates that the equipment sending this cause has received a request to route the call
26. See CPC 112 Appendix 2 MOS Grade Mean Opinion Score Grade is a measure of codec quality ranging from 1 to 5 These are derived from subjective tests with listening panels comprised of both experts and people from the street who compare the coded audio to the original and grade the relative quality MSN DN 1822 Zephyr Xstream The Telco menu selections on the Xstream where the MSN Euro INS 64 ISDN or 7 digit Directory Numbers USA ISDN can be entered during set up The Zephyr Xstream generally does not require that these numbers be present See Directory Number and MSN MSN Multiple Subscriber Number Euro ISDN This is a telephone number associated with an ETS 300 BRI line Provid ers of ETS 300 often give you three MSNs with a BRI although additional MSNs can be purchased An ISDN terminal will ring provide an alerting signal only when calls are made to the MSN or MSNs entered in that terminal If a terminal has no MSNs entered it will ring whenever there is a call to any of the MSN s on that BRI See ETS 300 and DN National ISDN USA amp Canada The US standardized multi platform ISDN protocol The first version is National ISDN 1 As of mid 1996 National ISDN 2 has been implemented in some areas and is fully backward compatible with National ISDN 1 NCTE Network Channel Terminating Equipment NCTE is a general term that can be applied to a CSU or NT1 or other equip ment terminatin
27. T1 lines have certain restrictions as to the data allowed No more than 15 zeros shall be sent in a row and average density of 12 5 ones must be main tained The CSU is responsible to ensure that these requirements are met The line encoding method AMI or B8ZS determines exactly how these requirements are met while still allowing recovery of the original data at the far end Your Telco will determine the method used on a specific circuit B8ZS is preferred El circuits have similar restrictions HDB3 is preferred for E1 circuits Listed Directory Number See LDN Line Echo Canceller LEC When a two way digital path is interfaced to an analog line echoes result from the leakage of the send audio into the receive When combined with delay these are annoying and can disrupt conversation An LEC works to remove these echoes usually using an adaptive digital filter See Hybrid Line Encoding T1 See Line Coding T1 Line Format T1 Modern T1 circuits usually use either Superframe sometimes called SF or D4 or Extended Superframe sometimes called ESF line formatting The type of framing used is determined by your Telco ESF is preferred See ESF and SF Line Side This is the side of a central office switch that the subscriber s telephone lines are connected to The main reason for distinguishing between this and the trunk side is that certain customer related features Such as CLASS and Centrex features are inapplicable to trunks Se
28. These will require a pull up resistor to drive TTL style logic inputs Most equipment has the pull up built into the input but if there is no pull up you ll have to add one connect ing it from the output to the 5v source on pin 8 An appropriate value is 2 2kQ Sink pull down current must be limited to 400mA maximum per output with total output restricted to 1 amp 200mA each output if all five will be used If you want to drive a relay lamp or high current LED you will need an external power source Connect your external power source negative to Talkshow System ground at pin 1 and positive to the relay The other side of the relay goes to the output pin The Talkshow System s 5v output is fed through a 1000 resistor and a 50mA fuse so cannot source much current 2 11 AC mains Power The AC receptacle connects mains power to the unit with a standard IEC International Electrotechnical Committee power cord and provides an on off switch The power supply has a universal AC input accepting a range from 100 to 240 VAC 50 60 Hz IMPORTANT SAFETY INFORMATION Surge Protection Precautions should be taken to prevent damage caused by power surges Standard line surge protectors can be used to offer some degree of protection It is the user s responsibility to ensure that protection adequate for their conditions is provided Grounding This equipment is designed to be operated from a power source which includes a third ground i
29. This output will be active low when the User button is pressed It will remain asserted until that button is released Ringing Out This output is active low whenever the system has one or more lines in the ringing state Priority Ringing Out Active low when a line designated as a Priority Line is ringing Talkshow System GPIO D Connector PIN DESCRIPTION Ground User Out Dump 1 Ringing Out 1 Priority Ringing Out 1 Record Start 1 Ring Mute In 1 N C 5VDC 50mA max O ON AU A VN Record Stop 1 10 User Out Dump 2 11 Ringing Out 2 12 Priority Ringing Out 2 13 Ring Mute In 2 14 reserved 15 N C 8 il N Ol tears O PARALLEL CONTROL Note for the Curious The DB designation for D Sub connectors is from Cinch Corp s part numbering system The proper Cinch designation for a 15 pin D Sub connector is a DA 15p plug or DA 15s socket not DB 15 Our thanks to Mike Schweizer for contributing this tidbit 24 Section 2 Input Characteristics The Talkshow System s inputs are designed to be universal They accept either a voltage source up to 24VDC or a closure to ground which may be provided by switches relays or logic outputs In the latter case either totem pole or open collector will work The inputs are active low A built in 1k Ohm pull up resistor is provided Parallel logic input circuit 5V 30 v max Output Characteristics Open collector to ground
30. Web only Select sync to either 48khz internal clock or sync to the AES input DETAILED CONFIGURATION amp REFERENCE 53 5 5 The Telco Menu LCD panel and Web page options ISDN Protocol Selects the ISDN protocol which must correspond to the ISDN circuits you are using It will apply to all ISDN lines There are a number of choices but the most common are described below For Europe and many other parts of the world ETS 300 This is the very common Euro ISDN according to the ETS 300 standard For USA amp Canada and a few other parts of the world Natl 1 1 National ISDN 1 is the most common protocol for the USA amp Canada and some other parts of the world May also be used with lines conforming to National ISDN 2 See the notes about SPIDs below DMS Cust For custom functional lines on the Nortel DMS 100 switch See the notes about SPIDs below AT amp T Cust This is the AT amp T PTP protocol sometimes called AT amp T custom Point to Point This protocol generally does not allow two simultaneous voice calls For this reason we recommend that the National ISDN 1 protocol be used if you are on the AT amp T Lucent 5ESS Telco switch If you are in Europe or most of South America you should choose ETS 300 If you have Net64 ISDN in Japan you should select INS64 If you are in Hong Kong you should choose AT amp T Cust If you are in the USA or Canada use whichever protocol your phone company has
31. and compatibility through standards promoting interoperability such as GR 303 Telcordia also offers educational and training programs open to all interested parties Bellcore was sold to SAIC in 1997 Telcordia is responsive to both RBOCs and independent Telcos Their web site is http www telcordia com Telos Customer Support 1 216 622 0247 Call us at this number 24 hours a day You may also ask for assistance by emailing to sup port telos systems com Telos Test Line Xstream Zephyr amp ZephyrExpress 24 Hour test line which can be called to verify correct operation of your Zephyr or ZephyrExpress codec Numbers are 1 216 781 9310 and 1 216 781 9311 The test line transmits in dual mono mode at 32K Hz and can be received in L3 Mono using one B channel or L3 Stereo using 2 B channels Telos also maintains an identically configured test line in Freising Germany at 49 81 61 42 061 dial this number twice Terminal Adapter See TA Tie Line See Tie Trunk Tie trunk A Trunk between two PBXs Note a tie line is a dedi cated circuit not a switched circuit See Trunk TR 008 See SLC 96 Trouble Ticket A Telco work order used to track Customer Re pairs within the Telco If you call someone inside the Telco s repair department they will need this number to proceed It will also be needed whenever you call to check on the status of a repair Always ask for this number when initiating a repair request Trunk
32. be useful to clean up the terrible caller audio then the norm at radio stations The result of asking it was the Telos 10 the world s first digital telephone hybrid a product that changed the face of radio forever Looking back at Telos history What if is found at the root of many more technologies broadcasters have since made ubiquitous MP3 bit rates are perfectly matched to the bandwidth of ISDN what if we combined them What if we built a box that let radio stations stream audio on the Internet What if we could use Ethernet to treat audio as data and move it around the radio station in real time By the way if you recognized the answers to those questions as the Zephyr ISDN codec AudioActive MPEG encoder and Livewire IP Audio standard give yourself a cookie Today as technology roars ahead we re constantly seeking new ways to apply it to the problems of everyday broadcast ing Voice over IP VoIP is particularly compelling for its ability to packetize high quality phone calls and direct them anywhere in the broadcast plant using ubiquitous standard Ethernet as a backbone A new digital tech called AEC Advanced Echo Cancellation is helping to literally A LETTER FROMTHE PRESIDENT ix eliminate the age old problem of feedback in open speaker environments And the prize of using the public Internet for remote connections as reliable as those of switched circuits is within broadcasters grasp
33. by wrong Telco provisioning of the line at the time of installation Cause No 58 Bearer capability not presently available This caused indicates the users has requested a bearer capability which has been implemented by the equipment but is not available at this time CLASS 1 4 service or option not implemented Cause No 62 Inconsistency in designated outgoing address information Service Not Authorized VN4 This cause indicates an inconsistency in designated outgoing address information and subscriber class or This cause indicates that a Terminal Diversion in a VN4 network has been rejected Cause No 63 Service or option not available unspecified This cause is used to report a service or option not available only when no other cause in this class applies Cause No 65 Bearer Capability not implemented Incompatible bearcap This cause indicates that the equipment sending this cause does not support the bearer capabil ity requested Cause No 66 Channel type not implemented This cause is returned when the called party has reached a channel type not supported Cause No 69 Requested facility not implemented This cause indicates that the network or node does not support the requested bearer capability and therefore cannot be accessed at this time Cause No 70 Restricted only This cause indicates that the calling party has requested an unrestricted bearer service but the equipment sending this cause only
34. caller on the handset If you are using a headset attached to the Desktop Director this will adjust the level in the headset as well Other Menu options After selecting menu from the main LCD screen the following are offered Change Director Mode Select the mode that the phone will be used by the operator Producer for screening calls Talent for placing calls on the air Highlight your choice and press Select to apply your changes Select Show If the Nx phone system is configured in Split Show mode it can chose between SHOW1 or SHOW or be inactive see section 5 7 Highlight your choice and press Select to apply your changes Select Ringer It is possible to change the ringer tone on each Desktop director There are four different choices Press Select to apply your changes Configure Shows This gives access to the programming functions for two studio modes where each pair of hybrids is assigned to one of two studios See Section 5 6 Ignore Ring Mute When set to On this Desktop Director will still sound its ringer even if Ring Mute is asserted on the GPIO or by the Assistant Producer or Vx protocols Adjust LCD Panel Gives access to the same menu as the LCD button on the main screen USER OPERATION 37 Diagnostics This gives a status report of the connection between the Desktop Director and the Talkshow System unit to which it is connected Two values are reported Near End th
35. cannot be applied to fiber optic or other non metallic paths See also channel Circuit Switched Data See CSD Circuit Switching A system where a dedicated channel is allocated to the users of that call for the duration of that call That channel is allocated for the duration of the call regardless if information is being transmitted at any given moment Bandwidth through the channel is fixed at no time may this bandwidth be exceeded If this bandwidth is not used it is wasted While inherently inefficient the dependable and reliable nature of circuit switching makes it ideally suited to real time voice and audio video conferencing applica tions When over loaded Circuit Switched networks will respond all circuits are busy try again later This is in stark contrast packet switched networks or to systems where statistical multiplexing is used See Statistical Multiplexing CLASS Custom Local Area Signaling Services A variety of enhanced features usually on analog lines that take advantage of the ability of modern SS7 technology s ability to transmit informa tion about the calling party CLASS includes such features as Caller ID Automatic Callback Call Trace initiated by subscriber Selec tive Call Screening etc Clear Cause See Cause Code CLEC Competitive Local Exchange Carrier Your local telephone service provider who is one of the new generation providers rather than a RBOC or Independent A CLEC is really ju
36. column can use up to two hybrids with the first call on the left column connect ing to Hybrid A the second to Hybrid B the third to Hybrid A the fourth to Hybrid B etc Only POH input 1 is used 64 Section 5 4 Column This supports Desktop Directors with the add on module to make four line select columns Each column is connected to one of the four Talkshow System hybrids the leftmost to A the next to B etc The four inputs and outputs are only accessible via AES and Livewire and 4 Column mode cannot be used with the analog I O option This also supports a Euro style operation method where each of four Telco lines has its own hybrid and fader As this is written this mode is only possible with Euro phone modules installed in an Axia Element mixing console Only POH input 1 is used Sharing the Talkshow System Two studio Modes These modes allow a single Talkshow System to be used with two studios Telephone lines may either divided and used independently or be shared with both studios having access to the same lines Each Desktop Director Console Director and call control software application needs to be attached to a show See below for instructions on how to do this The POH 1 input goes to Show 1 and the POH 2 input to Show 2 Note When using the two studio modes you will need to use Separate for the Mix Minus selection in the Audio menu See Section 5 4 2 Show Split This option causes the system to be
37. dialing func tions and interfaces to the user s data equipment as well as to the NT1 on the S or T interface Tandem Switch A switch which is between two others It connects two trunks together Long distance calls on a LEC line go through a long distance tandem that passes them through to the long dis tance provider s switch Tandem Tie Trunk Switching When a PBX switch allows a tie line call to dial out of the switch For example if switch A in Arkansas has a tie line to switch B in Boise Boise could use the tie line to make calls from switch A TD Transmit Data Data coming from the DTE towards the DCE or network Also a light on a modem or CSU DSU that lights to indicate presence of this signal TEI Terminal Endpoint Identifier An OSI Layer 2 identi fier used by an ISDN terminal to communicate with the serving CO Most equipment uses a dynamic TEI assignment procedure Telco Telephone Company Your local telephone service provider In the 21st century you generally have a choice of Telcos if you are a business in a major metropolitan area in the USA Competition is coming to the Telecom industry around the world Telcordia Technologies Formerly Bellcore The research and development organization owned by the telephone companies Telcordia represents the phone companies in developing standards for Telco equipment and in testing equipment compliance to those standards Promotes competition
38. ends are wired identically The outside pairs are not required and may be omit ted 2 6 Integration with a PBX A Telos Talkshow System will usually be used independently of any other phone system in your facility But it is possible to have one or more of the lines come off another phone system Depending upon the PBX and your preferences this can be either ISDN BRI or POTS You might be able to save money by using a T1 E1 or ISDN PRI as your Telco link and having the PBX generate local ISDN BRI or POTS lines for the Talkshow System Consider having at least one line directly from the Telco to keep your system up and running even if the PBX or its Telco connection goes down Using ISDN Telos equipment has been used satisfactorily using BRI off the following PBX Switches USA and Canada Nortel Meridian Option 11 2 wire U interface Protocol ID 6 user report Lucent Avaya Definity 2 wire U interface user report Avaya IP Office must use Q 931 mu setting on Telos user report Outside of the USA Siemens HiCom series 4 wire S interface user report ETS300 Euro ISDN is available from many PBX systems outside the USA and compatibility is generally good INSTALLATION 11 Unless your PBX is listed above its support of US Canada National ISDN 1 is not certain We recommend you arrange a test for compatibility before committing to the purchase of the PBX ISDN module Contact Telos support for the latest
39. feed from the console All callers hear the one console mix minus feed and each other The following table summarizes the effect of each mix minus mode together with each Show mode Show Mode One show Show Mode One show 4 column Show Mode Two show Mix Minus Mode Separate 2 in 2 out 1 POH Use analog AES or LW For AES POH may be analog or AES 4 in 4 out 1 POH Use AES in 4 channel mode or LW to access all the inputs and outputs For AES POH remains analog Per show 2 in 2 out 1 POH Use AES in 4 channel mode or LW to access all the outputs For AES POH remains analog Mix Minus Mode 1 in Multiple out 1 in 2 out 1 POH Use analog AES or LW For AES POH may be analog or AES 1 in 4 out 1 POH Use AES in 4 channel mode or LW to access all the outputs With AES in 4 channel POH remains analog Per Show 1 in 2 out 1 POH Use AES in 4 channel mode or LW to access all the outputs For AES POH remains analog Mix Minus Mode 1in 1 out 1in 1 out 1 POH Use analog AES or LW For AES POH may be analog or AES 1in 1 out 1 POH Use analog AES or LW For AES POH may be analog or AES Per show 1 in 1 out 1 POH Use analog AES or LW For AES POH may be analog or AES Livewire can always be used for any of the four caller and two POH inputs When a Livewire input is enabled the corresponding analog AES input is disabled Livewire can
40. flow to know when the attached device goes off hook It must be able to generate ringing current Glare On a POTS line an incoming call is signaled by periodi cally applying an AC ring voltage to the line Since there is a semi random period before the ring and pauses between rings it is pos sible to seize a line which is about to ring and answer a call when attempting to place an outgoing call When this scenario happens it is called glare Glare is much less likely if Ground Start or ISDN trunks are used See also Ground Start Trunk GR 303 See SLC 96 Grade of service This is simply the ratio of calls blocked to total calls in a decimal form Therefore a grade of service of P 08 would represent 8 blocking Telephone tariffs regulate the acceptable av erage grade of service which must be provided on public networks See also Blocking Ground Start Trunk A type of telephone trunk where the request to make an outgoing call i e request for dial tone is made by briefly grounding the Tip conductor Many PBX system use ground start trunks as they are less prone to glare than Loop start trunks Ground start lines are sometimes used with equipment designed for Loop Start lines This may or may not work generally it serves to prevent outgoing calls while incoming calls work normally Telcos may call these ground start lines See Loop Start Trunk See also Glare HDB3 High Density Bipolar 3 An E1 line coding m
41. for a long time There are devices on the market that convert from PRI Telco lines to BRI for internal use so this is a way that Nx Talkshow Systems could be used long into the future on ISDN Sharing Phone Lines One Nx Talkshow system Two Studio Mode POTS and ISDN lines Nx Talkshow Systems can be used with two studios independently Lines may be independent or shared within the Nx Talkshow system Each studio has two hybrids There are two Program 4 Section 1 on Hold inputs one for each studio Two sets of Telos Assistant Producer application can be used independently Note the following With analog audio in out there are only two total inputs and outputs So Nx Talkshow Systems must be used in a mode that combines the two hybrids assigned to each studio into one input output The AES option provides four input and four output channels total so all hybrids are accessible with each studio using one AES connection s two channels Program on Hold remains an analog input Livewire provides the full four hybrid input channels four hybrid output channels and two Program on Hold inputs Two Nx Talkshow systems Only for POTS lines not ISDN lines Facilities using POTS lines to the Nx Talkshow system can also share POTS lines that are bridged between two Nx Talkshow units This does not work with ISDN lines as you cannot bridge them like POTS lines This gives the capability for either 2 studios to have a
42. hold inputs and all needed control Making do with an older console Consoles made before around 1990 rarely had good support for mix minuses and almost never for more than one or two With one of these oldsters some clever improvisation is going to be needed Here we describe a possible scenario that can be used as a starting point for your situation We assume an older console with Program and Audition as the main busses There is another bus of some kind that can be adapted for mix minus application We ll call this the Utility bus All sources including the hybrid will be assigned to Program so the audience can hear them as usual We will also assign most of these sources to Utility as well just never the fader with the hybrid s own audio INSTALLATION 13 This arrangement is flexible allowing the operator to place any or all sources in Utility for the caller to hear In our example we have the fortunate case that the console permits the Utility bus to be fed pre fader letting the announcer easily use the telephone system for off air conversations PGM Bus r PGM Out Utility Bus gt To Rec Ch 1 O O To Rec Ch 2 A recorder can be attached to the Utility and hybrid outputs to record announcer phone audio This is often done as shown here with each signal to a separate track A drawback is the potential for the operator to accidentally put the hybrid in Utility in which case it is no longer a mix m
43. indicates that the equipment sending this cause has received a request to establish a call which has low layer compatibility high layer compatibility or other compatibility attributes e g data rate DN subaddress which cannot be accommodated This call can also be returned by a switch to a CPE when trying to route a call to an incompatible facility or one without data rate Cause No 90 Destination address incomplete Non existent CUG This cause indicates that the destination address is missing or incomplete or Indicates a non existent Closed User Group Cause No 91 Invalid transit network selection This cause indicates that an Invalid transit network selection has been requested Cause No 92 Invalid supplementary service parameter VN4 only Used to indicate an invalid parameter for supplementary service Cause No 95 Invalid message unspecified This cause indicates that the entity sending this cause has received and Invalid message only when no other cause in this class applies CLASS 1 6 protocol error e g unknown message Cause No 96 Mandatory Info missing This cause indicates that the equipment sending this cause has received a message which is missing an information element which must be present in the message before that message can be processed Cause No 97 Message type non existent or notimplemented This cause indicates that the equipment sending this cause has received a message with a mes
44. of the screen If it says Producer follow the directions in Section 4 1 to change this Director to Talent mode Next pick up the handset and press the left column line 1 button to select the line for dialing You should now hear dial tone on the handset Enter the phone number for a telephone line which has audio present using the keypad calling either the time or weather service works well for this Press the left column line 1 button and hang up the handset to place the call on the Phone 1 output Bring up Phone 1 on the console You should hear the audio from the phone line Look at the Input 1 meter on the Talkshow System and verify that this audio is not present If the phone audio is seen here you are not feeding a mix minus and you must fix this problem before you proceed IMPORTANT lif you are using one of the mix minus schemes that require only a single mix minus you should not have seen audio on either the 1 or 2 bargraphs in the previous test In that case you can skip the next test See Sections 2 7 and 5 4 for detailed information about the Talkshow System s mix minus options Now we will verify the second mix minus Press right column line 2 button to move the call Bring up Phone 2 on the console You should hear the audio from the phone line 30 Section 3 Look at the Input 2 meter on the Talkshow System and verify that this phone audio is not indicated here If phone audio is present here
45. other call screening software is to be used Connect any needed parallel GPIO This Installation section covers all of the above After completing these steps you will be ready to move on to configuration and testing 2 2 Connections to the Telco circuits The Telco interface module on the Nx6 supports up to six lines It has a second slot available which can take a second six line card converting the Nx6 to a 12 line hybrid Any combination of POTS or ISDN cards can be used The Nx12 comes with both cards installed and is ready to receive 12 lines If your Talkshow System will be behind a PBX private telephone system see Section 2 6 for more on this integration 6 Section 2 IMPORTANT Being a broadcast station you will probably have high call volume If your PBX has a PRI or T 1 coming from the phone company that will be shared for business operations it may need to be programmed to reserve some capacity for those calls One option would be to have the Telco provision configure the PRI or T 1 into separate trunk groups for the Business and On air telephone numbers You should discuss this with both your local telephone service provider and your PBX provider You should also inform your Telco that this will be a high volume application so they can be sure that your contests will not interfere with 911 and other emergency services 2 3 Connecting Talkshow Systems to Analog Lines Both the Nx6 and Nx12 use
46. outputs to each console and one mix minus caller send from each console Hybrid A gt Hybrid B E p Hybrid B a In 4 Column show mode you have 4 independent caller outputs to 4 faders on the console but require only 1 mix minus feed All callers hear the one console mix minus feed and each other One Show or 1st of two Shows Input 1 di Output 1 Output 2 Second of two Shows Input 2 Output 3 Output 4 1in 1 out This mix minus option means each console only has to make a single mix minus feed for the caller send audio and the Talkshow System makes only one mixed caller audio output of all callers to one fader on the console Like the 1 in multiple out mode this option sends the single mix minus feed from the console to all callers Its main purpose is to accommodate older consoles that have only one mix minus or when only one fader is available for telephone calls The disadvantage is that the operator has no control of the relative levels of the multiple callers It relies upon the Talkshow a a ALNAN a gt 11 1 aad 1 Hybrid A E gt Hybrid B a 1 1 us Output 1 Second of two Shows gt Hybrid A ii gt Hybrid B i Input 2 DETAILED CONFIGURATION amp REFERENCE 51 Output 2 With 4 Column Show mode you will have one caller output to the console containing a combined mix of all callers audio and require only one mix minus
47. package More and more Telcos are using ISDN Ordering Code IOC capability packages for ISDN ordering National ISDN protocol only If your telephone company uses these you need only tell them you need IOC package S and an RJ 11 style SJA48 jack We still recommend that you fax them these forms as well Hunt Groups Just as with analog lines ISDN BRI lines can be placed in a hunt group where a single number can be given to callers and the calls will roll over from busy channels to idle channels This is sometimes called ISG incoming service grouping or rollover lines This would usually be a requirement for an on air phone system You may choose to exclude 1 or 2 channels from the hunt group i e the Hot line and the Warm line Be sure to specify to your Telco how you expect this to work FAXABLE ISDN ORDER FORM for the USA amp Canada The following form can be used to place orders for ISDN lines in the USA amp Canada Com plete the top portion of the form and send all three pages to the phone company We suggest that you keep the form handy and show it to the installer when he or she puts in your line and ask the installer to verify with the switch programmer at the central office that your line is configured as ordered You only need to send the first page if the Telco uses IOC ISDN Order Code Capability packages for ISDN specification 120 Appendix 3 ISDN BRI LINE ORDER USA amp Canada Telos
48. possible 4 total independent hybrids per studio OR each Nx Talkshow system is sharing their lines between two studios each effectively having four studios sharing the same phone lines Line status is communicated between the two Nx Talkshow systems via Ethernet A Step up in Quality and Flexibility Whether being used with analog or ISDN lines Nx Talkshow Systems take on air telephone operations beyond the ordinary Its four hybrids let you have excellent conferencing quality with four simultaneous callers Or you can serve two studios with one Talkshow System with two hybrids for each The hybrids in these units include long tail echo cancellation to improve quality on calls originating from VoIP or mobile telephones Each hybrid is complemented by powerful signal processing functions to extract the best possible quality from real world phone lines There are filters AGCs multi band spectral equalizers and duckers Sophisticated feedback reduction processing using both acoustic echo cancellation and frequency shifting enhances performance when an open microphone to loudspeaker acoustic path is required Telos Desktop Directors and Call Controllers offer an easy to use but powerful interface for operators The Status Symbol icons provide clear information as to line status with high contrast LED displays Telos Assistant Producer software package gives you what you need for call screening and communication between the producer and
49. provide inputs 3 and 4 when they are not otherwise available at the XLR connectors Four caller audio outputs via Livewire are always possible regardless of mode in parallel and at the same time as the XLR audio outputs AES Mode On the Web interface the Setup Wizard can be used to set the AES mode If your Talkshow System is equipped with the optional AES module there are two AES input and two output XLRs which may be configured as follows 2 channel default 52 Section 5 4 In this mode there are up to two caller inputs outputs The two AES input XLRs can either take one audio input each using only the L chan nels or a combined L R AES signal may be connected to the first AES input XLR The Talkshow System will automatically sense if there is a second AES connection to XLR input 2 If present the Talkshow System will take the second audio input from this XLR If not present the Talkshow System will take the second audio input from the R channel of AES input 1 If your mixing console combines two caller sends into one AES L R plug this into the first AES input XLR and use the first AES output XLR for the caller audio If your console uses two separate AES connections use both XLR inputs and outputs Program on Hold is via the analog inputs 4 channel In this mode there are up to four caller inputs outputs The L and R of each AES signal are used creating four independent caller inputs and four caller ou
50. refers to when one privately owned phone system is tied to another privately owned phone system The most common application is when a key system is connected to analog ports of a PBX When it involves one PBX behind another it is a limited Tandem application See Tandem Switch and Tandem Tie Trunk Switching below Bell Labs The basic research facility that was AT amp T s primary research facility Bell Labs was originally part of Western Electric and was spun off with Lucent Technologies when AT amp T was broken up in 1984 Many very important discoveries were made at Bell Labs including the transistor communications theory and radio astronomy The future of Bell Labs seems bleak at the time of this writing Bell Operating Company See BOC See also RBOC Bellcore BELL COmmunications REsearch See Telcordia The research and development organization owned by the RBOCs Bellcore represents the RBOCs in developing standards for Telco equipment and in testing equipment compliance to those standards Bellcore also offers educational and training programs open to all interested parties Now Telcordia BERT 1 Bit Error Rate Test A test for digital lines which in volves looping a data path and sending a test pattern Data return ing is compared to the sent data to check for errors Depending on the Test Pattern used BERTs may or may not uncover problems A line which only has occasional problems will need a BERT of sufficient
51. supports the restricted version of the requested bearer capability Cause No 79 Service or option not implemented unspecified This cause is used to report a service or option not implemented event only when no other cause in this class applies CLASS 1 5 invalid message e g parameter out of range Cause No 81 Invalid call reference value This cause indicates that the equipment sending this cause has received a message with a call reference which is not currently in use on the user network interface This value only applies if the call reference values 1 or 2 octets long and is not the global call reference Cause No 82 Identified channel does not exist This cause is sent when the equipment sending this cause has received a request to use a channel not active on the interface for a call TECHNICAL DATA amp TROUBLESHOOTING 91 Cause No 83 A suspended call exists but this call s identity does not Cause No 84 Call identity in use This cause indicates that a call identity is in use Cause No 85 No call suspended This cause indicates that no call is suspended Cause No 86 Requested call identity cleared This cause indicates that the call having the requested called entity has cleared Cause No 87 Destination not a member of CUG This cause indicates that the address of an outgoing call attempt is not a member of the Closed User Group Cause No 88 Incompatible destination This cause
52. talent Because it connects via Ethernet IP it can easily work from remote sites Because the Telos control protocol is open and published third party software has become available offering yet more capability such as database look up of caller information and storage of contest results Livewire integration is standard When used with a compatible network and mixing consoles a single RJ 45 connects all audio and control signals With the power of the network new possibilities open such as displaying caller ID on the console s monitor Installation 2 1 Getting Started Nx Talkshow Systems mount in a 2RU high space in a standard 19 rack You do not need to pay special attention to cooling or ventilation as the unit will operate in any environment where the stirred air temperature around the unit is between 0 to 40 degrees Celsius 32 to 104 degrees Fahrenheit with a relative humidly of 0 to 98 non condensing Next installation steps are Connect the Telco circuits Learn about telephone service options Connect audio via either analog AES3 or Livewire Learn about mix minus Connect your Desktop Directors If needed connect a PC for Web user interface either directly cross over cable or via a LAN local area network You can use the front panel LCD button interface for most operations But you will need the Web interface for ISDN and Livewire configuration as well as the LAN connection if Telos Assistant Producer or
53. thanks to ever more sophisti cated coding algorithms Pioneering ideas aren t limited just to technology though We believe that investing in people pays off big so we ve assembled the largest R amp D team in the industry a talented staff of engineers scientists and broadcast professionals We re also the first and only broadcast equipment maker with a 24 7 support team ready to offer assistance any time of the day or night As they say radio never sleeps so neither do we As Telos navigates its third decade in broadcasting s choppy waters the culture of innovation and creativity that propelled Steve in the early days still guides us today We re still in love with radio still turned on by new technology still driven to ask What if If you re reading this you are too for which we thank you from the bottom of our hearts Michael Catfish Dosch President Telos Systems Introduction Welcome to Nx Talkshow Systems The Nx family of Talkshow Systems is setting new standards for studio telephone interface systems The latest generation of hybrid technology and processing by Omnia results in the cleanest and most consistent audio quality ever The Nx family of Talkshow Systems includes two devices The Nx12 which can accommodate 12 lines and the Nx6 which is a 6 line version of the Nx12 Throughout this manual We will refer to these devices collectively as the Talkshow System In instances w
54. the longest will be selected Note that if the handset is off hook the caller will be directed to the handset using the same priorities outlined above Producer Mode The Next button will select the line which has been ringing longest If no line is ringing it selects the line that has been on normal hold longest When Next is pressed any unlocked caller will be dropped It is also possible to configure the system to put the caller back hold instead See the Direct Hold feature Section 5 6 Block All ES Pressing this key will clear all lines except lines on air on the handset on hold and priority lines such as the studio hotline This allows you to prevent early callers from getting in on contests until after the DJ has made the announcement and turned off block all Ringing lines other than lines designated priority will be dropped and blocked from accepting any calls For ISDN lines the caller may get a busy signal or they may continue to hear ringing For analog lines the ringing calls will be connected to a silent audio channel Further calls will get a busy signal Pressing Block All again will release the lines and allow incoming calls Dialing Keypad These are used for dialing out which can be on the handset on the speakerphone or on air They remain active during a call for voice mail and other services Function Buttons Speakerphone Button This button places a call that is on the handset ont
55. through a particular transit network which it does not recognize The equipment sending this cause does not recognize the transit network either because the transit network does not exist or because that particular transit network while it does exist does not serve the equipment which is sending this cause or This cause indicates the prefix O is invalid for the entered number Cause No 3 No route to dest Prefix 1 dialed in error This cause indicates that the called party cannot be reached because the network through which the call has been routed does not serve the destination desired This cause is supported on a network dependent basis or A 1 was dialed when not required Redial without the 1 Cause No 4 No prefix 1 Special Information Tone The prefix 1 is not required for this number or This cause indicates that the called party cannot be reached for reasons that are of a long term nature and that the special information tone should be returned to the calling party Cause No 5 Misdialed trunk prefix National use This cause indicates the erroneous inclusion of a trunk prefix in the called party number Cause No 6 Channel unacceptable This cause indicates a called user cannot negotiate for a B channel other than that specified in the SETUP message Cause No 7 Call awarded and being delivered on an established channel This call indicates the user has been awarded the incoming calls and that call is b
56. time duration to catch that intermittent problem A five minute BERT of an ISDN BRI circuit will only catch severe prob lems 2 Bit Error Rate Tester The test equipment used to perform a Bit Error Rate Test Billing Telephone Number The main phone number which all calls on hunt group or a PRI are billed to This information may be required when configuring a PRI PBX Bit Error Rate The basic measure of errors on digital transmission paths It is usually expressed as the number of errors per number of bits For example the allowable bit error rate on a BRI circuit is 1 x 10 E7 one bit error in 107 bits or 1 error in 10 million bits Bit Error Rate Test See BERT Bit Rate The capacity of a digital channel ISDN calls are set up at a given bit rate either 64Kbps or 56Kbps The bit rate of a circuit switched connection cannot be changed during a call See Kbps BLEC Building Local Exchange Carrier A LEC who covers the occupants of a single building or a small group of buildings only Often Telecom services are provided by a BLEC as a service 106 Appendix 2 or incentive to potential tenants Ifa BLEC offers Long Distance Service it is covered by the same regulations as any other LEC Blocking When a circuit switched call cannot be completed The percentage of blocked calls to the number of calls attempted forms the basis of a statistic called grade of service While it is economi cally infeasible to build a network t
57. to find someone who knows what ISDN is While your usual account agent will be the normal entry point you may be talking to a number of phone people before you find one who understands your needs Some of the regional Bell companies offer a single point of contact number for switched digital services while some Telcos use resellers Note that in some cases the phone company will need to do a loop qualification line loss test before they can verify that ISDN will be available at your location 118 Appendix 3 Details Details Protocols In a perfect world all ISDN terminal equipment would work with all ISDN lines without regard for such arcana as 5ESS DMS100 CSV CSD SPIDs etc Unfortunately this is not always the case At their central offices the telephone companies use either AT amp T Lucent 5ESS Northern Telecom DMS100 or Siemens EWSD switching equipment While each will work with the Talkshow System there are some differences which need to be taken into account when lines are ordered and used Each has a protocol the language the user equipment and the telephone network use to converse on the D channel for setting up calls and the like There is a standard protocol that all switches may provide called National ISDN 1 NI 1 This protocol was standardized and specified by Bellcore the technical lab jointly owned by the phone companies However both AT amp T and Northern Telecom had versions of
58. value will not make the send louder due to the AGC limiter Input Setting Related to Input Level Input Setting 10dBV OdBu 4dBu 9dBu PPM 9dBfs PPM Analog input level 10dBVnom OdBunom 4 dBu nom 9 dBu PML n a Livewire AES 32dBfsnom 24dBfsnom 20dBfsnom 15 dBfs 9 dBfs PML The ballistic time characteristic of Talkshow System s meters changes depending on the Input Level setting so to match the expected console meter type When 10 0 or 4dBu is selected the meter has a VU standard integration time When 9dBu PPM or 9dBfs PPM are selected the meter has a PPM time characteristic The table above shows the level at the green red transition on the Talkshow System meter For VU this is the nominal 0 value for PPM this is the Permitted Maximum Level PML value The usual setting is 4dBu The 9dBu PPM setting is intended for Euro analog installations The 9dBfs setting is intended for Euro digital installations If Livewire inputs are used this should usually be set to 4dBu for the 20dBfs Livewire standard nominal level Europeans could choose the 9dBfs PPM setting if they are operating their Livewire network according to this standard Nominal Output Level This adjusts the level of analog AES and Livewire outputs Output Setting Related to Output Level Output Gain Setting 10dBV OdBu 4dBu 9dBuPPM 9dBfs PPM Analog output level 10dBVnom OdBunom 4dBunom 9dBuPML n a Livewire AES 32dBfsnom 24dBfsnom 20dBfsn
59. which will appear to be a line on the Desktop Director SLOT A SPIDs USA Canada only Telephone DN MSN BRI 1Line 1 Line 2 BRI 2 Line 3 Line 4 BRI 3 Line 5 Line 6 You will need to use the Web interface for ISDN configuration The following are step by step instructions See Section 5 3 to learn how to use the Web interface For additional information and numerous tips on ISDN see Sections 2 4 and 5 5 1 Open your Web browser and point it to the Talkshow System Click on the Telco menu 2 With ISDN modules installed in both Telco interface positions you will see a page similar to the one below ISON Telco settings Telco Protocol Natl1 1 v When ISON clock lost Hold last clock iv ISDN Interface A SPID 1 MSN ON 1 SPID 2 MSN ON 2 SPID 3 MSN ON 3 SPID 4 MSN ON 4 SPID 5 MSN ON 5 SPID 6 MSN ON 6 BASIC CONFIGURATION 27 3 Choose the ISDN Protocol For the USA and Canada this will usually be Natl I 1 For Europe and much of the rest of the world it will be ETS300 4 For Natl I 1 enter the SPIDs for each voice channel given to you by the telephone company Leave this blank for other protocols 5 Enter the DN Directory Number or MSN for each voice channel The DNs sometimes called Listed Directory Numbers or LDNs are your usually 7 digit phone number as listed in the telephone directory They normally do not include the area code 6 In some cases you may need to enter 10 digits for your DNs
60. which mode you are in You will see Producer if you are in this operation mode 38 Section 4 Answering calls You use the left column of line buttons to answer calls on the handset or headset The Next button will select the line which has been ringing longest If no line is ringing it selects the line which has been on normal hold longest When the Next button is pressed any lines not locked will be dropped or put on hold if your system is configured for direct hold operation Normal Hold The left hold button places a call on normal hold Screened Hold The right hold button places a call on screened hold This button s keycap may be labeled with a check mark if the Desktop Director will be used primarily for screening Designating the Next Call Normally when the talent presses his Next button the call waiting on hold for the longest time is selected to air However the Producer can force any line to the top of the queue You use the right column of line buttons to select which call is designated as next the call that goes on air when the talent presses his Next button The line status icon will change to the blinking check mark Hot Tip You can use this feature to re order the next queue at any time Simply press the right line buttons for the calls in reverse order from the order you wish to air them Press the right line button first for the call to be aired last then the button for the next l
61. you are not feeding hybrid 2 a mix minus and you must fix this problem before you proceed Drop the call by pushing the Drop button Ok now we just need to test the system as whole We suggest calling your spouse or partner and reporting that you will be home for dinner after all 1 7 8 Confirm that the Desktop Director you will be using says Talent at the top of the screen If it says Producer follow the directions in Section 4 4 to change this Director to Talent mode Press the left column line 1 button to select that line for dialing You should now hear dial tone through the mixing console Enter a phone number to make a call Bring up Phone 1 on the console You will hear the audio from the phone line Put on some headphones bring up the microphone and greet the party at the far end Check that your mix minus level is OK on the Talkshow System s Input 1 meter Your voice should sound normal in your headphones without any tinny or hollow artifacts and the caller should sound clear Move the call to Phone 2 by pressing the right column line 1 Repeat steps 4 through 6 using lines on the right column What now Your basic installation is complete Next you will want to look over other sections of this manual to familiarize yourself with the system and to set up the following parameters Desktop Director operation Receive EQ AGC amp Noise Gate Caller ducking adjustment Feedback reduct
62. 0 Sealing Current Unlike telegraphy teletypewriter and POTS lines most digital lines such as ISDN use a voltage rather than current mode of operation Sealing Current allows a controlled amount of current to be passed through a telecom circuit for purposes of healing resistive faults caused by corrosion Bellcore specifies sealing current on the ISDN U interface in the USA The Siemens EWSD switch does not provide sealing current Most other ISDN capable switches used in North America do SF Superframe A type of Line format supported on T1 circuits The Telco determines the line format and line encoding of your line ESF is the preferred Line Format on T1 circuits See Line Format Silence Suppression See Statistical Multiplexing SLC 96 A Subscriber Loop Carrier Circuit system manufactured by AT amp T now Lucent SLC 96 has its own version of T1 framing between it and the CO SLC 96 and similar SLIC systems may or may not perform a concentration function The interface is the Bell core TR 008 or the newer GR 303 interfaces that are specialized versions of T1 intended to allow transparent transport of analog CLASS features such as Caller ID Call Waiting etc The GR 303 interface is specifically intended to be used as a common point of interconnection between alternative equipment technologies and or networks i e voice over DSL voice over IP etc and the public switched network See the following link for addi
63. 1 Bank 0 ver 0 6 4 build Fri Jul 27 17 45 30 EDT 2007 Opank 1 empty update Warning System will reboot after changing current bank Do this by clicking the commit this version to Bank 0 check box followed by Apply 5 11 Backup and Restore Web Only You can save and restore your basic Nx system settings via a text file Select Save As to be prompted to save to a location of your choice Load the settings by browsing to the file to select it Note This option does not save the unit s network or Livewire settings The software package Axia iProbe can backup the network and Livewire settings but not the basic phone settings If you have iProbe then a combination of backup restore between this web interface and iProbe would fully recover the Nx settings Else you can manually reconfigure these options with ease thru the web interface and the front panel 5 12 Other Uses of the IP Ethernet Interface In addition to the Web user interface the Ethernet port can be used to support call control software applications and a telnet text based remote user interface Call control Software Client software such as Telos Assistant Producer and other 3rd party applications such as NeoWinners from NeoSoft can connect to 1P Port socket 9998 decimal If you desire to write your own application to access the Nx6 via this protocol please visit our website at www telos systems com where you can download the specifications of this protoc
64. 1 2 3 4 to FULL We suggest you start with the default value FULL Aggressive AGC is more noticeable but is also more effective at maintaining consistent levels Adjustment of this setting simultaneously changes as number of parameters within the AGC function such as attack amp release times thresholds and compression ratio An important feature of the Talkshow System s AGC is that it is cross coupled to other sec tions of the hybrid so that it can reliably distinguish between the real caller audio and residual hybrid leakage This allows a lot of gain on low level callers while preserving excellent hybrid performance 48 Section 5 Noise Gate Enables or disables a noise gate applied to the caller audio This is a downward expander that has the effect of reducing the noise on the caller audio when it falls below a threshold On some very low level calls this attenuation may be inappropriate because you may be able to hear the caller fade away when the level falls below the threshold It has three settings as follows Off Disables the noise gate Normal A slow setting with moderate noise attenuation Aggressive A faster setting with deeper attenuation for lines where noise is noticeable during normal conversation Receive EQ Caller Dynamic Digital EQ DDEQ Telephone audio frequency response varies widely since many factors can affect it we ve measured the response on a number of calls and the res
65. 2 13 14 15 Hardware Mapped Pin 1 69 70 Section 5 To use the Nx s LiveWire GPIO section to control other remote devices such as an external flasher or start stop on a recording device you would map the Nx s GPIO port to another Axia device s GPIO port in order to get a physical port on which to interface your electronics To do this we would map each port directly by IP address and port number Within the Livewire GPIO web page enter in the Channel field the IP address of the GPIO unit you wish to com municate with plus the port number using this syntax IPaddress Portnumber eg 192 168 0 220 5 For this example the Nx Talkshow system would be talking to the noted device on GPIO Port number five For the remote Axia GPIO device you would enter the IP address of the Nx Talkshow system and port 1 or 2 which corresponds to Show 1 or 2 For Nx Talkshow systems configured for a single show mode only GPIO port 1 will be needed One can still assign a channel number to the GPIO ports in the Nx Talkshow system and allow a Livewire enabled device with the same channel number to control or be controlled by the Nx system but that device must follow the noted logic functions above You would NOT enter the same channel number as the Nx s Livewire stream number s as this could cause adverse behavior between Axia consoles and the Nx unit itself Note The Nx Talkshow system does not generate or advertize a GPIO channel numb
66. 3 protocol Hunt Groups please give us the option specified below Ol Please put all MSNs for all BRT s in a single hunt ISG or Rollover group C Please create a hunt group with all MSNs except Specify CL No hunt groups Line Deactivation please provide the option specified below C Our application is critical and we need to ensure that lines are working at all times Please set the switch parameter Deactivation Allow to NO CL We would prefer to know if lines are working at all times Therefore we request that you set the parameter Deactivation Allow to NO if this is possible C Line deactivation is acceptable Please provide your default setting for this parameter You may call the manufacturer of the Nx6 Nx12 Telos Systems at 1 216 241 7225 for any additional information about ISDN compatibility Ask for ISDN Customer Support
67. 40 Section 4 If you take another line on the same column any locked lines will remain on air When a locked line button is pressed the call reverts to normal and the left or right pointing arrow icon is displayed The Drop and Hold buttons have no effect on a line in locked status but apply to any other active calls Using the lock feature you can organize any combination of calls to be conferenced You can also drop or add calls to an active conference group Locking VIP calls As described above any line on the air can be locked by hitting the line button a second time This is a handy feature for use with VIP guest callers by locking your guest you cannot drop the call by accident no matter how crazy things get in the studio Using the Director as a Telephone Making a call using the handset Pick up the handset select a line using the left column and dial If you wish to place this caller on air you can press a left or right line column button If you hang up the handset the caller will be lost unless you have placed them on hold or on air Using the studio microphone and cue speaker This is identical to placing callers on air just be sure that your microphone and phone faders are in cue not program If you experience feedback lower the cue speaker volume Making calls using the built in speakerphone Press the Speakerphone button select a line using the left column and dial If you wish to place this calle
68. 95 AUGIO 4 snc casi ad Be a dd o dos 95 Audio Performanc esti ara a 96 viii ISDN Telephone Line Connectivity o o o ooo o 97 Analog Telephone Line Connectivity 2 22 ee ee ee 97 Desktop Director Ports 2 ee ee 97 Control PORS ck eh Aa eR oe daa as 97 Two Studio Modes s sso ocs creeren enire rnn ee 97 Four Column Four Hybrid Mode o o o o o o ee 98 Euro Operation Mode lt gt o oe ossa eee e 98 8 Warranty and Application Caution 99 A1 USA SPIDs by Telco 101 A2 Telephone Terminology Guide 105 A3 USA ISDN Ordering Guide 117 Dealing with the PhoneCompany 20004 117 Details Details gsc 33 s bP Bb eS ew ewe RE ERS 118 FAXABLE ISDN ORDER FORM for the USA Canada 119 A4 Euro ISDN Ordering Guide 123 Details Details s s ee ee ee 123 FAXABLE ISDN ORDER FORM for Euro ETS 300 124 A Letter from the President of Telos What if A singularly great question because it opens the door to so many other questions and ideas Those who ask it are sometimes called dreamers as if that were somehow bad but history has often proven dreamers to be the avatars of new and exciting things What if was the query asked by Steve Church that led to the start of Telos Systems In 1984 Steve then a young broadcast engineer and part time talk show host won dered whether emerging Digital Signal Processing DSP technology might
69. Desktop Director Assembly Connections The black curly cord plugs into the handset The other end of this cord should be plugged into the small 4 position 4 pin miniature modular RJ 22 style jack near the front left of the unit This jack is identified with a telephone handset icon and is closest to the front of the Director The second jack at this location can be used to directly connect a telephone headset and is identified with a headset icon Route these cables through the small slots to the left so they do not interfere with the unit sitting flat on the table When a headset is attached it will automatically mute the handset microphone to prevent extraneous noise Disconnecting the quick connect inline connector on the headset cable will re enable the handset microphone 22 Section 2 Tip We have tested and recommend the Plantronics www plantronics com model P51 U10P Supra Polaris noise canceling unamplified monaural or model P61 U10P Supra Polaris noise canceling unamplified binaural headsets for direct connection to the Desktop Director Other headsets can be used however an external amplifier interface would be required between Desktop Director s handset jack and the handset to interface to the headset Line Designation Strip The Desktop Director ships with several pre cut paper designation strips to allow labeling the lines of the Director Place the paper strip on the director and then cover wi
70. Director must be within 2300 feet 700 meters of the Talkshow System The power supply must be 250 feet 76 Meters or less from the Desktop Director Multiple Desktop Directors on a Single Port Two Desktop Directors can be connected to a single port on the Talkshow System You will need to use an external power supply and modular splitter Telos Desktop Director Power Supply Kit part 2091 00001 splitter included You might choose this method to save on cable runs back to a Talkshow System located some distance away Your set up will look like this 120 feet 1100 feet 33 feet 10 meters Max This Director must have its termination resistor removed To connect two Desktop Directors to one port a power supply and RJ 45 modular splitter is required INSTALLATION 21 One of the Directors must have its termination removed This unterminated Director must be within 33 feet 10 meters of the power supply Desktop Director Termination Removal Desktop Directors are shipped with the internal termination resistors enabled When two Directors are used in a single port the termination on the Director closer to the power supply will need to be disabled To do so follow these steps 1 Disconnect the Desktop Director from the Talkshow System 2 Remove the handset and turn over the Director so you can see the bottom of the unit 3 Remove the round black cover This can be pried off with a knife blade Or a miniature scr
71. ISDN circuits Full metering for setup and operation Adjustable Caller Ducking override Conference linking lets you set up high quality conferencing between callers with no external equipment needed Caller ID support on both ISDN and POTS lines 2 Section 1 Nx Talkshow Systems can serve two studios each with two hybrids In the two studio modes lines can either be shared or exclusive to each studio In the latter case there is no possibility for accidental interaction Support for up to eight Desktop Directors Compatibility with Telos Assistant Producer and third party call control software applications Why ISDN Makes Sense for Broadcast Talk Show Systems Nx Talkshow Systems work with either traditional analog Telco lines or with ISDN Integrated Services Digital Network We recommend ISDN service for telephone connections whenever possible Contrary to the perceptions of some ISDN is widely available cost effective and offers many advantages for studio on air telephone systems It is yet another example of digital technology enhancing broadcast operations There is no comparable technology for getting audio to and from the public telephone network Even compared to VoIP ISDN is superior with its lower delay and higher Quality of Service To assist you to have a smooth trouble free installation we provide plenty of advice right here in this manual and live support by people with years of ISDN exp
72. ISDN which predated the NI 1 standard and some switches have not been upgraded to the new format There is also a newer NI 2 standard but it is designed to be compatible with NI 1 for all of the basic functions so this is also acceptable SPIDs Service Profile Identification SPID numbers are required in all but one of the AT amp T proto cols These numbers are given to the user by the phone company and must be entered correctly into the Talkshow System in order for the connection to function SPIDs usually consist of the phone number plus a few prefix or suffix digits There is frequent confusion between phone numbers and SPIDs even among Telco personnel While the SPID frequently includes the corresponding phone number this is not necessarily the case The intention of the SPID is to allow the Telco equipment to automatically adapt to various user requirements by sensing different SPIDs from each type or configuration of user terminal None of this matters with our application but we must enter the SPIDs nevertheless Most of the telephone companies have agreed to implement a standardized SPID for new installations The standard is area code phone number 0101 XXXYYYZZZZ0101 Your Telco service representative should give you one SPID for each B channel you order Don t let the phone company installer leave without providing you with the phone numbers of your B channels called Directory Numbers or DNs and your SPIDs Directory N
73. Mask and Gateway Address See The System Menu below for details But if you will just be doing configuration with a cross cable or isolated switch hub you can keep the default IP number You can also connect first with the default then change the IP values using the Web then reconnect with the new address After you have a network connection between your PC and the Talkshow System open a Web browser and enter http 192 168 0 230 substitute your actual IP number if you have changed from the default into the address entry field You will then see the Talkshow System s main Web screen Telos Nx12 Talk Show System Current state of the system network interfaces Desktop Directors ISDN SPIDs DNs Protocol type POTS Country selection Hookflash duration Auto drop enable and threshold Loop current checking Caller ID configuration Input Output levels and Meter ballistics AES output sync Hybrid Caller ducking EQ AGC Gate Mix Minus Routing Feedback Reduction Configure shows lines and audio connections with help from interactive diagrams view and edit entries in the phone book Configure Livewire Sources and Destinations Names Channel numbers Modes Configure Livewire general purpose 1 0 names channel numbers format Livewire clock mastership settings 802 1p priority and VLAN tagging DSCP Class of Service IP network address Software update and maintenance Backup and restore system settings Syslog messages Sta
74. Mute Button for more information TECHNICAL DATA amp TROUBLESHOOTING 79 Technical Data amp Troubleshooting 6 1 Overview A consequence of modern surface mount construction is that it is frequently no longer possible for local repairs to be made Special and expensive equipment is required to change parts As well today s equipment is very complicated and requires repair technicians to have detailed experience and training and have access to high end test equipment At the same time the advent of overnight delivery services means that equipment can be returned to the factory for quick turn around repair Therefore we do not expect you to fix this unit at the component level and we do not include schematics of the unit in the manual Upon request our support staff can fax or mail you the needed schematic Please see the first pages of this manual for proper procedures on returning units for repair There is the possibility however that you could repair the unit by swapping subsystems such as plug in modules or power supplies The Telos customer support crew is standing by to assist with this if you need it 6 2 System Software amp Firmware Viewing Version Information To assist with troubleshooting Telos customer support may ask you to provide some software version numbers There are two major software components to the system System software This is the software that can be updated by downloading new software
75. NCE 59 One Show default 2 Show Custom 2 Show Split 2 Show Shared 4 Column Audio Loop LCD Only This is for testing purposes Choices are Off This is the normal operating mode No loopback path is turned on Studio Input is sent as send audio and caller audio is presented at Studio Out Studio This loops the AES EBU input and analog signal after Analog to Digital conversion back to the AES EBU output and analog output via the Digital to Analog converter This can be used to proof the audio input and output sections No EC This is not a loop mode but a test mode It disables the echo canceller on the phone line so that measurements such as SNR and THD can be made from the Telco line to the output Factory Defaults LCD Only Selecting this option displays the Restore Defaults sub menu with the following options Return This takes you back to the System menu Restore Defaults This menu has two options No Goes back to the Restore Defaults sub menu Yes Erases all user configuration and sets all to the original factory defaults Reboot LCD Only Selecting this item displays the Restore Defaults sub menu with the following options Return This takes you back to the System menu Reboot System This menu has two options No Goes back to the Reboot System sub menu Yes Causes the system to reboot All callers will be lost This command is inte
76. NT1 sync light for a given ISDN BRI circuit occasionally cycle to the fast flashing state then the line or NT1 is marginal If the problem occurs on 2 BRI circuits simultaneously and they are both in the same interface slot then it is likely the problem is with that ISDN interface card Swapping cards between two of the interface slots will confirm or disconfirm this Is the problem confined to incoming call only In this case check the ISDN Telco protocol setting in the TEL If you believe your line is DMS Custom try the National ISDN setting or vice versa Are the DNs entered correctly If so try using 10 digits rather than 7 digits or vice versa What is consistent about the problem look for patterns Keep watching for a pattern Enlist the help of the users We ve seen apparently random prob lems that we eventually discovered only occurred when it rained or on a certain day each week or at a certain time of day or only when another line was in use In other cases it was related to temperature or dirty AC power Assume nothing and suspect everything Leave no stone unturned while searching for the answer Any other ISDN troubleshooting tips An important tool for troubleshooting incompatibility programming problems is the Talkshow System s integrated ISDN analyzer This enables you to look at the transactions between the Telco line and the Talkshow System so that a judgment can be made as to whether the problem lies with
77. Net problem redial Temporary Failure This cause indicates that the network is not functioning correctly and that the condition is not likely to last a long period of time e g the user may wish to try another call attempt almost immediately May also indicate a data link layer malfunction locally or at the remote network interface or that a call was cleared due to protocol error s at the remote network interface Cause No 42 Net busy redial Switching Equipment Congestion This cause indicates that the switching equipment generating this cause is experiencing a period of high traffic Cause No 43 Access user information discarded This cause indicates that the network unable to deliver user information i e user to user information low level compatibility or sub address etc to the remote users as requested Cause No 44 No channel available Requested circuit channel not available This cause is returned when the circuit or channel indicated by the requesting entity cannot be provided by the other side of the interface Cause No 46 Precedence call blocked Q 850 only This cause is used to indicate that there are no pre emptable circuits or that the called user is busy with a call of equal or higher pre emptable level Cause No 47 Resource unavailable New Destination This cause is used to report a resource unavailable event only when no other cause in the resource unavailable class applies or This ca
78. Nx6 Nx12 To Telephone Company Attention From Company Address City State ZIP Phone Contact Location for line Company Address City State ZIP Phone Contact Date needed Termination date Long distance carrier Number of BRI circuits required We request ISDN BRI circuits for use with the Telos Nx6 Nx12 digital telephone system This device interfaces studio audio equipment to ISDN It requires Circuit Switched Voice CSV We may also use this line for other equipment that makes calls using Circuit Switched Data CSD service Please advise us if there is a cost penalty for having both CSD and CSV If you use IOC ISDN Order Code Capability packages please use Capability Package S If you do not use IOCs use the information on the pages that follow The Nx6 Nx12 has an integral BRI terminal adapter which supports these protocols AT amp T 5ESS Custom Point to Point 5E4 2 or later National ISDN 1 or 2 Northern Telecom DMS 100 Functional PVC1 National ISDN 1 or 2 PVC2 Siemens EWSD National ISDN 1 or 2 AT amp T Point to Multipoint Custom is not supported by the Telos Nx6 Nx12 Please let us know which protocol you will provide and the switch We will provide the NT1 and need a U interface with 2B1Q line coding on a standard six pin 4 conductor RJ11 style modular jack USOC SJA48 of which only the center two conductors will be used Hunt Groups please give us the option specified below
79. Seize Outgoing call in process Request for call has been sent to Telco switch Or outgoing POTS call is pending system if off hook and waiting for loop current Seized Outgoing call in process Telco switch has acknowledged the call request Or Loop current has been detected on a POTS line after going off hook Call Proceeding Outgoing call in process Telco switch waiting for routing info from network Outgoing Ring Outgoing call in process far end has generated a call alert signal Disconnecting Request to disconnect has been made Awaiting acknowledgment Or the POTS interface has gone on hook and system is waiting for loop current to cease Incoming Ring An incoming call request has been detected and is being processed DETAILED CONFIGURATION amp REFERENCE 43 Accepting Call Talkshow System has accepted an incoming call and is waiting for the Telco switch to connect it Or incoming POTS call is pending system is off hook waiting for loop current Connected All signaling is complete the ISDN voice channel is available for use Or a POTS line is off hook and loop current has been detected Busy Line has been busied out after pressing the Busy All button Unable to accept inbound calls Desktop Director Status This screen shows the status each Desktop Director CH represents the channel number As it is possible to connect two Desktop Directors on the same port there are two channels per
80. System connections Even though you are sending a stream of data the phone network knows to translate this back to analog at the far end due to the call setup message sent on the D channel when the call is made If this ISDN BRI circuit will also be used with an audio codec such as the Zephyr or Zephyr Xstream then CSD should be ordered as it is required for audio codec calls CSV is for interworking with voice phone service and allows ISDN to interwork with analog phone lines and phones You will be ordering one or more ISDN BRI 2B D circuits with CSV required and CSD optional on each B channel Euro ISDN normally provides both CSV and CSD which is acceptable NT1s The ISDN standard specifies two reference points the U and the S interfaces The U is the single pair bare copper from the Telco CO A device called a Network Termination Type 1 converts this to the two pair S interface In Europe Asia South America and most of the world the NT1 is always pro vided by the phone company and only the S interface may be on user equipment If you plan on taking your TWOx12 to Canada or North America you will need to rent or purchase an external NT1 for use while in those countries One rental source is Silver Lake See their website at www silverlakeaudio com Hunt Groups Just as with analog lines ISDN BRI lines can be placed in a hunt group where a single number can be given to callers and the calls will ro
81. TELOS Nx6 Nx12 Advanced Digital Broadcast Telephone System USER S MANUAL Manual Version 22 August 2012 for software version 1 6 7 and later TELOS Nx6 Nx12 Talkshow Systems 2010 2012 TLS Corporation Published by Telos Systems a member of the Telos Alliance All rights reserved Trademarks Telos Systems the Telos logo and Nx6 Nx12 are trademarks of TLS Corporation All other trademarks are the property of their respective holders Notice All versions claims of compatibility trademarks etc of hardware and software products not made by Telos mentioned in this manual or accompanying material are informational only Telos Systems 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 which may affect the product specifications or to revise the manual without notice Warranty This product is covered by a two years limited warranty the full text of which is included in this manual Updates The operation of the Nx6 and Nx12 is determined largely by software We routinely release new versions to add features and fix bugs Check the Telos web site for the latest We encourage you to sign up for the email notification service offered on the site Feedback We welcome feedback on any aspect of the Telos Nx Talkshow Systems or this manual In the past many good ideas fr
82. able 5 10 Firmware Update Talkshow System firmware may be updated via the Web interface Open your browser connect to the Talkshow System and click on the System link Firmware update is at the bottom of the System page The first step is to get the new firmware onto your PC This will usually be via a download from the Telos website or an email attachment from Telos customer support The file will have the extension pkg When the firmware is available on your PC click the Browse button to locate it and then click Apply to upload it to the Talkshow System The browser will disconnect for ap proximately 25 seconds and then automatically re connect There will be a progress status screen and then the System page will return The new software will be in Talkshow System s Bank 1 Firmware version Hardware revision Telos Nx12 rev 1 Obank 0 ver 0 6 4 build Fri Jul 27 17 45 30 EDT 2007 Bank 1 ver D 7 1rc build Mon Aug 13 12 45 36 EDT 2007 commit this version to Bank O Warning System will reboot after changing current bank You can make it active by clicking the Bank 1 radio button and then Apply 72 Section 5 You can keep two firmware versions and choose between them at any time But when you eventually want to upload a yet newer version you will need to commit the firmware in Bank 1 to Bank 0 which clears Bank 1 to prepare it for the new upload Firmware version Hardware revision Telos Nx12 rev
83. ach line button in the left column and listen for dial tone You should see the icon as each line is selected Select line 1 and enter the phone number for line 2 using the numeric keypad You should now hear a ringing sound and see the ringing Status Symbol an expanding square on line 2 Press the right column line 2 button to answer this call You should see the icon Drop this test call by pressing the right Drop button Having completed that first call you know your lines 1 and 2 are working If you are using ISDN you have now tested the first BRI Now repeat this test by calling from line 2 to line 3 line 3 to line 4 etc until all lines have been tested Press the Drop button after the final call Tip Many phone lines have per minute usage charges Don t forget to drop the call once your testing and configuration are complete 3 3 Audio Input and Output The Talkshow System can take audio input from analog AES3 if the option is installed or Livewire sources When the AES module is installed the XLR connectors are used for AES and analog connections are not possible When Livewire inputs are activated the analog AES inputs are switched off However should LW inputs fail analog AES will automatically be switched on On the output side both analog AES and Livewire can be simultaneously active Levels Levels Levels Ok we assume that you are able to make a call to yourself And that you have
84. ada Integrated NT1 for direct connection to ISDN line via the 2 wire U interface 6 position 2 pin RJ 11 connector 2B1Q line coding Worldwide 4 wire S interface 8 position 8 pin RJ 45 connector Telephone Coding Modes Law ISDN Proto set to Natl 1 1 AT amp T Custom Q 931mu or DMS Custom A Law ISDN Proto set to ET S 300 Analog Telephone Line Connectivity Universal interface for worldwide application Programmable loop current ring signaling and flash time Includes caller ID decoding using Bellcore 212 modem standard Desktop Director Ports Four ports permitting connection of 4 Telos Desktop Directors Eight directors using external RJ 45 splitters and power Extended Desktop Directors count as 2 for power purposes Control Ports Ethernet 100Base T Web server for configuration and software updates Telnet for command line control and diagnostics Assistant Producer server allows up to 8 instances of Telos Assistant Producer version 3 5 or later to connect simultaneously General purpose Input Output 15 pin D Sub connector with 5 status outputs and 4 control inputs Two Studio Modes The Talkshow System can be used with two studios independently Lines may be independent or shared Each studio has two hybrids There are two Progam on Hold inputs one for each studio Two sets of Telos Assistant Producer application can be used independently Note the following With analog audio in out there ar
85. ally literate Called Party Address This is the destination phone number of a call delivered to a switch For instance this could be the CLID of a call delivered to a PBX using DID or two way trunks See also DID Caller ID A CLASS feature on an analog line that provides the number of the calling line as a burst of FSK data modem tones following the first ring Also called Calling Line Identification See also CLASS Calling Line ID See CLID See also Caller ID Calling Party Control See CPC CAS Channel Associated Signaling A bit based signaling method used on digital lines such as T1 that is periodically inserted into the low order bit also used for the audio transmission See Robbed Bit Signaling Cause Code A code returned by switching equipment to ISDN equipment as part of the call control signaling protocol The cause code indicates that a network call related event has occurred or has failed Since these codes actually come from the network the fact you are getting a cause code is an indication that that the ISDN circuit is operational at some level CCIS Common Channel Interoffice Signaling A signaling system where network information such as address and routing information are handled externally to the actual communications voice path SS7 Signaling System 7 is the internationally standardized CCIS system Deployment of CCIS increased efficiency since no com munications voice channels are used merely to rep
86. andles Interlata and interstate calls Also referred to as IXC 2 Interna tional Electrotechnical Committee A European standards body best known for the power plug now used throughout the world for AC power cords for use on office equipment and computers ILEC Incumbent Local Exchange Carrier A local Exchange Carrier which entered the marketplace before the enactment of the 1996 Telecom act i e a telephone company which is neither an Indi nor an RBOC See LEC and CLEC IMUX See Inverse Multiplexing In Band Signaling A signaling system where network information such as address and routing information are handled over the com munications voice path itself Usually the information is represent ed in the form of audible tones however DC loop current signaling also qualifies as In Band Signaling See also CCIS Incumbent Local Exchange Carrier See ILEC See also CLEC amp LEC Independent Any of the phone companies in existence at the time of divestiture that were not affiliated with the Bell System See RBOC LEC and CLEC Indi See Independent Interconnect Company A vendor of telecommunications CPE other than a BOC or AT amp T This term was originated by AT amp T and was meant to be derisive towards the fledgling industry when the courts said it was OK for end users to buy equipment from someone other than the Bell System This industry flourished in spite of AT amp T s disdain and ironically the RBOCs were n
87. ay use a line with both capabilities Just be sure the BRI circuit supports the Circuit Switched Voice CSV capability as well as data TIP If this line may be borrowed at times for use with an audio codec such as the Telos Zephyr you need to be sure that the Circuit Switched Data CSD capability is enabled Hunt Groups The most common configuration for on air phone system is to have the different phone num bers linked in a Hunt Group also called Rollover Lines or Incoming Service Grouping ISG A hunt group allows you to give out a single number to the audience and each call will hunt to an unused line Sometimes one or two numbers will be reserved for a hot line or warm line in which case those numbers would not be part of the hunt group 8 Section 2 In some cases ISDN lines configured to hunt may deliver all calls to a single DN phone number In this case you must enter this same DN for each of the DNs on each of the hunting channels Special Considerations for Euro ISDN ETS 300 users Some Euro ISDN implementations deactivate BRI circuits that have been idle for more than a given length of time Since this is a normal occurrence the Desktop Director and system front panel will not show these lines out of service when this happens While this will not interfere with normal operation it can give a false sense of security that everything is working when in fact the circui
88. base lookup is how a call to a mobile telephone user can handled transparently despite the fact that the user s location may change Also used to determine what carrier should handle a given toll free call See also CCIS Station Line A telephone circuit from a PBX to a telephone on that PBX Since this is a telephone to switch connection it is considered to be a line See Line and Trunk Station Side The user side of a PBX The side of the switch that the telephones are attached Also occasionally called the line side The main reason for distinguishing between this and the trunk side is that certain customer related features Such as Hold and Transfer are inapplicable to trunks See also Trunk Side Statistical Multiplexing A method of improving effective band width of a Telco channel Data is divided into packets which is sent on the channel when there is space available This is the key to the Internet Subscriber The customer of a Telecommunications company This term dates back to when a local Telephone Company was formed at the specific request of a group of customers who agreed in advance to subscribe to the service Superframe See SF Switch Telephone switching device which makes the connection when you place a call Modern switches are special ized computers ISDN service is provided from a Digital switch most commonly in the USA amp Canada an AT amp T model SESS Northern Tele
89. ber either alone or as part of a hunt group In the case of a normal i e analog DID Trunk a group of phone numbers are associated with that DID trunk or group of trunks and incoming calls include the DID number so the PBX can route that call to the correct DID extension These are one way i e inward only trunks This is exactly how ISDN PRI functions with the DID information coming in over the D channel There is a big difference between a normal DID Trunk and a Two way DID trunk over ISDN PRI For one thing ISDN PRI is digital Another distinction is that you cannot dial out over a true DID trunk while you can dial out over GUIDE TO TELEPHONE TERMINOLOGY 115 a PRI hence the conflicting designation Two Way Direct Inward Dialing Trunk Two Wire A circuit path where only a single pair of wires is used A hybrid is used to convert from two wire to four wire circuits No hybrid is perfect and those used by the phone company can be poor However the hybrids in Telos TWO 2101 family are approaching perfection U Interface The interface between the ISDN BRI line and the NT1 ANSII standards for the US call for 2B1Q line coding which is also used in some other places though not neces sarily This can be considered the ISDN phone jack in the USA amp Canada and is frequently in the form of a RJ 11 or RJ 45 style telephone jack The center 2 pins are used on these jacks V 35 A serial data i
90. bles the Phone Company line setup the punch blocks etc At the component level many computer problems stem from the same sources Never never never forget to check Vcc Step 4 Generate a hypothesis Before changing things formulate a hypothesis about the cause of the problem You probably don t have the information to do this without gathering more data Sometimes you will have no clue what the problem might be Sometimes when the pangs of desperation set in it s worthwhile to try anything practically at random You might find a bad plug an unconnected line or something unexpected Look around but be always on the prowl for a working hypothesis Step 5 Generate an experiment to test the hypothesis Change the ISDN connection to a known good line call known good phone or hybrid at the other end iflong distance doesn t work try a local call You should plan your tests to eliminate 50 of the possible problems in one test if possible Just keep careful track so you know what you have eliminated Step 6 Fix the problem A Final Thought Constantly apply sanity checks More than three decades ago the Firesign Theater put out an TECHNICAL DATA amp TROUBLESHOOTING 81 album called Everything You Know is Wrong Use that as your guiding philosophy in trouble shooting a Talkshow System set up For example just because you checked the Telco line last night and it was fine does not mean that it s OK now At 3 00 AM wh
91. cept through an external mix minus Thus Separate requires the mixing console to provide the cross connection for conferencing when required With One Show show mode and in any of the Two Show show modes there are two hybrids per show so using the Separate mix minus mode means you have two caller outputs to each console and two mix minus caller inputs from each console One Show or 1st of two Shows Input 1 Hybrid A Output 1 Input 2 Hybrid B Output 2 Second of two Shows Input 3 Hybrid A Output 3 Input 4 Hybrid B Output 4 In 4 Column show mode togther with Separate mix minus mode you will have four independent caller outputs to four faders on the console and you will need four mixminus feeds 1in multiple out Using this option means each mixing console only has to make a single mix minus feed for the caller send audio for each show but separate hybrid caller audio outputs are sent to the console This mode is useful when your console is limited in the number of mix minus feeds it can make but you want to keep individual faders per hybrid 50 Section 5 This option sends the single mix minus feed from the console to each caller Internally within the Talkshow System the cross connected audio from the other callers is mixed in so each caller can hear the other callers With One Show show mode and in any of the Two Show show modes there are two hybrids per show so using this mix minus mode means you have two caller
92. ch exchange implies a PBX that supports dialing by end users Many PBX s can now offer ISDN BRI service usually over the S Interface A few vendors are now offering BRI over the U interface as well PRI over DSX 1 T1 or El is also offered in some cases Be wary of these ISDN protocols since they have not been as well tested as the versions running on public switches They may or may not work with a given piece of CPE PIC Primary Interexchange Carrier USA This is your default 1 carrier used for interLATA calls In some areas you may have two PICs one for interLATA calls and one for intraLATA long distance calls in which case it stands for Primary Intraexchange Carrier In some areas intraLATA long distance calls are still handled by your RBOC in others you now have a choice You may be able to discover who you current PIC is by dialing 700 555 4141 PMP USA AT amp T Custom Point to Multi Point Custom ISDN Protocol Not supported by the Zephyr or ZephyrExpress See Custom ISDN and ISDN Protocol POP Point Of Presence The local facility where your IEC main tains a switch This is where your long distance calls get routed so that your IEC can handle them Also used to describe the local access point of an Internet Service Provider POT Plain Old Telephone A black rotary dial desk phone Usually a Western Electric model 500 set Outdated term POTS Plain Old Telephone Service Regular old fashioned anal
93. ch sending this cause is clearing the call because a threshold has been exceeded for multiple protocol errors during an active call Cause No 102 Timeout disconnect Recovery on timer expiration This cause indicates that a procedure has been initiated by the expiry of a timer in association with error handling procedures Cause No 103 Parameter non existent or not implemented passed on National use Mandatory Information Element of incorrect length VN 4 only This cause indicates that the equipment sending this cause has received a message which includes parameter s not recognized because the parameter s are not defined of are defined but not implemented by the equipment sending this cause This cause indicates that the parameter s were ignored In addition if the equipment sending this cause is an intermediate point then this cause indicates that the parameter s were passed unchanged or This cause indicates that the equipment sending this cause has received a message with a mandatory information element of incorrect length Cause No 110 Message with unrecognized parameter discarded Q 850 only This cause indicates that the equipment sending this cause has discarded a received message that includes a parameter that is not recognized Cause No 111 Protocol Error unspecified This cause is used to report a protocol error event only when no other cause in this class applies Among other things this cause may be
94. change A telephone exchange which is assigned to Radio and TV stations Promoters and other users that will be receiving large numbers of simultaneous calls The idea is to group all of these users on a single exchange so when all routes into that exchange are in use normal users on other exchanges will not ex perience blocking of incoming or outgoing calls Trunks from other local exchanges into the choke exchange are deliberately limited to just a few paths so callers will get an all trunks busy instead of completely blocking their local exchange However when one of the choke exchange users experiences a large number of calls as when your station runs a contest the other choke exchange users will be blocked because all trunks into the choke exchange will be busy In the modern network using CCIS signaling such as SS7 actual trunks are not used to convey busy or all trunks busy conditions Thus blocking due to a station contest should not occur as the busy status in response to a call attempt is conveyed over the separate SS7 network Therefore the need for choke exchanges has pretty much disappeared Nonetheless many Telcos still insist that Broad casters use special choke lines for call in lines See blocking and concentration Circuit A physical path through which electrical signals can pass It consists of a network of conductors and other components sepa rated by insulators Technically this term
95. com DMS100 Functional Custom PVC1 Northern Telecom DMS100 switches BCS 31 and above CO Values Line Type Basic Rate Custom Functional EKTS No Call Appearance Handling No Non Initializing Terminal No Circuit Switched Service Yes Packet Switched Service No TEI Dynamic Bearer Service CSD CSV on both channels CSV is required CSD may be omitted if requested by user 10XXXXX Long Distance Access Yes 122 Appendix 3 Give us Two SPID numbers depending upon number of active B channels Two Directory Numbers PROTOCOL AT amp T Point to Point Custom Software version 5E4 2 and above CO Values Line Type DSL class Custom Point to Point PTP B1 Service On Demand DMD B2 Service On Demand DMD Maximum B Channels MaxChan 2 CSV Channels Any Number of CSV calls 2 CSD Channels Any CSV is required CSD may be omitted if requested by user Number of CSD calls 2 Terminal Type A 4 4 4 Number Display No Call Appearance Pref Idle 10XXXXX Long Distance Prefix Yes Turn off features such as packet mode data multiple call appearances Electronic Key Telephone Sets EKTS shared directory numbers accept special type of number intercom groups network resource selector modem pools message waiting hunting interLata competition etc Note for AT amp T Lucent 5ESS on loop extended lines REX test must be disabled for this line
96. com model DMS 100 or Siemens model EWSD The switch and related software running on it will determine which ISDN protocol s will be available to customers connected to it See also PBX Switched Circuit A channel which is not permanent in nature but is connected through a switching device of some kind The switch ing device allows a switched circuit to access many other switched circuits the usual dial up type of telephone channels Once the connection is made however the complete capacity of the channel is available for use As opposed to a dedicated circuit or a packet based connection Synchronous Data A form of serial data which uses a clock signal to synchronize the bit stream Since unlike asynchronous data no start and stop bits are used data throughput is higher than with asynchronous data ISDN and T 1 use Synchronous data See also Asynchronous Data T Link A proprietary Telos 2 048mbps channelized link This link uses the DSX 1 electrical protocol and has 30 channels at 64kbps each T1 A common type of digital telephone carrier widely deployed within the US Canada and Japan Has 24 64Kbps channels called DS0 s The most common framing scheme for T1 robs bits for signaling leaving 56kbps per channel available for data use T 1 An alternative expression for T1 See T1 TA Terminal Adaptor The electronic interface between an ISDN device and the NT1 The terminal adapter handles the
97. compatible Ethernet switch is present should Livewire streaming be enabled 68 Section 5 When Livewire is enabled the corresponding Talkshow System Analog or AES3 inputs are switched off However should a Livewire input fail the Talkshow System automatically switches the input to Analog AES providing a back up On the other hand the Talkshow System analog AES outputs will be simultaneously active when Livewire is enabled If you need to specify the source and destination Livewire channels independently click the Expert Mode link to open a new page with these options Livewire GPIO The parallel GPIO hardware port provides logic inputs and outputs from the Nx Talkshow system The various functions of the logic are described in section 2 10 Those familiar with the Axia product line will recognize the GPIO section of the web interface It is important to understand that the GPIO port within the Nx Talkshow system is a virtual port and the Parallel port functions have been duplicated to the LiveWire GPIO ports This can be used to connect to other remote logic devices via an Axia GPIO node or PowerStation Element GPIO port when configured as a PORT to PORT connection GPIO Configuration Port 1 Inputs BRS Outputs Pe Port 2 Inputs BRE Outputs Pe Name Channel iT i 2 Cf Apply Owners of Axia Consoles integrating the Nx Talkshow systems will discover that the GPIO Logic definition for a PHONE source doe
98. configured your audio input output settings Congratulations you just have some tweaking to do and your basic installation and configuration will be complete BASIC CONFIGURATION 29 Bring up some audio on the mixing console Observe the two Input meters Confirm that audio is present and that levels are OK Section 5 1 has more information on the Talkshow System s metering Tip If the levels are too high hybrid performance will be significantly diminished and leakage may occur First set the Nominal In Level in the Audio menu to a nominal level close to that used in your facility i e 10 0 4 or the Euro PPM 9dBu or 9dBfs settings This control sets the level for analog AES and Livewire inputs Section 5 4 has more You should see the audio from the phone line on the Output LED bargraphs Ifyou have two mix minuses properly set up you should see the audio that is output from hybrid 1 on hybrid 2 s input and vice versa Bring up the fader for Phone 1 on your console Adjust the Talkshow Systems Nominal Out Level in the Audio menu This sets the level for all outputs Check that all are providing correct levels to your mixing console Time to test it out Ok you made it Another hour and you can go home for the day First we will check that the mix minus for each hybrid is correct Then we will make a test call Confirm that the Desktop Director you will be using is working and says Talent at the top
99. d type like this to indicate what you would type and Courier type like this to show what the Talkshow System replies First you must log into the Talkshow System gt gt Login Telos lt return gt Nx6 V0 6 3b 2007 07 19 15 58 Logged in with user rights You should now be able to get the most up to date listing of commands by typing help or P gt gt help lt return gt Type help lt command gt to get specific help about a command Type help to get a short list of commands Type help to get a detailed list of commands Type help lt chars gt to get a list of commands starting with lt chars gt starts a decimal number a binary number S a hex number Current number base is 16 Separate multiple commands on a line by List of available commands CPC acc autoans bootstat bye call cidalert csinfo daa daacheck dd disc 74 Section 5 dmr dmw droponcptd fpkey gpio help hookflash hybrid isdn isdnlog lawmode lcgood lcignore lcinfo lcoffset lcwait lines log login logon lw mcheck meter minonhook mixpoll mr mw potscountry prio quit reboot ringtimo test ti tslot For a description of a command type help or followed by the name of a command Example gt gt CPC lt return gt Command CPC lt time in msec gt Description Specify the time in msec a loop current drop must last before a far end drop is recognized When set to 0 then l
100. dified AMI is also used on the ISDN S interface See line coding T1 See Also B8ZS ANI Automatic Number Identification A system originally de signed for use by Interexchange carriers ECs that transmits the billed party number along with a call Note that the billed party number is not necessarily the number of the line placing the call ANI predates SS7 and can operate with analog as well as digital trunks See also CLID and Caller ID Asynchronous Data A form of serial data communication that is not clocked To keep the bit stream synchronized start and stop bits are added which cuts down on throughput RS 232 computer data is commonly asynchronous data In contrast to synchronous data Acoustic Echo Canceller AEC When an acoustic path exists from a microphone to a loudspeaker in a teleconferencing applica tion the far end participant will hear his voice coming back after a delay This echo is annoying and can disrupt conversation An AEC is a signal processing function that works to remove this echo B Channel Bearer Channel One of the multiple user channels on an ISDN circuit Used to carry user s data i e coded audio data in the case of Zephyr or ZephyrExpress B8ZS Bipolar 8 with Zero Substitution A T1 line coding method This is the more modern line coding method of the two commonly available See Line Coding T1 See also AMI Bearer Channel See B Channel Behind the PBX This is our own term and
101. displayed if you failed to dial a 9 or 8 for an outside line if required Also may be returned if you have some types of restrictions as to the number of calls etc Cause No 113 Bearer service unavailable VN 4 only This cause indicates that the bearer service requested is not available or is unimplemented Cause No 114 End to End Information transfer impossible VN 4 only This cause indicates that the requested network does not support the user end to end transfer capability CLASS 1 7 interworking class Cause No 126 Switch to conversation mode VN 4 only This cause is used to indicate that conversation mode has begun TECHNICAL DATA amp TROUBLESHOOTING 93 Cause No 127 SW56 disconnect Internetworking unspecified This cause indicates that an interworking call usually a call to SW56 service has ended May also be seen in the case of a non specific rejection by your long distance carrier try again at a different rate SPECIFICATIONS 95 Specifications General Interface for connecting three ISDN BRI S0 or six analog telephone lines to professional studio equipment Includes four high performance digital hybrids and a digital matrix switch All hybrids can be dedicated to a single studio for excellent conferencing results or the Talk show System may be used to serve two studios with two hybrids each Audio interfaces included as standard are analog and Livewire Audio over IP
102. e also Trunk Side 2 The user side of a PBX Also called the station side Line Termination See LT GUIDE TO TELEPHONE TERMINOLOGY 111 Local Access and Transport Area See LATA Local Exchange Carrier See LEC and CLEC Long Distance If your local Telco is a former Bell Operating Company then any call outside of your LATA or any Interstate call is considered long distance and is handled by an IEC The above is true regardless of whether you are referring to a dedicated line or a dial up call 2 However under the current state of deregulation toll calls within a LATA may now be covered by the IEC and in some cases RBOCs are being permitted to handle InterLATA calls Loop The telephone circuit from the CO to the customers prem ises Generally refers to a copper cable circuit Loop Current Disconnect Supervision Another name for CPC See CPC Loop Qualification Process of actually measuring the loss on a prospective ISDN line to see if it can be used for ISDN service The actual loss on the line usually measured at 40 kHz is the determin ing factor whether ISDN service can be offered without a repeater Generally ISDN is available up to 18 000 feet from the serving Central Office It may not be available within this range or may be available further from the CO Only a loop qualification can tell for sure Not all Telcos will extend ISDN lines with repeaters Loop Start Line A plain old telephone line The telephone
103. e been adjusted to the requirements of your installation then the Nx6 is ready for next steps You may confirm that ISDN lines are ready with the ISDN status screens Press any menu button twice and use the up and down arrows to scroll through these screens If all is OK you will see the Ready indication for each ISDN B channel If you do not see READY as the ISDN status first cycle power to the Talkshow System off and then back on If you still do not seen READY on the ISDN status screen review the informa tion in Sections 2 4 2 5 and 3 2 and check both your ISDN settings and set up information If the information there does not help there is in depth troubleshooting information in Section 6 Tip Users in the USA amp Canada should see READY status whenever the system is idle On certain ETS 300 ISDN lines the system will go from READY to DEACTIVATED This is no cause for alarm 5 1 Level Metering The Talkshow System includes four LED bargraph meters The ballistic time characteristic of these meters changes depending on the Audio menu Input Level setting to match the expected console meter type In VU mode they have the usual 300ms time constant In PPM mode they have a 10ms attack and 2 8sec release characteristic The meters are selectable to various points in the system with Meter Mode in the Audio menu Note that the marked meter names are correct only when the Studio 1 amp 2 mode is selected See Section 5 4 below
104. e number dialed DP Dial Pulse A method off sending address information by either 1 Causing brief interruptions in loop current or 2 Caus ing brief changes of state of a bit on a digital circuit using Channel Associated Signaling In other words rotary or pulse dialing See also DTMF and CAS DS Digital Signal Level Zero The smallest unit of measure of the standard rate hierarchy used by the Telcos i e all other rates are a multiple of the DS rate For example the T1 rate is 24 times the DS rate and the El rate is 32 times the DS rate 64 kbps See also B channel DS1 Digital Signal Level 1 The second level up the digital rate hierarchy used by the Telcos This is 24 times the DS rate for a total of 1 544 mbps See DSW See also T1 DS2 Digital Signal Level 2 Data rate of 6 312 mbps 4 times the DS1 rate See DS and DS1 DS3 Digital Signal Level 3 Data rate of 43 232 mbps 28 times the DS1 rate See DS and DS1 DSL Digital Subscriber Line Traditionally refers to an ISDN circuit or sometimes a T1 line although the term is also frequently used to mean the next generation beyond ISDN Sometimes xDSL is used to indicate that the writer is referring to any of a number of emerging DSL technologies DSU Data Service Unit See CSU DSU DSX 1 Digital Cross Connect level 1 USA amp Canada primar ily Defined as part of the DS1 T1 specification and is a closely related signal The type of s
105. e number of lost or corrupted data packets in the Talkshow System to Desktop Director direction since this Director was powered on Far End the number of lost or corrupted data packets in the Desktop Director to Talkshow System direction since this Director was powered on Version Reports the currently running Talkshow System software version number and the currently running Desktop Director software version number About General information Changing Between Producer and Talent Modes A Desktop Director can be in either Producer or Talent mode This can be changed at any time independently for each Director in the system Producer mode can be thought of as screener phone mode while Talent mode can be thought of as on air mode To change between Pro ducer and Talent modes 1 On the main screen press the soft key button below the word Menu You should now see a screen that looks similar to the following Main menu Change Director Mode Select Show a The words Change Director Mode should be highlighted 2 Now press the Select button You will now see a screen like this Select Director Mode Producer screener Talent Contalro 3 Press V or A to change the setting 4 Press Select to confirm your choice or press the soft key below the word back to return to the main menu without any change taking effect Using the Director in Producer Mode The first text line on the LCD tells you
106. e only two total inputs and outputs So the Talkshow 98 Section 7 System must be used in a mode that combines the two hybrids assigned to each studio into one input output The AES option provides 4 input and 4 output channels total so all hybrids are accessible with each studio using one AES connection s two channels Program on Hold remains an analog input Livewire provides the full 4 hybrid input channels 4 hybrid output channels and 2 Program on Hold input Four Column Four Hybrid Mode The Talkshow System can be used with an extended Desktop Director which has four tele phone line columns Each has an associated hybrid Note the following With analog audio in out there are only two total inputs and outputs So the Talkshow System must be used in a mode that combines the hybrids into two groups of two each The AES option provides 4 input and 4 output channels total so all hybrids are accessible Program on Hold remains an analog input Livewire provides the full 4 hybrid input channels 4 hybrid output channels and a Program on Hold input Euro Operation Mode European broadcasters traditionally prefer to have an independent signal path from each telephone line through an independent hybrid and to a mixing console fader dedicated to each No line selection matrix switch is used The Talkshow System supports this operation style with Axia mixing consoles for up to four lines A standard Desktop D
107. eing connected to a channel already established to that user for similar calls e g packet mode X 25 virtual calls etc TECHNICAL DATA amp TROUBLESHOOTING 87 Cause No 8 Call is proceeding Preemption Call in process please standby or This cause indicates a call has been preempted Cause No 9 Preemption reserved This cause indicates a calls has been preempted due to the circuit being reserved for reuse Cause No 10 Prefix 1 not dialed DMS Custom only This cause indicates that the prefix 1 was not included in the called party address and is re quired Cause No 11 Excessive Digits Received call is proceeding More digits were dialed than expected Called number has been truncated to the expected number Cause No 14 Excess digits received call is proceeding More digits were dialed than expected Called number has been truncated to the expected number Cause No 16 Far end disconnect Normal call clearing This cause indicates that the call is being cleared because one of the users involved in the call has requested that the call be cleared Under normal situations the source of this cause is not the network Cause No 17 Busy try again later User busy This cause is used to indicate that the called party is unable to accept another call because the user busy condition has been encountered This cause value may be generated by the called user or by the network In the case of use
108. em The 100Base T connector has two integrated LED indicators The green Link LED indicates the presence of a live physical connection to a working Ethernet hub or other device If the green indicator does not light you should check your network wiring Absence of the link light could also indicate a hardware failure of the Talkshow System The amber Activity LED indicates network activity on this segment of the network Depending on network activity this indicator may only illuminate occasionally flickering of this light is normal For details on using Web based configuration see Section 5 3 For information on using Telos Assistant Producer and other call screening applications see Section 5 10 INSTALLATION 23 2 10 Parallel GPIO Remote Control This port provides inputs to and outputs from the Talkshow System Inputs can be connected to logic outputs or switch closures Outputs can be connected to external device logic inputs When any of the two show modes are used functions labeled 1 are used for Show 1 and functions labeled 2 are used for Show 2 When show mode is One Show only the functions for 1 are available The available functions are Record Control Outputs See Section 4 3 Ring Mute In This input will mute the ringers on all Desktop Directors connected to the corresponding Show Individual Desktop Directors can be set to ring regardless of this input see Chapter 4 User Out Delay Dump
109. en the problems seem intractable and you are ready to give up engineering remember that the system has worked and will work Never panic you are smarter than it is 6 4 Troubleshooting ISDN Problems Status Screen for ISDN troubleshooting This screen shows the status of the ISDN connection You can determine if the line is connected and whether the line and the Telos Talkshow System are successfully configured In addition information about calls is displayed here One of the following status words will be shown for each active ISDN channel Deactivated No line connected or connected to wrong jack on the Talkshow System OR ET S 300 permits the Telco to put a line to sleep Deactivated will be displayed when this occurs SPID Pending The SPID has been sent to network DMS Cust And Natl 1 1 only Network has not ap proved SPID SPID Error The SPID is incorrect or line is grossly misconfigured Ready The unit sees a live ISDN S interface ie external NT1 is present or internal NT1 has completed startup If the SPIDS are present they have been sent and approved Pending Seize Outgoing call in process Request for call has been sent to Telco switch Seized Outgoing call in process Telco switch has acknowledged the call request Call Proceeding Outgoing call in process Telco switch waiting for routing info from network Outgoing Ring Outgoing call in process far end has generated a call alert signal Disconnecting Request t
110. er You cannot arbitrarily assign any unused channel number and assign that to another Livewire GPIO port and expect the inputs and outputs to work This will not map the two ports together Quality of Service These are identical to the Livewire Node options They should normally be left at the default values Axia Consoles Axia mixing consoles are based on Livewire This networked approach delivers many advantages among them that Livewire connections are bi directional multi channel and include control This means that a single RJ 45 can interface all necessary Talkshow System signals to an Axia console Livewire simplifies mix minus because each channel is two way so return feeds are baked in and not an afterthought Axia is a division of Telos so as you could imagine we take good care of phones and codecs in our mixing consoles Easy to use mix minuses are available on every fader channel Separate DETAILED CONFIGURATION amp REFERENCE 71 on air and off line recording mix minuses are supported Each channel has a dedicated Talkback There is a one button off line record mode And there are dedicated modules for phone control These provide seamless integration since they communicate with the Talkshow System over the same Ethernet that is used for all other connections Both USA style operation with the well known Telos two column line selector approach and Euro style with dedicated faders to each Telco line are avail
111. erience Many thousands of Telos ISDN interfaces and Zephyrs are deployed around the world Currently the dial up telephone network is nearly entirely digital Only the last mile copper connections from the telephone Central Office exchange to the customer site remain with the archaic more than a hundred years old analog scheme ISDN offers us a way to link our studios directly into the Telco digital infrastructure without the usual impairments of analog connections In all digital studio facilities the telephone interface can maintain a digital path all the way using either AES or Livewire networked audio on the studio side While the application of digital signal processing to the problem of separating announcer and caller audio pioneered by Telos and used in all of our telephone interfaces has made a dramatic improvement over analog only hybrids using digital phone lines for on air calls further improves performance for several reasons 1 Exceptional Send to Receive Isolation Traditional analog phone lines multiplex both speech directions in order to use a single pair of wires for a conversation This causes what s referred to as leakage when the announcer s audio is present on the hybrid output creating an annoying hollow or tinny sound Telos digital adaptive hybrids reduce this problem But ISDN offers independent and separated signal paths so our hybrids only have to operate on the far end anal
112. ers typically 10 or 20 some times higher are purchased from the LEC or CLEC for use on the PBX The number of DID numbers usually substantially exceeds the number of trunks in the system DID Trunk A Direct Inward Dialing Trunk A trunk channel which can only receive calls A group of telephone numbers DID numbers are associated with a given trunk group however there is no one to one correspondence between the individual channels and these numbers The PBX uses the DID number given it by the phone company to route the channel to the correct DID extension within the PBX This allows some or all PBX stations to receive calls directly without going through an attendant or auto atten dant Note that there are usually more DID numbers than there are DID trunks See DID number and DID extension Direct Inward Dialing See DID Directory Number USA amp Canada Your seven digit telephone number without the area code as found in the telephone directory Directory 1 amp 2 Zephyr The Utility menu on the Zephyr where the 7 digit Directory Numbers can be entered during set up The Zephyr generally does not require these numbers DN See Directory Number See also Directory 1 amp 2 and MSN DN 1 amp 2 DNIS Dialed Number Identification Service A service typically offered by a long distance company on 800 lines that provides the number dialed by the caller This allows a caller to receive specific treatment depending on th
113. es 2x input and 2x output AES XLRs in place of the standard analog providing 4x in 4x out channels Conform to AES3 standard Sample rates 32kHz to 96kHz Rate conversion Input and output independently selectable Output Clock AES input or 48kHz internal Input Level Adjustable 27 to 9 dBFS nominal Output Level Adjustable 27 to 9 dBFS nominal Livewire Livewire connection is standard for all hybrids in out and POH audio in Audio Performance Frequency Response 0 5dB 20Hz to 20kHz swept sine procedure measured from analog input to output with unit in loopback mode THD N Analog to analog studio loop mode 1Khz 18dBu test level lt 0 006 Dynamic Range Analog in to Analog out studio loop mode 10Hz 20Khz a weighted gt 92 dB SNR Analog output referred to 12dBm phone line signal 4dBu studio out 10Hz 20Khz a weighted gt 72 dB Trans Hybrid Loss Analog or ISDN phone line ducking gate AGC EQ all off relative to 4dBu input level gt 55 dB Switching Matrix and Conferencing All digital audio routing and switching SPECIFICATIONS 97 6 telephone lines 4 hybrids 4 studio inputs 4 studio outputs 2 Program on Hold inputs ISDN Telephone Line Connectivity Protocol Compatibility National ISDN 1 and 2 DMS 100 Custom Functional AT amp T 5ESS Custom Point to Point Euro ISDN conforming to the NET 3 ETS300 protocol Interface one of the following USA amp Can
114. ethod This is the more modern line coding method of the two commonly avail able See Line Coding T1 See also AMI and B8ZS Hunt group A group of telephone channels configured so that if the first is busy engaged the call goes to the next channel if that channel is busy it goes to the next channel etc Hunt groups may hunt from the highest to the lowest the lowest to the highest or on some other arbitrary pattern But the order of hunting will usually be fixed beginning with one channel and working through hunt ing until an unused channel is found The term may have origi nated back in the old manual switchboard days when the operator literally hunted for an unused jack to plug a cord into This arrange ment is very common in business scenarios where a single incoming 110 Appendix 2 number the Listed Directory Number is given to the public but multiple incoming channels are supported See also LDN Hybrid A device which converts from a two wire signal such as POTS lines or a 2 wire intercom to a four wire system separate send and receive paths such as used in the pro audio world While this task is theoretically quite simple the fact the impedance of most phone lines varies widely across frequency complicates matters The Telos 10 telephone system was the first practical DSP based hybrid and applied the then brand new technology to this problem IEC 1 InterExchange Carrier Long Distance carrier H
115. ewdriver may be pushed firmly under the lip and then lifted out to pop the cap loose 4 You will see a black rectangular transformer labeled T1 To the right of T1 locate and remove the jumpers labeled JP1 amp JP2 These are small jumper blocks which can be lifted off with needle nose pliers 5 You ll want to save the jumper blocks You can store each of them on one of the jumper pins 6 Replace the access cover removed in step 3 by pushing it firmly in place 7 Reconnect the Desktop Director cable 8 Check for proper operation Desktop Director Cable Each Desktop Director is supplied with a 25 foot cable If you need another length you can make your own cable The Director connection is electrically identical to the ISDN S interface so the wiring scheme is the same A straight through not crossover cable is used between the Talkshow System and the Director TIA 568A or 568B standard Category 5 5e cables may be used Category 3 cable is acceptable for this wiring But since category 5e is so common and low cost we recommend that you use it instead Tip Power over Ethernet and our ISDN based Director use different connections for power It is wise to carefully label wiring used with RJ style connectors to reduce the risk of applying power to something that does not expect it We suggest that you use different colored cables or connector boots on the Desktop Director cables so that you can tell them from your Ethernet cables
116. f that fails try DMS Note After you enter or change your SPIDs or DNs you must reboot the Talkshow System Therefore once you have entered SPID1 SPID2 DN1 and DN2 you should turn the power off and then back on Then go to the Status screen and verify Ready status for both channels Clock Loss This setting is relevant only when there are ISDN lines that go to sleep or there is some other unusual ISDN Telco condition Most users should leave this set to the default setting of Hold Last SPIDs and MSN DNs Web Only There will be entry locations for SPIDs and MSN DNs for all line positions whether needed or not After entering the numbers you must click Apply and then Reboot Now at the bottom of the page to activate the ISDN lines ISDN Telco settings ISDN Protocol ETS300 EU a when ISDN clock lost Hold last clock e ISDN Interface A This section is disabled because slot A does not contain an ISDN card SPID1 MSN DN 1 sip2 MSN DN 2 SPID 3 MSN DN 3 SPID4 MSN DN4 SPID 5 MSN DN 5 SPID6 AE MSN DN 6 J E You can check what you actually have by looking at the first line ofthe LCD status screen where you will see POT S ISDN or EMPTY for the A and B line groups depending on how your system is configured Press any menu button twice to get this screen When you are using Natl 1 1 or DMS 100 for your ISDN Protocol mode you must en
117. f this icon Screened Hold Call is on hold and has been screened by the Producer Call is on hold has been screened and is Next Press the Next button to put this line on air Press the X Button to cancel Block All Mode Call is on the handset USER OPERATION 33 Call is on the speakerphone Line is not operational Line Button Columns Talent Mode Each line has 2 buttons the left column for Caller 1 and the right column for Caller 2 Pressing a line button places the caller on that line on air using the corresponding hybrid Pressing a line button when a call is already on air locks the call To unlock press the line button a third time If the handset is lifted or hands free is active the LED next to the hands free button is illumi nated the left line column will connect the line to either the handset or the Director s internal speaker mic rather than to the usual on air hybrid The right line column will continue to work as usual To move a caller on the handset to air press the line button on either the left or right columns Producer Mode The left row of buttons is used to answer calls on the handset or headset Pressing a left column line button while you have a call on the handset or in hands free mode locks the caller To unlock press that Line button a third time The right row of buttons is used to place a caller on Screened Hold For more details on how screened hold work
118. following procedure Highlight the IP address Press OK Press V repeatedly to delete existing IP address Press A until the desired digit is displayed Press OK to accept that digit To delete a incorrect digit press V Press A until the desired digit is displayed OS Se ee NS Once all characters are entered press OK twice to save your entry Return takes you back to the System menu Syslog Server amp Syslog Filter Web only Permit logging to external computers running Linux standard Syslog for troubleshooting purposes Contrast LCD Only Adjusts the contrast of the LCD Values range from 0 to 10 The usual setting is 5 Show Mode LCD Only The Web Setup Wizard is another way to select show mode The term Show comes from our Series 2101 system which supports up to 120 lines and more than 64 hybrids in multiple studios Each Show selects a group of Telco lines and an audio I O Next queues and other functions are distinct for each show The Nx6 and Nx12 have only 6 and 12 lines respectively and 4 hybrids so they are consider ably simpler Single studio modes have only Show 1 However Talkshow Systems also support two studio modes in which case there will be Show 1 and Show 2 Desktop Directors and call screening applications connect to the appropriate show and each show has associated audio inputs and outputs See Section 5 8 for details on the various show modes Choices are DETAILED CONFIGURATION amp REFERE
119. g a digital line at the customer s premises In many countries the NCTE is provided by the Telco The USA is not one of those countries Network Channel Terminating Equipment See NCTE Network Termination See NT Nortel Manufacturer of the DMS 100 switch as well as many other pieces of telecom and network equipment Now called Nortel Networks NT Network Termination The electrical and protocol specifica tions for the user end of an ISDN line See also LT NT 1 An alternative expression for NT1 See NT1 NT1 Network Termination Type 1 The termination at the cus tomer premises of an ISDN BRI circuit The NT1 performs the role of line termination of the U interface and Codes Decodes from the line s 2B1Q coding scheme The customer end of the NT1 in terfaces using the S or T interface The NT 1 is frequently part of the Terminal Adapter and is built in to Zephyr Xstream Zephyr ZephyrExpress Telos TWO and TWOx12 systems sold in the USA amp Canada See also NCTE NTBA Network Termination Basic Access The term used for NT 1 in some countries See NT1 See also NCTE OOS Out of Service An alarm light or condition on a T1 or trunk PBX Private Branch Exchange A privately owned switch Basi cally a PBX is a private business telephone system which also interfaces to the telephone network In some circles PBX implies a manual switchboard whereas PABX Private Automatic bran
120. ge the settings above the diagrams below will 11 Lined Unused l C a update to show the configuration of the rear panel ie Line 12 Shared z lv Ej connectors and internal audio routing Settings will not be saved until you press Apply N a Lines shared with joooo Show Mode Select how the Nx Talkshow system will allocate the hybrids to the phone lines Choices One Show All phones lines assigned for one studio two hybrids total Four Column All phone lines assigned for one studio four hybrids total Requires an Extended Desktop Director to control the lines to hybrids 3 and 4 or by use of the Call control software in 4 hybrid mode Two Shows Shared All phones lines are shared between two shows Show 1 amp Show 2 for two studios two hybrids each show studio Two Shows Split Phone lines are split between the two shows for two studios For Nx12 lines 1 6 for Show 1 lines 7 12 for Show 2 For the Nx 6 lines 1 3 for Show 1 lines 4 6 for Show 2 two hybrids each show studio Two Shows Custom Phone lines are custom assigned by the user to either Show 1 Show 2 or Shared between shows using the Show Line Configuration section The Show field options for each line are only valid while in this mode and ignored for any other show mode selected DETAILED CONFIGURATION amp REFERENCE 61 When enabled the system will place the phone call currently in use either on the hybrid or on the Desktop Director on HOLD w
121. gnostics menu for each Director Select any available Desktop Director connection DD connection The Far end menu gives a detailed error status coming from the Desktop Director side The Near end menu gives a detailed error status coming from the Talkshow System side Tip If you suspect a malfunctioning Desktop Director check its error status before calling Telos support Having these details at hand will allow us to best assist you 44 Section 5 5 3 Using the LCD Buttons and Web for Configuration You use the various configuration settings to adjust audio levels and processing options set the ISDN parameters etc You may use either the front panel LCD button user interface on the Talkshow System main unit or the Web interface to access configuration options The Web interface offers the full set of configuration options while the LCD button includes only the most often needed items The LCD Button User Interface Each of the three buttons below the LCD display calls up the menu item group associated with the button s name fe from phone AGE OUTPUT AUDIO TELCO SYSTEM A s Livewire Pressing any menu button Audio Telco or System twice displays the Status screen Menu Navigation amp Item Selection After you select a menu group you can use the A and V arrow buttons to scroll through the available items You may select a highlighted item for modification by pressing the OK button Once an item is selected you can
122. hat would have no blocking the Telcos are held accountable by the utility commissions to keep blocking below tariffed levels The concept of blocking cannot be applied to packet networks they just lose packets instead only circuit switched networks Blue Alarm Also called an Alarm Indicating Signal AIS A keep alive signal sent if a problem occurs mid span in a T carrier system The blue alarm signal is required because in some cases T 1 repeaters will become unstable if inadequate 1 s density is not maintained BOC Bell Operating Company One of the regional telephone companies that were owned by AT amp T before divestiture in 1984 ice New England Telephone Ohio Bell etc The 22 BOCs were divided among the RBOCs at divestiture See RBOC Both Way Trunks see combination trunks BRI ISDN Basic Rate Interface The common form of ISDN with 2 Bearer Channels and one D channel All three channels are on a single copper pair and encoded with type 2B1Q coding BRITE Basic Rate Interface Transmission Extension A technol ogy where ordinary T 1 trunks or any other digital carrier system are used to extend ISDN BRI service BRITE uses 3 DSO channel per BRI One for each B channel and one for the D channel See Repeater BTN See Billing Telephone Number Business Office The part of the phone company where you call if they mess up your bill to report problems and to order service Not necessarily technic
123. he armed LED blinking When in armed state LED blinking pressing the button cancels the recording feature When the Talkshow System is used in any of the two studio modes the record logic will only operate from Desktop Directors accessing SHOW1 The Record button is disabled for any Director connected to SHOW LCD Screen and Associated Buttons The LCD is used for general information and for configuring the Desktop Director Items can be selected and changed using the six buttons immediately below the LCD screen The buttons in the row immediately below the LCD screen are soft keys and their function will be indicated on the bottom row of the screen Here is the main default screen ODI Hubrid 1 4 Talent f dial uol ume Menu Navigation The LCD screen and menu navigation buttons are used to adjust volume LCD screen backlight and contrast and to change configuration items The function of the three soft keys immedi ately below the LCD screen will vary depending on context and will always be displayed on the screen In most cases the left soft key offers a back function This takes you back to the main screen without saving unconfirmed changes except for in the Volume screen where changes are effected immediately The A amp V buttons scroll through the options in a menu list and the Select button chooses a menu item or confirms an entry Dial Menus Pressing the softkey below the word Dial will display the fol
124. hen the operator host selects a different line This allows users to go back to the line if they need to get more information from the caller Choices are None Talent Producer Both Affects all shows globally The Direct Hold feature allows the operator to seize the next call which has been ringing for the longest time and put the caller he was talking with back on hold in one simple step by pressing the Next key or directly selecting another line button If no call is ringing the process remains the same with callers on hold Pressing the Next key seizes the call which was on hold for the longest time and puts the previous caller on hold This feature also works when manually selecting a new line in a column If existing unlocked calls are present the lines will be placed on hold The choices are None feature disabled Talent enabled only for Desktop Directors in Talent mode Producer enabled only for Desktop Directors in Producer Both enabled for both Producer and Talent modes Enable Auto Answer Line to Hold Enables system to auto answer call and place it right to HOLD queue Caller hears POH audio for assigned show Choices are Off Show 1 Show 2 Show 1 amp 2 For single show mode select just Show 1 Show 2 will not do anything Sharing lines with a second show or second Nx system The Auto Answer feature is designed to allow both systems to handle auto answer but only one show system will actual
125. here there are differ ences between the two units they will be singled out Nx Talkshow Systems have four advanced digital hybrids each with its own AGC noise gate and caller override dynamics using carefully tuned DSP algorithms Each unit also includes DDEQ a sophisticated multi band equalizer which analyzes and adjusts received audio spectral characteristics so that calls sound smooth and consistent despite today s wide variety of phone sets and connection paths Exclusive to Telos Systems the hybrids incorporate special echo cancellation for tricky VoIP and cellular calls as well as adaptive function that reduces the possibility for feedback in open speaker applications Features of the Nx Talkshow Systems include Excellent send to receive isolation using the latest generation of Telos field tested Adap tive Cancellation technology Each hybrid automatically and continuously adjusts to phone line conditions eliminating the hollowness feedback and distortion that occur when send audio leaks into the caller output signal Sophisticated AGC limiting functions and three band Digital Dynamic Equalization DDEQ ensure controlled caller audio levels and spectral consistency Four high performance adaptive digital hybrids Studio grade 20 bit audio conversion combined with noise gating and shaping extracts the best possible audio quality from real world telephone lines High precision sample rate conversion for
126. hnologies include BRITE Virtual ISDN Lightspan and Total Reach Some Telcos do not use repeaters Compatibility between a given NT1 CPE and a repeater is less certain than if that CPE where directly connected to the switch Robbed Bit Signaling A signaling scheme that borrows bits on each T1 channel for use as signaling channels On SF T1 s there are two bits the A bit and the B bit in each direction On ESF T1 s there is also a C and D bit in each direction although they are rarely used Using these bits various older analog trunk interfaces can be emulated over a T1 For instance dial pulse address signal ing using 10 pulse per second rotary style digit groups over these bits Since robbed bit signaling interferes with the least significant bit only 7 bits can be used for sensitive data applications leaving only a 56kbps channel for data applications See also CAS and CCIS Rollover See Hunt Group S interface The electrical interface between the NT1 and the Ter GUIDE TO TELEPHONE TERMINOLOGY 113 minal Adapter or other ISDN equipment ISDN equipment with built in NT1 s do not necessarily provide access to the S interface the Zephyr Xstream Zephyr ZephyrExpress and Telos TWO do Multiple devices can share an NT1 by connecting on the S interface Also known as the S passive bus S0 European term for ISDN BRI See BRI and ISDN2 S2M European term for ISDN PRI See PRI and ISDN 3
127. hown that the 48 volt version usually has better quality transformers and other components resulting in better hybrid performance The other thing to investigate is whether the PBX offers Loop Current Wink or CPC Calling Party Control on the analog ports Without this supervision the Talkshow System will not detect that a caller on hold has hung up and your talent will get a recording please hang up and dial again or dial tone when attempting to air such calls 2 7 Studio Audio Connections Mix Minus The Talkshow System must be fed send to caller audio that is free of the caller audio a mix minus A mix minus is a mix of all of your audio sources that will be placed on air or recorded except the caller audio thus the mix minus designation The European term M 1 mix minus one is perhaps a clearer name A mix minus is also sometimes referred to as a clean feed The important thing to remember is that the hybrid must not chase its tail the condition when its output makes its way somehow back to the input 12 Section 2 Hot Tip Many hybrid installation problems are caused by an inadvertent signal path which creates a loop from the hybrid s output back to its own input Some consoles allow this when certain control combinations are selected by the user In some cases it may be as simple a mistake as assigning the hybrid to whichever bus is feeding the hybrid This is the first place to look when strange or e
128. ignal switched by a Digital Cross Connect System DACS The FDL is stripped off at the DACS interface DSX 1 is also the type of signal that arrives at the user side of a CSU on a T1 line A DSX 1 cable is limited to 655 feet 200 meters DTE Data Terminal Equipment When using serial communica tions such RS 232 V 35 or X 21 the DTE is the device sending receiving from a modem or CSU DSU In contrast to DCE DTMF Dual Tone Multi Frequency The standard tone pairs used on telephone terminals for dialing using in band signaling The standards define 16 tone pairs 0 9 and A F although most terminals support only 12 of them 0 9 and These are also sometimes referred to as Touch Tones Note that while digital data terminals have the same symbols ISDN uses common channel signaling over the D channel and therefore does not necessarily generate any tones at all However many terminals still generate the tones since they will still be used on occasion to access services such as voicemail or automated attendant at the far end using in band tones E1 A common type of digital telephone trunk widely deployed outside the US and Canada Has 31 available 64 kbps channels called DSOs plus a sync control channel for a total data rate of 2 048 mbps E 1 See E1 ESF Extended Superframe A type of Line format supported on T1 circuits The Telco determines the line format and line encoding of your line See Line For
129. ill be indicated by the idle line state on the Desktop Director This state is shown with a single dot in the center of the LED icon display being lit You can also check the line status on the Talkshow System main box LCD Press any menu but ton twice and then press the V button to cycle through the status screen You should see a screen that has A ISDN B ISDN at the top If you do not see this screen just repeatedly press the Vor A button until you see this screen If you do not see READY for each ISDN channel on the Status screen try rebooting the system from the System Reboot menu If you still do not get READY you will need to check 28 Section 3 both your ISDN lines and the ISDN configuration settings IMPORTANT Users in the USA amp Canada should see READY status for each line whenever the system is idle On certain ETS 300 ISDN lines the lines may go from READY to DEACTIVATED after a time interval This is no cause for alarm and you can proceed with the following steps Once you see READY for each channel on the Status screens you can proceed to make a call and verify that the ISDN lines are in fact working We will start with the simplest case a call from your line 1 to your line 2 as follows Confirm that the Desktop Director you will be using has power and says Talent at the top of the screen If it says Producer follow the directions in Section 4 1 to change to Talent mode Pick up the handset and push e
130. in the Talkshow System or with the programming of the Telco switch Usually the inability to dial is a problem on the Telco side such mis programming and not the Talkshow System itself If you are convinced that the problem lies with the ISDN line configuration you may be able to convince the Telco to have an engineer look at the line with an analyzer This works much like the Talkshow System s internal analyzer but with more detail and in a format that is more readily understandable by telephone people This procedure may sometimes be the only way to TECHNICAL DATA amp TROUBLESHOOTING 85 get attention to a line problem It is hard to deny evidence provided by their own gear In rare cases there could be a software bug in the Talkshow System that reacts badly to something coming from the ISDN line Telos customer support will communicate the problem to our engineering guys for resolution See Section 7 41 for information on connecting to the Talkshow System s command line to turn on ISDN logging 6 5 Troubleshooting Desktop Director Problems Most problems with the Desktop Directors are cable related First try a new cable If that fails to solve the problem try the Desktop Director with a short cable plugged directly into each of the Director Ports on the back of the Talkshow System Note that early production Call Controllers require a modification to operate with the Talkshow System Telos customer support can arrange for
131. ine to be aired before that and so on pressing the right line button for the call to be aired next last Block All Pressing this key will clear all lines except lines on air on the handset on hold and priority lines such as the studio hotline This allows you to prevent early callers from getting in on contests until after the DJ has made the announcement and turned off Block All Ringing lines other than priority lines will be dropped and then be blocked from accepting any calls For ISDN lines the caller may get a busy signal or they may continue to hear ringing For analog lines the ringing calls will be connected to a silent audio channel Further calls will get a busy signal Pressing the Block All button again will release the lines and allow incoming calls Conferencing callers on the handset headset The usual operation of the line button is such that selecting a line will cause an existing line in use to be dropped or placed on hold if your system is programmed to do so To conference multiple callers you can use the lock function With the first call on the handset press the left line button to lock that call Now you can select another line without losing the first call If you wish to add a third party lock the second line by pressing the line button for the second caller You can add additional lines in this manner To unlock a line press the line button for that line At any time you can drop all unlocked call
132. information or to report your experiences with other switches Using POTS Most PBX or VoIP telephone systems can provide analog ports for analog equipment such as modems and fax machines You will need such ports to connect a POTS Talkshow System to a PBX With all calls passing through the PBX calls into the main station number can be transferred over to the on air system If the PBX supports hook flash transfer on the analog ports you can even use this feature of the Talkshow Systems to transfer lines from the on air system back to the PBX This can also be done if the PBX and Talkshow System6 share Centrex lines A few years ago we advised that hybrid performance was best when equipment was connected directly to the lines from the Telco Today this is not always true If the PBX is connected to the Telco using digital lines such as T1 E1 PRI S2M etc performance through a digital PBX will generally be better than direct connections to copper analog circuits This is particularly so if you are located a significant distance from the Telco central office The biggest variable when connecting to a PBX analog port is the quality of the analog port We have found that many PBX manufacturers offer more than one analog port option The better port can usually be distinguished by its higher cost and the fact that it may offer 48 volts rather than 24 volts The Talkshow System will work with either voltage however experience has s
133. information that follows If you know who to call and you are ready to provide complete information the ISDN ordering process can be simple and straightforward This guide takes you through the process step bystep Telos support is always available to assist On one ISDN BRI circuit there are three simultaneous bi directional channels two 64kbps bearer or B channels for the transmission of user information and one 16kbps data or D channel for call set up and status communication That is why this is sometimes called 2B D service You can use the B channels but only the phone companies and equipment manufacturers can usually use the D channel ISDN comes into your facility on two wires just like plain old telephone service POTS From the perspective of telephone network routing each channel appears to be a separate line with it s own number and independent dial out capabilities Since each has to be dialed or answered separately they appear to be lines to users also The Talkshow System s internal ISDN interface is used to connect to digital ISDN telephone lines It easily adapts to the various types of service offered by the range of Central Office CO switches installed by telephone companies in the USA and Canada Dealing with the Phone Company As is often the case when broadcasters interface with phone people the lines of communication can get a little tangled The first order of business is
134. inus To avoid this error the signal path could be permanently disconnected by removing the summing resistors or some such creative operation The Talkshow System has multiple hybrids and works best if two faders can be assigned to the telephone system with two associated mix minuses one for each telephone line column on the Director This is probably not going to be easy with an older console But the Talkshow System has an option to work with a single external mix minus by making an internal cross connection of the hybrids See Section 5 7 for more on this 14 Section 2 Using a small mixer In this scenario a small audio mixer is used to record interviews off the telephone line using a single hybrid The mixer s main bus is fed to the recording device Both the microphone and the hybrid will be brought up on the faders so the interview can be recorded Most small mixers such as those made by Mackie have one or two Aux send busses so we will use these to feed the telephone system We will turn up Aux for the microphone but we will make sure it is turned fully off for the each channel that has the corresponding caller audio Using a production style console The production style consoles often used for TV audio will have multiple Aux send busses that can be used in a similar way to the small mixer example above Each hybrid is sent from an Aux bus and everything the caller needs to hear is mixed into that bus taking care to keep the h
135. io When feeding unbalanced devices use pin 2 for the signal and pin 3 for the ground Do not tie pin 3 to pin 1 Note that this is different from most devices with active balanced outputs The Talkshow System has a sophisticated output stage that emulates a transformer source AES Digital Outputs The AES outputs are only present just like the AES inputs if the AES card was purchased with the unit or has since been installed in your unit The single AES card handles both inputs and outputs Characteristics are AES3 compliant 110 Ohms The output is synced and sample rate converted to either the frequency of the AES Input 1 signal or an internal 48kHz clock depending upon a menu selection AES3 Output XLR PIN DESCRIPTION 1 Ground 2 Signal 3 Signal 18 Section 2 Since like the inputs each AES output has two hybrids you may have to use a routing switcher distribution amp or console routing function to direct the two signals appropriately for your installation An alternative would be to bridge two AES inputs across each output which is generally an acceptable practice Then the mixing console s input selector would be used to extract the correct hybrid signal for each telephone fader channel AES 1 Left is from Hybrid A AES 1 Right is from Hybrid B AES 2 Left is from Hybrid C AES 2 Right is from Hybrid D 2 8 Desktop Director Ports At the time this manual was released the following de
136. ion when using open speakers Send EQ AGC Mix minus modes See Section 4 See Section 5 4 and 5 11 See Section 5 4 and 5 11 See Section 5 4 and 5 11 See Section 5 4 and 5 11 See Section 5 4 Show Modes one Talkshow System serving two studios etc See Section 5 7 31 User Operation 4 1 Desktop Director The Telos Desktop Director is both a control surface for putting calls on air and a telephone set for conversations with callers on the studio telephone lines The Director has two modes of operation Producer screener mode and Talent on air mode Tip Since certain functions work differently in Talent versus Producer mode it is important to understand that there are two modes and how to switch between them Status Symbol Icons The Desktop Director uses Status Symbol icons displayed on the LED matrixes located between the two columns of line select buttons Here is what each icon means Line is ready for incoming or outgoing calls Line is in use by another studio in two studio modes only Line is ringing Call is on hold 32 Section 4 Call is On Air Column 1 Call is On Air Column 2 Call is On Air amp Locked Column 1 Call is locked and cannot be dropped or placed on hold To unlock press the line button to the left of this icon Call is On Air S Locked Column 2 Call is locked and cannot be dropped or placed on hold To unlock press the line button to the right o
137. irector is used for off air conversation and call screening 8 Warranty and Application Caution TELOS Talkshow System LIMITED WARRANTY This Warranty covers the Products which are defined as the various audio equipment parts software and accessories manufactured sold and or distributed by TLS Corp d b a Telos Systems hereinafter Telos Systems With the exception of software only items the Products are warranted to be free from defects in material and workmanship for a period of two years from the date of receipt by the end user Software only items are warranted to be free from defects in material and workmanship for a period of 90 days from the date of receipt by the end user This warranty is void if the Product is subject to Acts of God including without limitation lightning improper installation or misuse including without limitation the failure to use telephone and power line surge protection devices accident neglect or damage EXCEPT FOR THE ABOVE STATED WARRANTY TELOS SYSTEMS MAKES NO WARRANTIES EXPRESS OR IMPLIED INCLUDING IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE In no event will Telos Systems its employees agents or authorized dealers be liable for inciden tal or consequential damages or for loss damage or expense directly or indirectly arising from the use of any Product or the inability to use any Product either separately or in combination with other equipmen
138. istance competition has reached your area you can determine your Intra LATA long distance carrier by dialing 1 area code 700 4141 What does the ISDN Cause phrase say This phrase which will appears on the ISDN protocol log after dialing comes directly from the 84 Section 6 telephone company equipment and can be valuable for troubleshooting Generally these phrases are self explanatory Getting them means you are at least talking to the Central Office equip ment There is a full description of the phrases in Section 6 7 Condition Intermittent ISDN Problems These are among the most difficult kinds of problems to solve Rest assured however that we have yet to lose one First try disconnecting the ISDN line and waiting a full 60 seconds leave the Talkshow System on and then reconnect the line Does the problem occur on all of the ISDN BRI circuits or just a single BRI circuit 2 lines If the problem is limited to a single BRI circuit i e both lines on that BRI are showing the problem then the problem may be with that BRI line or with that ISDN interface card Move the BRI line in question to a different interface card don t forget to reprogram the SPIDs If the problems moves it must be with the BRI itself If it remains with certain lines on the system then it is probably in the ISDN interface card Swapping cards between two of the interface slots will confirm or disconfirm this If the
139. leveland OH 44114 USA 1 216 241 7225 phone 1 216 241 4103 fax 1 216 622 0247 24 7 Technical Support SupportO 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 voltage which may be sufficient to constitute a risk of shock This symbol wherever it appears alerts you to important operating and maintenance instructions Read the manual CAUTION THE INSTALLATION AND SERVICE INSTRUCTIONS IN THIS MANUAL ARE FOR USE BY QUALIFIED PERSONNEL ONLY TO AVOID ELECTRIC SHOCK DO NOT PERFORM ANY SERVICING OTHER THAN THAT CONTAINED IN THE OPERATING INSTRUCTIONS UNLESS YOU ARE QUALIFIED TO DO SO REFER ALL SERVICING TO QUALIFIED PERSONNEL WARNING TO REDUCE THE RISK OF ELECTRICAL SHOCK DO NOT EXPOSE THIS PRODUCT TO RAIN OR MOISTURE USA CLASS A COMPUTING DEVICE INFORMATION TO USER WARNING This equip ment generates uses and can radiate radio frequency energy If it is not installed and used as directed by this manual it may cause interference to radio communication This equipment complies with the limits for a Class A computing device as specified by FCC Rules Part 15 Subpart J which are designed to provide reasonable protection against such interference when this type of equipment is operated in a commercial environment Operation of this equipment in a residential area is likely to cau
140. ll over from busy channels to idle channels This is sometimes referred to as ISG incoming service grouping or rollover lines This would usually be a requirement for an on air phone system You may choose to exclude 1 or 2 channels from the hunt group ie the Hot line and the Warm line Be sure to specify to your Telco how you expect this to work Line Deactivation The Euro ISDN standard allows ISDN BRI lines to be deactivated by the Telecom when not in use It is impossible to distinguish a deactivated line from a line which is defective and out of service If your application is critical you may wish to request that the Telecom disable line deactivation FAXABLE ISDN ORDER FORM for Euro ETS 300 The following form may be used to place orders for ISDN lines in countries using the ETS 300 protocol These should give the Telecom all of the information they need ISDN BRI LINE ORDER ETS 300 Euro ISDN Telos Nx6 Nx12 To Telephone Company Attention From Company Address City State ZIP Phone Contact Location for line Company Address City State ZIP Phone Contact Date needed Termination date Number of BRI circuits required We request ISDN BRI ISDN 2 S0 circuits for use with the Telos Nx6 Nx12 digital telephone sys tem This device interfaces studio audio equipment to ISDN It requires Circuit Switched Voice CSV The Nx6 Nx12 have an integral BRI terminal adapter which supports the ETS 300 Net
141. llow these steps 1 Press the MENU softkey 2 Scroll down until Select Show is highlighted and then press lt SELECT gt 3 Press lt V gt or lt A gt arrow keys to select Show 1 or Show 2 Selecting a Show using a Call Controller A given Call Controller can be associated with Show 1 or Show 2 by using a special key se quence To assign a Console Director to a show follow these steps 1 All lines must be idle or must be on hold before you can change shows 2 Press the following button sequence to associate the Console Director with Show 1 lt gt lt S gt lt H gt lt O gt lt W gt lt 1 gt or 74691 3 Press the following button sequence to associate the Console Director with Show 2 lt gt lt S gt lt H gt lt O gt lt W gt lt 2 gt or 74692 Selecting a Show using a Console Director The Console Director is always in Talent mode A Console Director can be associated with Show 1 or Show 2 by using a special key sequence To assign a Console Director to a show follow these steps 1 When selecting a show no line can be on the hybrid All lines must be idle or must be on hold before you can change shows 2 Press the following button sequence to associate the Console Director with Show 1 Hybrid 1 lt gt lt S gt lt H gt lt O gt lt W gt lt 1 gt or 74691 3 Press the following button sequence to associate the Console Director with Show 2 Hybrid 2 lt gt lt S gt lt H gt l
142. lowing screen Phone Book back 36 Section 4 Select the Redial option to redial the last number you dialed It will store the last 20 phone numbers dialed out Use the UP and DOWN buttons to highlight the number you wish to dial and press the SELECT button to dial The Phone Book gives the user the ability store frequently dialed numbers You can store up to 10 phone numbers Select this option and use the UP and DOWN buttons to highlight the number you wish to EDIT and or select Once highlighted press the SELECT button to dial the programmed number You can edit these entries via the Desktop Director or through the web interface When in Split Show mode each show has their own Phone Book Volume Menus Pressing the soft key button below the word Volume will display the following screen just Uolume finger Speaker Handset Press A or V to choose which level you want to adjust Pressing will decrease while pressing will increase the volume level for the item displayed The change will take place immediately Pressing the soft key button under the word back or the Select button will return you to the main screen The following volume levels can each be individually adjusted Ringer Adjusts the volume of the ringer Note that the ringer will usually be turned off whenever a microphone channel is on Speaker Adjusts the level of the Desktop Director s loudspeaker Handset Sets the level of the
143. ly seize the line and place it on hold The other show Nx system will get a status that the line is on hold There is no option to allow one show or system to answer the line over another show or system just which system reacts first to the ringing line will place it on hold Fix Locked Lines to Hybrid When enabled prevents a locked line from being moved to a different hybrid Choices are Off Show 1 Show 2 Show 1 amp 2 On for single show mode Block Priority Lines from Next Queue When enabled prevents lines assigned as Priority Lines to be inserted into the NEXT queue hence prevents users from placing these lines on a hybrid by using the NEXT button Choices are Off Show 1 Show 2 Show 1 amp 2 On for single show mode Show uses GPI from GPI option for the show can be selected to use either the backpanel sub D connector or use Livewire protocol signaling Use Show 1 selection for single show mode Settings will be saved when you press APPLY 62 Section 5 Show Line Configuration This section allows additional configuration of the lines under the following fields Name Add a custom name to the lines ie Request 1 HotLine Show Line sharing options for configuring the Two Shows Custom mode only These options will be ignored if you have selected any other Show Mode under the Show Configuration section Selectable options for this field are Unused Show 1 Show 2 and Shared Priority When checked selected lines
144. mat ETS 300 The pan European ISDN protocol standardized by ETSI This protocol is used throughout Europe and has been adopted in many other countries outside the USA amp Canada See also MSN ETSI European Telecommunications Standards Institute Euro ISDN See ETS 300 Exchange Another name for a Central Office most often used in European countries Also used in the USA amp Canada to refer to a particular 3 digit prefix of a 7 digit telephone number See CO Extended Superframe See ESF FDL Facilities Data Link A bi directional data link available on T1 circuits when the ESF line format is used The FDL is primarily used by the Telco to poll the CSU for error statistics Four Wire A circuit path using separate pairs for send and GUIDE TO TELEPHONE TERMINOLOGY 109 receive This term is also used when referring to digital channels that inherently have discrete send and receive paths regardless of the number of pairs or other media used See also Hybrid Frame A unit of data which is defined by the specific communica tions protocol used See Line Format T1 FXO Foreign Exchange Office termination A line or port meant to connect to connect to the POTS output of a Central Office Such an interface goes off and on hook to signal status and expects to receive ringing current FXS Foreign Exchange Station termination A line or port mean to connect to a telephone Such an interface must look for current
145. modification of these units at no charge Section 2 has details on the allowable wiring configurations for the Desktop Directors and other control surfaces There are distance and wiring restrictions Note that when two Directors share a single port an external power supply is required and the termination resistor must be removed on the unit closer to the power supply The diagnostics on the Desktop Director and the Talkshow System s status screen will help isolate bad wiring or hardware If errors are accruing try systematically swapping ports and swapping Desktop Directors to see if the problem stays with a given port Desktop Director or cable 6 6 Replaceable Modules and Hardware Servicing WARNING DANGER Remove all connections including the Telephone Audio and Electrical Mains Power cord before opening the unit Failure to do so could result in injury or death Do not open the unit unless you are qualified to do so Removal of the top plate gains access for service To do so remove the 10 Philips head screws Replaceable modules While we do not expect you to do much repair or troubleshooting of the individual circuit boards the system does have several removable modules which can be replaced The system has the following replaceable modules 1 Power Supply Part 1281 00038 2 ISDN Line interface card Part 1701 00140 U interface USA and Canada or Part 1701 00139 S interface Worldwide 3 Analog POTS Line
146. most common complaints of talk hosts is that they go to a line where they expect a caller to be waiting only to be met with a blaring annoying dial tone The chance of this happening with an ISDN line is nearly zero Another common error is the condition where a talent punches up a line that looks free but which actually is just about to begin ringing and connects to a surprised caller This condi tion called glare results from the delay in the ring signaling which comes from the nature of the analog line s ringing cadence ISDN eliminates the ambiguous status period We don t know why the condition of picking up a line which was about to ring and getting an incoming call instead of dial tone is called glare Perhaps it s because the looks air staff give engineers when this happens too often If you find out be sure to let us know 7 Better Line Monitoring Call Setup and Supervision Since there is a full time connection between the central office and the terminal on the D channel it is possible to detect when a line is not working On an analog line one discovers a problem only from a failed attempt to use the line With ISDN you can quickly detect when a line is a not working Even when ISDN Basic Rate Interface BRI lines are not routinely available ISDN Primary Rate PRI lines are BRIs have 2 channels per line while PRIs have either 23 or 30 PRIs are well entrenched for use with large corporate PBXs so are likely to be around
147. n dialed See Called Party Address CSD Circuit Switched Data A dial up data communications channel which once established looks like a transparent data pipe Also the type of ISDN service required to utilize this capability of an ISDN circuit In contrast to CSV 108 Appendix 2 CSU Channel Service Unit The NCTE used in the USA amp Canada to terminate a T1 line Typically the CSU must be provided by the end user See NCTE and DSX1 CSU DSU A device which incorporates the functions of a CSU Channel Service Unit and a DSU Data Service Unit Most com monly it interfaces between a Switched 56 or Dedicated Digital Service circuit and a user s data equipment such as the Zephyr CSV Circuit Switched Voice A dial up communications circuit for voice grade communication Also the type of ISDN service re quired to use this capability of an ISDN circuit In contrast to CSD Custom ISDN USA amp Canada An ISDN protocol which pre dates National ISDN 1 In most cases National ISDN 1 is also available The Northern Telecom DMS 100 switch Supports DMS Custom Functional ISDN The AT amp T Lucent 5ESS switch Sup ports Custom Point to Point PTP and Custom Point to Multi Point PMP The ISDN protocol has no relation to where one may call Telos equipment does not support PMP D Channel Data Delta Channel Depending on who you ask it is Data or Delta The channel which handles ISDN network related data between the use
148. n the caller gain is reduced The effect is somewhat seesaw like If Feedback Reduction in the Audio menu is enabled the Talkshow System will increase ducking in the announcer to caller direction As a guideline you will need more ducking when using open speakers This helps reduce the oc currence of feedback and also reduces the chances of the caller hearing himself via the speaker to mic path which has an unnatural quality and can be disturbing In a morning zoo type of scenario the setting should be even lower as there will typically be multiple mics used with open speakers Fortunately this scenario coincides with a more duplex style of operating where pre recorded bits may be played to callers and must be heard without interruption The default setting is 7 You may choose how much of this effect you prefer from Half Duplex 16dB which makes the Talkshow System operate like a one way at a time speakerphone to Full Duplex 0dB which disables ducking A good starting value would be 6dB DETAILED CONFIGURATION amp REFERENCE 77 Feedback Control We ve carefully designed the Talkshow System so that problems with feedback should be rare However despite excellent trans hybrid loss and a number of other features you can probably induce feedback if you try sufficiently hard such as by cranking up your studio monitors to very loud levels Some suggestions for what you should do if feedback does occur follow Enable
149. nd Reduced Feedback during Multi line Conferencing When conferenc ing is required on analog circuits hybrids are needed to separate the two audio paths in order to add gain in each direction When the gain around the loop exceeds unity the unpleasant result is feedback Since the conference path in a studio on air interface usually includes four AGC functions the hybrids must be sufficiently good to cover the additional gain that may be dynamically inserted by these AGCs With digital telephone lines the hybrid function is more effective and more reliably so across a variety of calls That means more gain can be inserted between calls before feedback becomes a problem 6 Digital Call Setup and Supervision AAnalog lines use a strange mix of signaling to convey call status Loop current drop signals that a caller has disconnected and blasts of 90 Volts at 20 Hz mean someone wants you to answer Should we be using a mechanism designed to bang a hammer against a metal bell to transmit network status information in the 21st Century ISDN uses a modern digital approach to controlling calls and conveying status information about them ISDN call set up times are often only a few 10s of milliseconds enhancing production of a fast paced show Perhaps more importantly when a caller disconnects while waiting on hold the ISDN channel communicates this status change instantly This contrasts with the usual 11 second delay on most analog lines One of the
150. nded to save you a trip to the back of the rack This reboot is equivalent to removing power and then restoring it Firmware Update Web Only See Section 5 10 60 Section 5 5 7 The Setup Wizard Menu Web Only This menu option is a powerful tool and can be of great assistance to those setting up their units for the first time It has the most pertinent information regarding your basic configuration It includes a graphical representation of the back panels that can show you the correct connections to make base on your proposed configuration choice Show Configuration The Nx Talkshow system basically has two show modes with subsets or variations of these modes found in these options This section provides options related to the shows you will select and configure as needed Show Configuration i Show Line Configuration Show Mode Two Shows Custom v Line Name Show Priority SEREN Hold previous when select Line None 1 Line 1 Shared z E m Enable auto answer line to hold fo Off z 2 Line 2 Shared E L Fix locked lines to hybrid Off x pe 3 3 Line 3 Shared E m Block priority lines from next queue cd 4 Line 4 Shared a C im Show 1 uses GPI from Backpanel z 5 Line 5 Show 1 E B Show 2 uses GPI from Backpanel z 6 Line 6 Show 1 3 r E T Line 7 Show 2 Y F E 8 fines shw2 A TF r Mix Minus Routing mode Single 9 mes Show E K r AES channel mode c 10 fune 0 show2 v r As you chan
151. ne column If no call is cur rently designated next the call which has been on Screened Hold the longest will be taken If a screener is not used the line which has been ringing longest will be taken If no line is ringing the line on hold the longest will be taken Block All Pressing this button will clear all lines except lines on air on the handset on hold and prior ity lines such as the studio hotline Calls ringing in on lines other than priority lines will be dropped and the lines will then be blocked from accepting any calls Pressing Block All again will release the lines and allow incoming calls This allows you to prevent early callers from getting in on contests until after the announcement is made and Block All is switched off Hot Tip As you may know there have been several lawsuits in the USA regarding to the fairness of radio contests By consistently activating the Block All feature before announcing a contest you will reduce the liability for yourself and the station Conferencing multiple callers on air using call locking If you only need two calls at once you can just use the two line columns with one call on each However if you need more than two or if your system is serving two studios and you only have one active line column you can use the lock function for conferencing Press an already on air line button a second time to lock the line The status symbol will change to display the locked icon
152. ng s s 6 5 Oke ee ee a ae wae ee 41 5 2 Status Displays co s ee a ar E a 42 Lin Status o os hae aa SAG eee es 42 Desktop Director Status ee ee 43 5 3 Using the LCD Buttons and Web for Configuration 44 The LCD Button User Interface 2 ee ee 44 The Web User Interface 2 ee 44 5 4 TheAudioMenu 1 aawa e eae ee ee 45 Disableilivewite ooo a ee a Ra 45 Nominal Input Level o o o o o o o o o o oo ee 46 Nominal Output Level o o o o o o o oo o 46 Meter Mode ooo e Sas a A 46 Phone Send Adjustment o oo o o 47 CaMlIDUCKING gt si ad a dona do aoe 47 ReceiveAGG o 2 vi Pe wa AAD e 47 Noise Gates u meaa CR RM a oe oN Re ae wee es 48 Receive EQ Caller Dynamic Digital EQ DDEO 48 Feedback Reduction o o 48 Enable Adapt Burst s s s caei o e a E 48 MIXMINUSIMOd 2 ires a See a ce aa aa a See a dad 49 RES MOd y cti o ad 51 5 5 Metelo MenU 2 raa emea ea AA 53 ISDN Protocol 2 6 0 Bice ai di dea le de coe le 53 vi A AA ER eo ES 54 SPID SNA MSN DNS 00000 wie eR Soe oe we 54 POTS COUNT is ica oS Aw ole lee Gok ok AR ee 56 CallerlD Alert Signal Web only 2 2 2 eee eee 56 Ring in Before CID Web only o o o o o ee ee 56 HOOKHASH s vis ook ooh ee dee deal E oe ER ae 56 CPC loop drop threshold ms Web only 57 Loop Current Checking Web only o o o o o o o o o 57
153. ng connection in addition to the power leads Do not defeat this safety feature In addition to creating a potentially hazardous situation defeating this safety ground will prevent the internal line noise filter from functioning Basic Configuration 3 1 Step by Step Here is a step by step guide to basic Talkshow System configuration for a new installation Enter TCP IP addresses if used for Web interface remote control and Livewire audio networking See Section 5 3 Set Input Source for Analog AES3 or Livewire See Section 5 8 if you will be using Livewire If you are using Livewire configure it using the Web interface Connect your mixing console mix minus es to the Talkshow System inputs and the Talkshow System outputs to console inputs If you have analog Telco lines configure for your region See Sections 3 3 and 5 5 If you have ISDN Telco service Are you familiar with ISDN If not read Section 2 2 Configure ISDN See Section 3 4 Confirm ISDN is working Read up on Desktop Director Operation See Section 4 Configure the various audio options for processing etc See Section 5 4 If you will split the Talkshow System between two studios see Section 5 7 You may use either the Web interface or the front panel LCD button interface for most con figuration operations The Web may be more convenient since you have a PC s rich interface at your service and you can work by remote S
154. nnected to a U interface However as many as eight terminals may be paralleled onto the S bus In the USA amp Canada a direct connection to the bare copper two wire U interface is required Therefore the USA amp Canada interface module includes an integral NT1 and has RJ 11 style jacks In the rest of the world the telephone company provides the NT1 and the four wire ISDN S interface will be used with an eight pin RJ 45 style jack Normally we ship Talkshow Systems to North America with the U interface module and to other areas with the S interface module In either case each ISDN interface has three connectors Since each ISDN connection has two channels this means each interface module handles 6 dialtones or lines Data and Voice ISDN lines may be used for voice signals encoded in standard fashion to allow inter working with analog telephones or may be used to transmit digital data streams The latter mode is used for such applications as high speed Internet access It is also the mode used with MPEG codecs in which case the ISDN line may be carrying voice signals but is doing so in a format that is not compatible with the analog part of the telephone network The distinction is made in the automatically conveyed Setup message that begins each call Normally Talkshow Systems use only the voice mode so data capability is not necessary Often voice costs extra whereas this is rarely true for data Of course you m
155. nterface connector pin out for synchronous data V 35 uses balanced signal and data lines The Zephyr Zephyr Xstream models 9X01 9X00 support V 35 using part 9812 cable Variant The particular protocol i e National ISDN 1 or ETS 300 running on a specific switch Not all variants are valid for a specific switch The switch brand and model plus the variant defines the ISND protocol Applies to configuring the 2101 See ISDN Protocol Virtual ISDN An alternative to repeaters which uses a local Telco Switch to act as a repeater and which then sends the signal onto another switch which supports ISDN See also Repeater Work Order See Trouble Ticket X 21 A serial data interface for synchronous data popular in Europe X 21 uses balanced data and unbalanced signal lines The Zephyr models 9X01 and 9X00 support X 21 using part 9822 cable Yellow Alarm An alarm on a T carrier circuit that is returned by the local equipment if it is in a Red Alarm state A Red Alarm indi cates a loss of inbound signal a Yellow alarm indicates indirectly a loss of outbound signal See Red Alarm Blue Alarm and LOS USA ISDN ORDERING GUIDE 117 A3 USA ISDN Ordering Guide This section applies to users in Canada amp USA and is intended to help you when it comes time to order your ISDN lines We assume you are familiar with ISDN s basic concepts If you have not read the ISDN section in the manual doing so now will help you better understand the
156. o disconnect has been made Awaiting acknowledgment Incoming Ring An incoming call request has been detected and is being processed Accepting Call Talkshow System has accepted an incoming call and is waiting for the Telco switch to connect it Connected All signaling is complete the voice channel is available for use A caveat regarding the Ready indication If you have this indication with the Talkshow System s Telco mode set to AT amp T Cust and you cannot dial it is likely that your line is Natl I 1 or DMS 100 The AT amp T Custom PTP protocol does not require SPIDs so the Talkshow System 82 Section 6 does not send or check them it just assumes the line is OK in this regard so it gives the ready indication though the line is actually waiting for the SPIDs Note It is normal for Euro ISDN ETS300 lines to go to the Deactivated state when not used If the system can dial normally under these circumstances this indication is not a cause for concern Is the problem only with one BRI circuit or does it move when you switch lines around TIf the problem remains with the line then call your Telco On the other hand if a line that works in one interface fails to work in another interface of the same Talkshow System then you have an problem with your ISDN interface card or possibly elsewhere in the Talkshow System Are you using the Talkshow System s internal NT1 USA amp Canada only The LED indicator of the status of
157. o is present Intermediate settings range from 1 to 15dB Section 5 11 has more information on this function Ducking is desirable for a variety of reasons Allows the announcer to override the caller by causing the received caller audio to be attenuated ducked when the announcer speaks This is often desired for aesthetic effect and allows the announcer to remain in control of the conversation Helps to have open loudspeaker monitoring of callers without feedback Dynamically improves effective trans hybrid loss to reduce leakage when necessary There is also a ducking of the announcer audio to the caller This is only a few dB unless Feed back Reduction is enabled as described below This send ducker has a fixed value Receive AGC The receive Automatic Gain Control serves to improve level consistency of caller audio Nominal levels of the telephone network vary as much as 30dB from call to call In the past decade our experience has been that loud calls have become just as much a problem as low level callers Therefore we have changed the AGC to act as a platform leveler The Talkshow System uses a dB linear approach to AGC with a feed forward topology which provides a consistent sound regardless of the drive level This smart compressor normalizes levels while retaining the natural dynamics of the caller s voice This selection controls the aggressiveness of the AGC process taking values from OFF 1 4
158. o this Desktop Director s built in speaker phone If the handset is on hook this button places the next line selected on the left column on the speaker phone Mute Button This button mutes the handset microphone and hands free microphone on the Desktop Director The LED beside this button will light to indicate that these microphones are muted Pressing this button again will cancel the muting function Hook flash Transfer This button performs a hook flash briefly interrupts loop current on POTS lines Depending on the telephone lines used this key may serve to activate features such as transfer or 3 Way Calling This button does nothing on ISDN lines User Assigned Control Button This button controls an output on the parallel control port This output can control equipment external to the Talkshow System such as the delay dump input on an obscenity delay unit USER OPERATION 35 Record Control This is intended to control an external recording device It operates only if connected Press ing the button once when nobody is on air arms the recording feature LED blinking Once armed taking a call starts the recorder automatically LED solid When the call is dropped the recorder will be stopped and the system will re enter the armed state Pressing the button when a call is already on air starts the recorder LED solid While recording LED solid pressing the button stops the recorder and puts the system back in t
159. og loop start phone service PRI ISDN Primary Rate Interface A form of ISDN with 23 B Channels and one D channel All 24 channels are on a single cable Functionally related to T1 telephone circuits In Europe PRI has 30 B Channels and one D Channel and one Sync channel See also B channel and D channel Provisioning The act of configuring an ISDN or other telecom munications path Also refers to the complete line configuration information PS2 Power Power provided on pins 7 and 8 of the S interface cable This power is used so that a NT1 can provide power to a ter minal usually a phone In some cases it is used to allow a terminal to power an NT1 The USA versions of the Zephyr and Zephyr Express supply PS2 power in the S jack The Zephyr Xstream and Telos TWO provide this on all versions This power arrangement is also used in the Telos TWOx12 and 2101 Studio Interface to power Desktop Directors PTP USA AT amp T Custom Point to Point Custom ISDN Protocol Point to Point lines have only one incoming phone num ber which must be dialed twice to connect to both lines the first call goes to line 1 and the second call rolls over to Line 2 See Custom ISDN and ISDN Protocol RBOC Regional Bell Operating Company USA One of the regional companies formed when AT amp T was forced out of the local telephone business Each RBOC or baby bell owns a number of
160. og line if any The result is much lower leakage 2 Higher Send Levels We don t have to be concerned about regulations designed to prevent crosstalk on analog lines so we can increase send to caller levels 3 Lower Distortion The analog digital conversion chips used in telephone central offices are poor compared to the converters used in professional audio equipment Fidelity is not an important consideration when telephone equipment designers choose parts for this func tion In a professional interface for studio applications we can afford to design in much better converters Noise shaping functions permit a larger word length converter to provide significantly better distortion and signal to noise performance 4 Lower Noise Because they are digital ISDN lines are not susceptible to induced noise Analog lines are exposed to a variety of noise and impulse trouble causers as they snake across town on poles and through your building Hum is the main problem given most lines proximity to pole transformers and AC power wires but there is also impulse noise from INTRODUCTION 3 motors switches and other sources Digital lines convey the bits precisely and accurately from the network to your studio equipment without any perturbation so the audio remains clean Even when the caller is using an analog phone line the digital connection on the studio side makes for noticeably lower noise and better overall quality 5 Higher Gain a
161. ol and an SDK for Windows see the downloads page section entitled Series 2101 development Kit Telnet A telnet session can access certain advanced functions and logging primarily for troubleshoot ing when requested by Telos support Use your favorite telnet client to connect to the IP address and port 23 of the Talkshow System Some Web browsers support telnet Entering telnet 192 168 0 230 23 into the browser s address field might work When starting a telnet session a Login name is requested The default is Telos DETAILED CONFIGURATION 8 REFERENCE 73 Windows Telnet Since Windows comes with a simple telnet client we will give you some tips on using this particular software Run the HyperTerminal application at Accessories Communications Hy per Terminal Then from the File menu select New Connection Enter a name such as Talkshow System and select an icon When the window below appears enter the Talkshow System s IP number 192 168 0 230 this is the Talkshow System default use your actual IP number if you have changed it enter 23 for Port Number and select TCP IP for the Connect using field Pressing OK will then take you to the telnet transaction screen Connect To Ax Nx6 g Enter details for the host that you want to call Host address 192 168 0 230 Port number 23 Connect using TCP IP Winsock Note that in the following section we will use bol
162. om users have made their way into software revisions or new products Please contact us with your comments Service You must contact Telos before returning any equipment for factory service We will need the serial number located on the back of the unit Telos Systems will issue a Return Authorization number which must be written on the exterior of your shipping container Please do not include cables or ac cessories unless specifically requested by the technical support engineer at Telos Be sure to adequately insure your shipment for its replacement value Packages without proper authorization may be refused US customers please contact Telos technical support at 1 216 241 7225 All other customers should contact your local representative to make arrangements for service We support you By Phone Fax 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 00 AM to 5 00 PM USA Eastern Time Monday through Friday Our fax is 1 216 241 4103 By E Mail Technical support is available at Support Telos Systems com All other inquiries at Inquiry Telos Systems com Via World Wide Web The Telos Web site has a variety of information which may be useful for product selection and support The URL is http www Telos Systems com 10987654321 Telos Systems USA Telos Systems 1241 Superior Avenue E C
163. om_ 15 dBfs 9 dBfs PML Meter Mode Selects the audio that will be indicated on the LED meters 4Studio Out Each studio output goes to a meter from the top down 4 Studio In Each studio input has a meter from the top down POH in The two Program on Hold inputs go to the top meters Phone 3 amp 4 Send and receive to from hybrids 3 amp 4 Phone 1 amp 2 Send and receive to from hybrids 1 amp 2 Studio 3 amp 4 Input from and output to studios 3 amp 4 Studio 1 amp 2 default Input from and output to studios 1 amp 2 the meter markings correspond to this one DETAILED CONFIGURATION amp REFERENCE 47 Phone Send Adjustment This setting adjusts the level of the hybrid into the telephone line after the send limiter and is calibrated with OdB representing the usual send level into the line The default is OdB which should be OK for most telephone lines If callers consistently com plain about low audio level and you have verified that this is a problem you can increase the send level However doing so can reduce effective hybrid performance and increase leakage Conversely lowering this setting can reduce leakage Call Ducking This control adjusts the amount of ducking that occurs to the received telephone audio when send to caller audio is present A value of Full Duplex means that the telephone audio is never ducked or attenuated while Half Duplex applies maximum attenuation when send audi
164. ome configuration can only be done by Web such as setting ISDN Directory Numbers and SPIDs and making Livewire channel assignments See Section 5 3 for detailed information on using these interfaces 3 2 Configuration for Telephone Lines The Talkshow System has two Telco interface positions designated A and B The Nx6 will only have slot A populated Slot B can be populated if you choose to convert the Nx6 to a Nx12 The Nx12 will of course have both slots populated There are three module types and you can determine which you have by noting the connectors 26 Section 3 Analog POTS There will be 6 RJ 11 jacks ISDN U interface There will be 3 RJ 11 jacks ISDN S interface There will be 3 RJ 45 jacks With ISDN there are two voice channels for each physical connection so only half as many jacks are needed The line list in the LCD Telco menu will correspond to the installed module types and your configuration entries will vary accordingly Analog Phone Line Setup When POTS lines are used the only configuration required is to set the Country in the Telco menu This adjusts the Talkshow System for correct impedance loop current and signaling If your country is not listed try using CTR21 a European standard or USA ISDN Setup Determine which physical circuit will represent each line of the system It might help to fill out the chart below before proceeding Recall that each BRI has two voice channels each of
165. on the AT amp T Lucent 5ESS Telco switch We ve included some useful diagnostic aids in the system to help you determine where a problem is originating A good starting point is the ISDN status accessible on the LCD by pressing any menu button twice Condition System says Ready but attempted calls do not complete goes back to Ready or stays at Seize pending Are your SPIDs and Telco setting correct Can you dial from Line 1 to Line 2 If you can successfully call from one line to the other you know that your SPIDs and ISDN protocol settings for that line are correct and that the ISDN circuit itself is ok Try a call to another local number If you cannot complete this call try dialing with a 9 8 7 or 6 before the number Sometimes ISDN lines are set up as Centrex service even if you did not request this Is your long distance carrier OK If you can successfully call locally but long distance calls don t go through the Talkshow System usually displays Far end disconnect or No route available as the ISDN Cause phrase You may try another long distance carrier by dialing the appropriate 1OXXXXX pic code Examples of this dialing sequence in the USA amp Canada only are 1010288 1 area code phone number AT amp T 1010222 1 area code phone number MCI 1010333 1 area code phone number Sprint In the USA you can determine who your long distance carrier is by dialing 1 700 555 4141 If Intra LATA long d
166. onfiguration 25 3 1 Step by Step ce ee ee 25 3 2 Configuration for Telephone Lines o oaaae 25 Analog Phone Line Setup nonna 26 ISDNSCtUD ir dci ai e er ae wa a ee er A te 26 IstheISDN working o o oo o 27 3 3 Audio Inputand Output o o o e 28 Levels Levels Levels o o 28 Timetotestitoutl sois rca rr wna eas 29 WHatd0W 2ooocrsss rr bee bed ee ere ea ee es 30 User Operation 31 4 1 Desktop Director o ooo e e 31 Status SYMDORICONS 0 0 cd as a a date Be 31 Line Button Columns sordeia hk a Ge a k 33 Hold amp DrOp 222 Miaweke he had ae Rhadeke eden 33 A Wiese See wae EERE Sa EWS eo Bae ee ye g 33 Talent Mode eg sav aale tia Bh ELSE a ba e y 33 Block All sg ee ee Ved aod a ee a A Rol Cd a BB 34 Dialing Keypad soseri 2k ee e eucnakcatacbew aches 34 FunctionButtons 4 26 46 6 6 osc ee eee eee ee we 34 TELOS NX TALKSHOW SYSTEMS MANUAL v LCD Screen and Associated Buttons aaa ee 35 Dial Menusi s ri ea Oe a eke ea ae ew Oe a 35 Changing Between Producer and Talent Modes 37 Using the Director in Producer Mode o o o 37 Using the Director in Talent on air Mode 39 4 2 The Call Controller o oaaae 40 4 3 The Console Director o o o o o o o o o ooo o 40 Detailed Configuration amp Reference 41 FISES pes a A A Be aw aoe SO 4 5 V level Meteri
167. oop cur rent drops are not detected at all A particularly useful command is for ISDN logging This turns the Talkshow System into an ISDN protocol analyzer so you can see the communications between the Talkshow System and the Telco network gt gt Isdnlog lt return gt isdnlog Command isdnlog lt bitmask gt Description Specify ISDN logging mode bitmask b0 general logging bl Layer2 b3 layer3 b4 IIEs b5 raw packet b6 Telesoft stack Usually OxF is a good choice 5 13 System Signal Flow and Audio Processing ISDN Telco Analog Input Level Telco Output Level Send Audio Processing The Talkshow System s send to caller audio processing consists of the following functions Sample rate conversion High pass filter DETAILED CONFIGURATION amp REFERENCE 75 Frequency shifter Send AGC Limiter Sample Rate Conversion AES and Livewire sources are sample rate converted to the Talkshow System s internal sample rate of 8kHz All inputs are re clocked to the Talkshow System s internal clock derived from the ISDN clock if at least one ISDN BRI is present High Pass filter A high pass filter with a 300Hz break frequency improves hybrid performance and enhances intelligibility by removing unnecessary low frequencies from the input audio This function is always enabled Frequency Shifter A Telos innovation the frequency shifter pitch shifter inserts a small unnoticeable shift in f
168. or XXXYYYZZZZ0101 if requested Northern Pittsburgh Telephone XXXYYYZZZZ000 Pacific Bell XXXYYYZZZZ0101 or SPID1 XXXYYYZZZZ01 SPID2 XXXYYYZZZZ02 or 01YYYZZZZ000 or SPID1 XXXYYYZZZZ1 SPID2 XXXYYYZZZZ2 or XXXYYYZZZZ00 or XXXYYYZZZZ Quest See US West Rochester Telephone DIYYYZZZZO SBC See Pac Bell Ameritech or Southwestern Bell SNET Southern New England Telephone 01YYYZZZZ00 or 01YYYZZZZ000 or XXXYYYZZZZ0101 Southwestern Bell XXXYYYZZZZ0101 or XXXYYYZZZZ or 01YYYZZZZ000 or 01YYYZZZZ00 or XXXYYYZZZZO1 Sprint Centel XXXYYYZZZZ100 or XXXYYYZZZZ0000 or SPID1 XXXYYYZZZZ000 USA SPIDS BY TELCO 103 SPID2 XXXYYYZZZZ100 Standard Telephone GA SPID1 XXXYYYABCDCD SPID2 XXXYYYEFGHGH last two digits of phone number repeated US West XXXYYYZZZZ1111 or 01YYYZZZZ00 or 01YYYZZZZ000 Verizon See Bell Atlantic or GTE Vermont Telephone XXXYYYZZZZ0101 GUIDE TO TELEPHONE TERMINOLOGY 105 Telephone Terminology Guide You ll get better results from the Telco if you understand and speak the lingo We have tried to include the typical acro You ll get better results from the Telco if you understand and speak the lingo We have tried to include the typical acronyms used by Telco personnel We ve put the definition under the most commonly used acronyms AMI Alternate Mark Inversion A T1 line coding method This is the older of the two commonly available Mo
169. or your country 5 6 The System Menu LCD Panel and Web page options This menu consists of items that affect basic system operation TCP IP Setup The Talkshow System and PC hosting the Web browser should have IP addresses in the same range the first 3 of the four numbers of the IP address should match and should have the same subnet mask Host Name Web only The IP network text name The default is Nx6 or Nx12 depending on which system you purchased 58 Section 5 IP Address This is the IP address for this Talkshow System An IP address is a series of 4 numbers sepa rated by periods This is a numeric only field enter the digits as described above You should enter no leading zero in each of the four numbers if you are given 192 168 000 012 enter 192 168 0 12 Subnet This is your subnet mask The subnet mask is set to the size of your local network All packets addressed to a destination outside this local area are sent to the gateway router The subnet mask is a series of 4 numbers separated by periods in the same format as the IP address A subnet with 256 addresses would have this mask value 255 255 255 0 Gateway This is the IP address of a gateway router connecting your LAN to the Internet You will need to get this from your network administrator or Internet service provider To enter IP addresses using the LCD button interface Select the TCP IP Settings sub menu To enter an IP address use the
170. orrect SPIDs will be indicated by the line status indication Wait on the Talkshow System ISDN status screen Often the Telco will leave the last 00 or 01 off the end of your SPID when giving it to you For the latest list check Telos Systems web page A standardized SPID format for national ISDN of XXXYYYZZZZ0101 was phased in during 1996 and is used by most of the Telcos for new lines In the list below XXX Area code YY Y exchange ZZZZ phone number Alltel Ohio XXXYYYZZZZ0101 Ameritech XXXYYYZZZZ0111 or XXXYYYZZZZ011 or DIYYYZZZZO11 or XXXYYYZZZZO1 Bell Atlantic 01YYYZZZZ00 or 01YYYZZZZ000 or 01XXXYYYZZZZ000 or XXXYYYZZZZ100 Bell Atlantic North Former NYNEX XXXYYYZZZZ0101 or XXXYYYZZZZ0000 Bell Canada XXXYYYZZZZ00 or XXXYYYZZZZ0101 BellSouth XXXYYYZZZZ0101 or XXXYYYZZZZ0100 or XXXYYYZZZZ01 or O1YYYZZZZO or 01YYYZZZZ000 102 Appendix 1 Century Telephone XXXYYYZZZZ00 Cincinnati Bell 01YYYZZZZ000 or SPID1 00YYYZZZZ01 SPID2 00YYYZZZZ02 or 00YYYZZZZO1 Fort Mills Telephone XXXYYYZZZZ Frontier Communications 01YYYZZZZ000 GCI Alaska XXXYYYZZZZ00 GTE XXXYYYZZZZ0101 or XXXYYYZZZZ0001 or 01YYYZZZZ000 or XXXYYYZZZZ00 or 01YYYZZZZO0 or XXXXYYYZZZZ or SPID1 XXXYYYZZZZ00 SPID2 XXXYYYZZZZ01 or SPID1 XXXYYYZZZZ0000 SPID2 XXXYYYZZZZ0001 or SPID1 XXXYYYZZZZ0000 SPID2 XXXYYYZZZZ0101 Nextlink XO XXXYYYZZZZ or XXXYYYZZZZ1
171. ort an all trunks busy or far end busy conditions It also decreased toll fraud sub stantially since it removed the potential for access to the signaling information that was inherent to in band signaling schemes CCIS also enables CLASS features as well as sophisticated re routing features for intelligent network applications See also in band signaling See also SS7 Central Office See CO Centrex Central Exchange Service An enhanced business tele phone service intended to offer most of the features of a PBX but where the lines are all from the LEC out of a public switch Offers CLASS like features for business users such as 4 digit inside dial ing hold transfer attendant etc CEPT Conference on European Posts amp Telecommunications This is a European standards body that formerly set the standards for telephone interfaces for 26 countries CEPT Format The usual rate and frame format for E1 circuits 2 048 mbps See El CEPT Rate See CEPT format See also E1 Channel An actual path you can talk or send data over This is what you are paying the phone company for For instance ISDN BRI lines can be ordered with 1 or 2 active channels and these channels can be configured for voice calls CSV data calls CSD or both alternate CSD CSV A channel does not necessarily have it s own unique telephone number See ISDN Channel Associated Signaling See CAS See also Robbed Bit Signaling Choke Ex
172. ot al lowed to sell CPE under the terms of the break up of AT amp T With the current state of deregulation the RBOCs are slowly re entering this business Interexchange Carrier See IEC Interwork The ability of two different type of networks to com municate seamlessly For example ISDN can interwork calls to both the POTS network and the Switched 56 network Inverse multiplexing A method of breaking apart and later com bining 2 data streams that does not involve the telephone network Inverse multiplexing is the most common way of achieving this function in Codecs operating at 112 or 128kbps over ISDN IOC Capability Packages ISDN Ordering Code system This system was devised by the National ISDN User s Forum and Bellcore to simplify ordering new ISDN lines in the USA and Canada Using a single code specifies all line specifications The Telos Zephyr ZephyrExpress TWO and TWOx12 support IOC package S Search for document NIUF 428 94 at NIUF s web page http www niuf nist gov for more information ISDN Integrated Services Digital Network A relatively new and highly flexible type of telephone service which allows dialing on digital channels with multiple bi directional Bearer channels each with a capacity of 56 or 64 Kbps and a single bi directional D channel See BRI and PRI ISDN Protocol The language used for communication between the Telco s switch and the customer s Terminal Adapter
173. ounced peaks in this response characteristic flattening out those peaks with an equalizer can signifi cantly improve the feedback margin A spectrum analyzer connected to the output of your mic processor will help detect the frequency of those peaks If necessary soften acoustic room reflections by adding curtains or wall treatments The discussion above is intended to help in those situations where open speakers are a require ment Whenever possible of course it is best to use headphones to listen to the caller audio We have found the best scheme is to have the phone monitor speaker mute when the mic is turned on If your mix minus has provided a means for the mic signal to be active even when the mic channel is turned off the system can be used like a speaker phone when the taking calls off air When the call is on the air the speaker mutes however the talent will generally be wearing headphones when on mic in any case In addition to muting the monitor speakers when the mic is turned on any Desktop Directors should have their ringers muted as well The Nx12 Nx6 has one GPIO input per studio that can be used to disable all Desktop Directors ringers whenever the monitor speakers are muted This is also the default behavior when the Nx12 is used with an Axia Element console system Individual Desktop Directors ringers can be re enabled via their menus and ignore the Ring Mute signaling See sections 2 10 Ring Mute In and 4 1
174. p point 24dBu Bridging impedance 10k ohms 16 Section 2 Analog Input XLR PIN DESCRIPTION 1 Ground 2 Audio 3 Audio The inputs are designed to be sourced from balanced line level signals Usually shielded cables have the shield wire connected only on one end to prevent ground loops Older equipment with a transformer output stage may need a terminating resistor across pins 2 and 3 consult the manual for your equipment for how to use it with high impedance inputs When using unbalanced sources we recommend using pin 2 and 3 to connect the signal and ground respectively Usually pin 1 is not connected in order to avoid ground loops The sensitivity of the send inputs are adjusted using Nominal Input Level in the Audio menu The Input meter indicates the level after the adjustment and before the AGC See Sections 3 3 5 1 and 5 4 for additional information on level adjustment Note The Telos Talkshow System uses the currently favored pin outs for three pin XLRs You can easily remember the correct signals when wiring connectors by using the phrase George Washington Bridge Pin 1 G Ground Pin 2 W White typical color in mic cable if there is no white there will be a red conductor and Pin 3 B Black AES Digital Inputs These are inputs for AES EBU format digital audio signals and are only present if the unit was purchased with the AES option or ifthe AES card has since been installed in
175. plug in interface modules to attach to POTS regular analog loop start telephone lines The Country setting in the Telco menu configures the characteristics of the interfaces for your country If you do not see your country listed you should use CTR21 or USA These modules have six modular jacks The type of jack will be a six position four pin RJ 11 style The connection is to the center two pins of this jack Note Unlike our older products Nx Talkshow Systems do not have the loop through provision for POTS lines The POTS lines used with the Talkshow System should be dedicated The yellow and black leads pins two and five of the modular connectors should be disconnected and insulated IMPORTANT When we say POTS lines we mean just that plain old analog loop start telephone lines A rule of thumb if a line works with an analog modem or fax machine it will work with Telos Talkshow Systems Analog Ground Start lines can be used for incoming calls only However you will need to set Loop Check in the Telco menu to No for these lines to be recognized properly Be careful not to connect Talkshow Systems to PBX ports intended for proprietary telephones These sometimes have voltages that could damage the units 2 4 ISDN Technology 101 Since ISDN will be new to some users we give you an introduction to the technology here If you are using analog telephone lines POTS or already know ISDN basics you can skip on to the next section
176. port These are numbered left to right as viewed form the rear of the unit with two channels per port Each channel is assigned a Desktop Director number DD notification Thus by looking at the Director number on its LCD upper left corner it possible to find out exactly which port this one is connected to One of two messages will be displayed Activated The Desktop Director is up and running on this channel Deactivated No Desktop Director is connected to this channel or if it is it is not being recognized by the Talkshow System Whenever a Desktop Director is activated an Errors number is indicated This corresponds to the number of data packets lost since this Desktop Director was powered on An occasional error is not cause for alarm however if you see multiple errors accumulating over time this should be investigated Possible causes are bad wiring a bad port on the Talkshow System or a defective Desktop Director Note If a Desktop Director Is connected to an external power supply unplugging the connection at the back of the Talkshow System will not reset the error counter A few errors 2 3 when pow ering up the Talkshow System is normal If a Desktop Director Is frozen the Talkshow System will deactivate it If this ever happens unplug the Desktop Director then plug it back in On the Talkshow System LCD it is possible to investigate further by pressing the OK button This will give access to a detailed Dia
177. provided If you do not know you may be able to guess Guessing the ISDN Protocol setting in Canada amp the USA If you were given only a single phone number and no SPIDs try AT amp T Cust If you were given 2 SPIDs with the format 01 7 digit phone 0 try AT amp T Cust If you were given 2 SPIDs with the format 01 7 digit phone 000 or 01 7 digits 011 try Natl I 1 If you were given 2 SPIDs with the format area code 7 digit phone 4 digits for example 0101 0100 0111 or 1111 try Natl 1 1 If that fails Try DMS Cust If you were given 2 SPIDs with the format area code 7 digit phone 1 digit for example 1 or 0 try DMS Cust If this fails try Natl I 1 If you were given 2 SPIDs with the format area code 7digit phone 2 digits for example 01 11 02 or 00 try DMS Cust If that fails try Natl I 1 If you were given 2 SPIDs with the format area code 7digit phone 3 digits for example 000 001 try Natl I 1 If that fails try DMS Cust If you were given a single phone number and a single SPID with the format of 01 7 digit phone 0 try AT amp T Cust Do not enter this SPID If you were given a single phone number and a single SPID with the format of 01 7 digit phone 00 try Natl I 1 If that fails try AT amp T Cust 54 Section 5 If you were given a single phone number and a single SPID with the format of area code 7digit phone 4digits for example 0101 0100 0111 or 1111 Try Natl 1 1 I
178. r s equipment and the Telco switch Used to carry data to set up calls and receive calls Some Telco s also allow users to use the D channel to access the packet data network with appropriate terminal equipment D4 See Superframe See also Line Format DCE Data Communication Equipment When using serial com munications such RS 232 V 35 or X 21 the DCE is the device sending receiving from the Telco line i e a modem or CSU DSU In contrast to DTE DDS Digital Data System See Dedicated Digital Service Dedicated Circuit A permanent channel between two locations As opposed to a Switched Circuit Dedicated Digital Service a Hardwired or Nailed Up digital circuit which is permanently connected between 2 points Typically 56Kbps or 64Kbps Dedicated digital lines are frequently cheaper than ISDN for full time service Also called Digital Data System or DDS DID Direct Inward Dialing The ability for an outside caller to dial to a PBX extension without going through an attendant or auto attendant See also DID Number and DID Trunk DID Extension or DID station A specific phone within a PBX which can be called from the public telephone network without go ing through an attendant or auto attendant DID Number A phone number used to route calls from the telephone network to a specific phone in a PBX the DID exten sion DID requires special DID trunks or ISDN PRI two way DID trunks Blocks of DID numb
179. r determined user busy it is noted that the user equipment is compatible with the call Cause No 18 No far end response No user responding This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within the prescribed period of time allocated before timer T303 or T310 has expired Cause No 19 No answer No answer from user user alerted This cause is used when the called user has provided an alerting indication but not a connect indication within a prescribed period of time before timer T301 has expired Cause No 20 Subscriber absent This cause is used when a mobile station has logged off radio contact is not obtained with a mobile station or a personal telecommunications user is temporarily not addressable at any user network interface Cause No 21 Call rejected This cause indicates that the equipment sending this cause does not wish to accept this call although it could have accepted the call because the equipment sending this cause is neither busy nor incompatible This cause may also be generated by the network indicating that the call was cleared due to a supplementary service constraint Cause No 22 Number changed This cause is returned to a calling party when the called party number indicated by the calling 88 Section 6 party is no longer assigned The new called party number may optionally be included in the
180. r generally means AT amp T PTP protocol You must not have SPIDs entered if using this protocol In Europe you have either ETS300 EuroISDN or the local protocol such as ITR6 in Germany or VN2 in France The Talkshow System only works with ETS300 In the case of Natl I 1 or 2 and DMS 100 carefully verify that you have the correct SPID from the Telco and that you have properly entered it into the system It must be correct in order for anything to work We ve seen a few cases where it was given incorrectly by Telco personnel so it might be a good idea to re check with them if things are pointing in this direction TECHNICAL DATA amp TROUBLESHOOTING 83 Hot Tip If you suspect the SPIDs given you are wrong or the basic line provisioning configuration is incorrect call the Telco and ask to speak to someone who can log into the switch They will probably need to have someone call you back Only once you have this person on the line and they are logged into the switch ask them to look at the SPIDs programmed there Also get this person s fax number and send them the ISDN ordering information from the Appendix of this manual Sometimes the information in the business office computer is wrong By verifying what is programmed into the actual switch you can save time The AT amp T PTP protocol generally does not allow two simultaneous voice calls For this reason we recommend that the National ISDN 1 protocol be used if you are
181. r on air you can press a left or right column line button Answering calls with the handset Pick up the handset and select a line using the left column or press Next to select the line which has been ringing longest Answering calls using the speakerphone Press the Speakerphone button and select a line using the left column or press Next to select the line which has been ringing longest 4 2 The Call Controller The Call Controller together with a user provided telephone emulates a Desktop Director See the Call Controller manual for additional information Because there is no LCD the dialing keypad is used to change modes as follows PRODUCER 77638237 changes to Producer mode TALENT 825368 changes to Talent mode DD 33 is current ID for this unit SHOW1 74691 connects this controller to Show 1 SHOW2 74692 connects this controller to Show 2 4 3 The Console Director Telos provides sub assemblies to a number of mixing console manufacturers allowing them to offer you a console mounted control surface compatible with the Talkshow System 2x12 and Series 2101 systems The Console Director works much like the Desktop Director in Talent mode Detailed Configuration amp Reference First Step After power up and initialization the status screen appears When you see this screen and if ISDN or POTS lines are connected and functioning correctly and the Talkshow System s basic configuration settings hav
182. rectory Number When a number of Telco chan nels share the same hunt group it is customary to give out only one phone number for the group although generally each channel will have its own number The number given out is the Listed Direc tory Number since that is the number that would be listed in the Telephone Directory and given to customers Sometimes called a Pilot Number See also DN and Hunt Group LE Local Exchange European term for Central Office See CO LEC Local Exchange Carrier Your local telephone service pro vider which is either an RBOC or an Independent In other words a traditional phone company In contrast to CLEC or IEC Line An electrical connection between a telephone service pro vider s switch LEC or CLEC and a telephone terminal or Key system An electrical connection between a telephone service pro vider s switch and another switch is technically called a trunk Note that some type of physical lines offer more than one channel LE a BRI circuit has 2 channels called B channels This term is a confusing one so we try to avoid using it See Channel See also Station Line Line card The circuit in the Telco switch to which your line is con nected On an ISDN circuit the line card performs a role analogous to the NT1 in adapting to and equalizing the circuit to establish OSI Layer 1 Line Coding T1 The clock signal for T1 is derived at the far end from the data bits themselves Therefore
183. requency to the send audio to prevent feedback buildup when the system is used with open speakers Send Automatic Gain Control Limiter This AGC helps maintain consistent audio levels to the caller This function is always enabled This benefits from a carefully crafted algorithm utilizing a feed forward topology with near in audible artifacts below the limit threshold At moderate levels it is AGC like while at high peak levels it is limiter like In addition to making levels more consistent to callers this performs the protection limiting required to meet Telecom regulations Receive Caller Audio Processing The Talkshow System s receive audio processing consists of the following functions High pass hum filter Automatic Gain Control Platform leveler Noise gate Dynamic equalization Sample rate conversion High pass hum Filter This filter removes hum and other unwanted low frequency noise from the caller audio This filter has a break frequency of 100 Hz Receive Automatic Gain Control This AGC serves to improve consistency of caller audio levels This function is very important to effective hybrid performance as caller levels can vary by as much as 30 dB The Talkshow System offers a new approach to AGC to accommodate the fact that loud callers have become just as great a problem as low level callers The AGC is reset for each caller to a neutral position to most rapidly accommodate both low and high le
184. rratic performance is experienced The quickest test is tobring up only the hybrid in question on the board and select a line Dial tone should not appear on the send meter of the hybrid in question Using a modern broadcast console s mix minus capability Most modern broadcast consoles have provision for multiple mix minus busses The best allow selective feeds to the phone system This is useful since sometimes you want only one micro phone feeding the phone but sometimes you want three or four mics during the morning show for instance and sometimes you want to play some audio piece that callers need to hear and react to such as contest sound effects etc Some even provide for separate on air and off line recording telephone modes When multiple hybrids are used each caller will need a separate mix minus that includes the other hybrids so that callers can hear one another Hot Tip The Axia Element mixing console was designed with today s extensive use of phones and remote codecs in mind It has all the features described above It can have as many mixmi nuses as faders It offers an integrated phone controller module that works smoothly with the Talkshow System over Ethernet It has a one button Record function that does everything needed to record an announcer phone call off line Because it uses Livewire networking only one RJ 45 is needed to connect the Talkshow System s four hybrids input and output two program on
185. s see the section on the Next button below Hold amp Drop DIE Talent Mode These affect only the corresponding line button column above them A locked line will not be affected All unlocked lines in that column will be dropped Producer Mode The left hold button is normal hold while the right hold button places a caller on screened hold For more details on how screened hold works see the section on the Next button below The right hold button may be replaced by with a keycap labeled with a check mark if a Desktop Director will be used primarily for screening Next EJ Talent Mode This button will cause the call designated by the call screener as next to be taken on line column 1 Any unlocked caller will be dropped It is possible to configure the system to put the caller back hold instead see Direct Hold option Section 5 6 A producer working as a call screener can place a call on screened hold and the check mark icon 34 Section 4 will appear for the line This indicates that the caller has been checked by the screener and is ready to go on air Calls placed on screened hold in this manner are being queued up from the first selected to the last one selected by the screener The next caller line icon is represented by a blinking check mark If there is no screener to select and order calls the line which has been ringing the longest will be selected instead If no line is ringing the line on hold
186. s by pressing Drop or you can place all unlocked calls on hold or screened hold by pressing the appropriate Hold button USER OPERATION 39 Using the Director in Talent on air Mode The first text line on the LCD tells you which mode you are in You will see Talent if you are in this operation mode Answering calls Each line has two buttons the left column for Caller 1 your console fader might be labeled Phone 1 or something else and right column for Caller 2 Pressing a line button places the caller on that line on air on the corresponding fader When you select the line the Status Symbol icon will change to an arrow pointing to the line column you used Note In some cases when an Talkshow System system is shared between two studios only the left column is used Pressing a line button when that line is already on air locks the caller You can use this capability to prevent accidentally losing a guest To unlock press that line button a third time If the line has already been screened the Screened symbol a check mark will be displayed If the caller has not been screened the hold symbol or ringing symbol will be displayed Drop and Hold A call may be dropped or placed on hold by using the buttons below the line columns Also if you select another line on the same column any other active non locked lines will be automati cally dropped Next This button will take the call designated as next on the left li
187. s not match with the Nx Talkshow system s GPIO logic The consoles PHONE source definition is implemented for hybrids that do not com municate via the Livewire Protocol and require hardware connections to control the hybrids If you use the Nx Talkshow system with an Axia console you likely will be integrating with the console s Phone module The communication between the console and Nx Talkshow system uses the same protocol used by the call screening applications Assistant Producer or VX Pro ducer It is recommended to disable the logic control option within the source profile to prevent adverse behavior of the Axia console or the Nx hybrids See the console s manual for details on source configuration options regarding disabling the Logic control The function of the Nx Talkshow GPIO inputs and outputs for the shows are noted here GPIO Show 1 port 1 vGPO Port Bit 1 1 1 2 1 3 1 4 1 5 vGPI Port Bit 1 1 1 2 1 3 1 4 1 5 GPIO Show 2 port 2 vGPO Port Bit 2 1 2 2 2 3 2 4 2 5 vGPI Port Bit 2 1 2 2 2 3 2 4 2 5 Control IN Ring Mute IN 1 Control IN User Out Dump 1 Ringing Out 1 Priority Ringing Out 1 Record Start 1 Record Stop 1 Control IN Ring Mute IN 2 Control IN User Out Dump 2 Ringing Out 2 Priority Ringing Out 2 Record Start 2 Record Stop 2 DETAILED CONFIGURATION amp REFERENCE Hardware Mapped Pin 11 12 13 14 15 Hardware Mapped Pin 1 Hardware Mapped Pin 11 1
188. sage type it does not recognize either because this is a message not defined or defined and not implemented by the equipment sending this cause Cause No 98 Message not compatible This cause indicates that the message received is not compatible with the call state or the mes sage type is non existent or not implemented Cause No 99 Info non existent or not implemented call processed This cause is sent when the equipment sending this cause has received a message which includes 92 Section 6 the information elements not recognized because the information element identifier is not define or it is defined but not implemented by the equipment sending the cause However the information element is not required for the equipment sending the cause to process the message Cause No 100 Invalid information element contents This cause is returned when the equipment sending this cause has received an information ele ment that it has implemented however one or more field of the information element are coded in such as way e g truncated invalid extension bit invalid field values etc that has not been implemented by the equipment sending this cause Cause No 101 Message not compatible with call state Protocol error threshold This cause is returned when the equipment sending this cause has received a message that procedures indicate is not a permissible message to receive at this time or This cause indicates that the swit
189. se interference If it does the user will be required to eliminate the interference at the user s expense NOTE Objection able interference to TV or radio reception can occur if other devices are connected to this device without the use of shielded interconnect cables FCC rules require the use of shielded cables CANADA WARNING This digital apparatus does not exceed the Class A limits for radio noise emissions set out in the Radio Interference Regulations of the Canadian Department of Communications Le present appareil numerique nemet pas de bruits radioelectriques depassant les limites applicables aux appareils numeriques de Class A prescrites dans le reglement sur le brouillage radioelectrique edicte par le ministere des Communications du Canada TELOS NX TALKSHOW SYSTEMS MANUAL iii Table of Contents W support YOU ia 00 A ee he A a ae i A letter from the President of Telos 2 2 2 2 eee eee ix 1 Introduction 1 Welcome to the Nx Talkshow Systems 2 ee ee 1 Features of the Nx Talkshow Systems o o o o o o ooo 1 Why ISDN Makes Sense for Broadcast Talk Show Systems 2 Sharing Phone Lines s o o o o o 3 A Step up in Quality and Flexibility o o o 4 2 Installation 5 Z1 Getting Stated is a o a la ee ee S 5 2 2 Connections to the Telco circuits o o o o o o ooo 5 2 3 Connecting the Talkshow System toAnalogLines 6 2
190. softkey NOTE The default password is 1234 Press lt V gt repeatedly to highlight Set Priority Lines and then press lt SELECT gt Use lt V gt and lt A gt to highlight each line For each line use the lt gt or lt gt Softkeys to select from the following options Normal Line is included in the block all group Priority Line is excluded from the block all group If this line is ringing the Priority Ring Output pin 2 on the parallel control port will go true Press the SELECT button to accept your changes Note You must reboot the system after configuring this setting Be sure to attach the phone lines before rebooting To change the password Pe ON oS Press the MENU softkey Press lt V gt repeatedly until Configure Shows is highlighted and then press lt SELECT gt Enter the password and then press the OK softkey Note The default password is 1234 Press lt V gt repeatedly to highlight Change Password and then press lt SELECT gt Enter a four digit password Press each digit carefully Then press the OK softkey If you are uncertain if you have entered the desired password press the BACK softkey instead of OK Press the BACK softkey repeatedly until back to the main menu 66 Section 5 Selecting a Show on the Desktop Director and other controllers In the two studio modes each Desktop Director or other controller must select a Show Selecting a Show using a Desktop Director To do so fo
191. split as follows Show 1 Hybrid A amp B has access to lines 1 through 3 Lines one and two are normal lines while Line three is a priority line Line three is not affected by the Block AIP function Show 2 Hybrid C amp D has access to lines four through six Lines four and five are normal lines while Line six is a priority line Line six is not affected by the Block Al function In any of the two studio modes the system maintains two separate Next and Screened Next queues The two studios can be operated completely independently 2 Show Shared This option causes the system to be split as follows Show 1 Hybrid A amp B has access to lines one through three Lines one and two are normal lines while Line three is a priority line Line three is not affected by the Block Al function Show 2 Hybrid C amp D also has access to lines one through six When a line is shared between two studios it becomes unavailable in one studio if it is in use in the other studio In this case the Status Symbol for this line will show a small x on the other studio s Director If a shared line is placed on hold it can be picked up in either studio Users of the Telos Series 2101 system will recognize this behavior as that system works in the same way 2 Show Custom Note that the easiest way to configure the line selection is with the setup wizard web page rather than through the
192. ssing 74 Receive Caller Audio Processing o o o o ee ee 75 Technical Data amp Troubleshooting 79 GA Overview 6 oc aena a Oe ee A Wark eal 79 6 2 System Software amp Firmware 2 ee ee 79 Viewing VersionInformation o ee 79 6 3 General Troubleshooting o oo o 79 Thinking about Problem Solving o o o o o 79 6 4 Troubleshooting ISDN Problems 81 Status Screen for ISDN troubleshooting 81 Condition Can t Get ISDN Ready Indication 82 Condition System says Ready but attempted calls do not complete 83 Condition Intermittent ISDN Problems 84 6 5 Troubleshooting Desktop Director Problems 85 6 6 Replaceable Modules and Hardware Servicing 85 6 7 ISDN Cause Phrases Values o o o o o o o o o oo 86 CLASS 1 1 normal 86 CLASS 1 2 resource unavailable 88 CLASS 1 3 service or option not available 89 CLASS 1 4 service or option not implemented 9 CLASS 1 5 invalid message e g parameter out ofrange 90 CLASS 1 6 protocol error e g unknown Message 91 CLASS 1 7 interworking class 2 ee 92 Specifications 95 General sce ee aot a a ee oe ee eee 95 Processing Functions lt s oo o o
193. st an indepen dent albeit one formed after the divestiture of AT amp T See LEC and Independent CLI Calling Line Identity European term for CLID See CLID CLID Calling Line Identification This is the ISDN and SS7 equivalent of Caller ID I E the number of the calling party See also Caller ID and ANI CO Central Office The Telco facility to which your local tele phone circuit lead Contains Switches and Trunks as well as the GUIDE TO TELEPHONE TERMINOLOGY 107 local telephone circuits Codec COder DECoder A device which takes digitized audio and codes it in order to reduce the transmission bit rate and which can also simultaneously decode such coded audio Strictly speak ing a codec does not include an ISDN terminal adapter and related equipment Simple codecs are also used in digital telephony These use a simple companding scheme to reduce channel noise COL COnnected Line number European Term The number to which you have connected This may not be the number you dialed if call forwarding is used Combination Trunk A trunk channel which can both make and receive calls This generally refers to analog ground start or loop start trunks although the term can be applied to ISDN BRI or PRI channels as well Each combination trunk normally has a telephone number although they are frequently part of a hunt group and only one number may be published for that group Also called a Both Way
194. stem periodically checks each analog line port for loop current When current is detected the single green dot icon appears on Desktop Directors and the line is ready for use This process is called Loop Check Yes is the default When you are using ground start lines this should be set to Off Note The Talkshow System does not support outbound dialing on ground start lines However setting Loop Check to No will allow incoming calls to work normally Note Ground Start analog lines are not fully supported by the Talkshow System Only Incoming calls are allowed You can take advantage of this to get a feature that might be sometimes useful preventing outgoing calls Enable Auto Call Drop Web only Calls can be automatically disconnected when the far end hangs up Most USA style Telco lines provide a momentary loop drop for this purpose but non USA lines and most PBXs provide a tone signal Call Progress Tone Disconnect instead The Talkshow System can respond to one both or neither of these signals The type of tone signal detected for CPTD is determined by the POTS Country setting Ring Timeout web only The time to wait after an incoming POTS ring signal before the end of the ringing state is detected and line is considered idle The default setting follows the selection of the POTS Country option but can be overridden with range of 4 to 11 seconds We strongly recommend leaving this as Default to use the default value f
195. swork out of figuring out cabling to the unit and to facilitate an easy installation Routing details explanations and diagrams depicting the various Mix Minus routing modes are outlined in the Audio Section of the manual see section 5 4 Settings will be saved when you press APPLY DETAILED CONFIGURATION amp REFERENCE 63 Audio Connections TELCO INTERFACE A TELCO INTERFACE B POH INPUT Gvewire q ANALOG mor mo aa Shee 4 ae 104100 10100 Livewire mmm 82 O 0 6 8 a D 1 a POH 5 7 Phone Book Menu Web and Desktop Director options Here you can enter up to 10 frequently dialed numbers for each Studio or show These numbers can be selected on the Desktop Director through the menu options Studio 1 relates to Show 1 Studio 2 relates to Show 2 If you are running in One Show mode then only the entries for Studio 1 will be accessible from the Desktop Director Press APPLY to save the option Note that one can enter and edit these entries thru the Desktop Director menu as well 5 8 Show Modes Recall that the options for Show Modes are One Show 4Column 2 Show Split 2 Show Shared 2 Show Custom Note It is not necessary to reboot after configuring this setting Single studio Modes One Show default Four hybrids are used to improve conferencing quality but they are summed and mix minused internally to appear as if they were two Or one depending upon Mix Minus mode Each Desktop Director
196. t O gt lt W gt lt 2 gt or 74692 Connecting Assistant Producer to a Show IMPORTANT You must use Assistant Producer 3 5 or newer If you are an Assistant Producer 2 0 user contact Telos Customer Support for information on upgrading 1 From the AP Server menu choose a Server the IP address of a Talkshow System 2 From the AP Connection menu choose Connect Show and from there choose from the desired show Show 1 or Show 2 3 From the AP Connection menu choose Desktop Director and choose the Desktop Director you wish to control NOTE A Desktop Director s ID number is displayed on the upper left of the display For example DD2 To obtain the ID number of a Call Controller type lt gt lt d gt lt d gt and it will be displayed on the screen For additional information see the AP 3 5 User s Manual DETAILED CONFIGURATION 8 REFERENCE 67 5 9 Livewire The Web interface is used for Livewire configuration The pages are similar to those for Livewire Nodes so if you are already familiar with those you will be immediately comfortable with the Talkshow System s Livewire configuration interface Consult Introduction to Livewire available on the axiaaudio com website for a description of Livewire and an explanation of the various terms and options Note that the terms Sources and Destinations are from the perspective of the Livewire network and have the same usage as for Livewire Nodes Hybrid Channels
197. t could have stopped working It is impossible for Talkshow Systems to distin guish between a deactivated circuit and a malfunctioning circuit until you try to use it For this reason if you are using ETS 300 and your phone system is mission critical you should ask your telephone company to disable this function Ordering Help Appendix 3 to this manual describes the ISDN ordering process for USA Canada in detail and Appendix 4 does the same for ETS 300 Euro ISDN Both include ready to go forms that can be sent directly to your telephone company 2 5 Connecting Telos Talkshow Systems to ISDN Talkshow Systems use plug in modules to attach to ISDN These ISDN modules have three modular jacks The type of jack will be either a 6 position RJ 11 style or an eight position RJ 45 style depending upon whether it is the version for USA amp Canada or the rest of the world The correspondence between the interface slot to the system line number for is from left to right when viewing the system from the rear Therefore the leftmost jack in slot A would be lines one and two and the rightmost jack of an ISDN card in slot B would be lines 11 amp 12 All ISDN BRI circuits must use the same ISDN protocol setting ISDN in USA amp Canada Using the U Interface Connect the ISDN line from the telephone central office directly to the RJ 11 style U interface modular jack on the rear panel Each module has 3 jacks and each jack has an associated LED
198. t or materials or from any other cause In order to invoke this Warranty notice of a warranty claim must be received by Telos Systems within the above stated warranty period and warranty coverage must be authorized by Telos Systems If Telos Systems authorizes the performance of warranty service the defective 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 Systems authorized dealers are not authorized to assume for Telos Systems any additional obligations or liabilities in connection with the dealers sale of the Products Telos products are to be used with registered protective interface devices that satisfy regulatory requirements in their country of use USA SPIDS BY TELCO 101 1 USA SPIDs by Telco Here is a list of known working SPIDs Service Profile IDs by telephone company compiled from our direct experience helping clients with ISDN installations Your SPIDs may be different of course The SPID is distinct from your telephone number and does not necessarily contain your area code or telephone number although this is generally the case Inc
199. te dump Upon clicking any of the menu links you will be prompted to enter a user name and password The default login user name is user and the password is lt blank gt The various listed menus provide access to the full set of configuration options a superset of those available on the front panel The ISDN SPID and DN entries are only available on the Web interface It s not very convenient to enter numeric data on the LCD button interface so we thought youd appreciate having a PC s keyboard for that The Setup Wizard and Livewire menus are only available through the Web interface 5 4 The Audio Menu LCD panel and Web page options This group consists of settings that control the audio levels processing options mixing etc Disable Livewire Allows user to disable the output Livewire streams of the Nx box This option does not en able the streams but only disables them One must log into the web interface to re enable the streams Selecting this option will prompt you if want to disable them or not 46 Section 5 Nominal Input Level Adjusts to the level expected from the mixing console This adjustment affects all of the input options analog AES and Livewire It is applied before the send AGC Be careful to avoid clipping of the Talkshow System inputs Clipped send audio will create a problem for the system and you will probably experience poor hybrid performance Increasing the input level past the correct
200. telephone audio as follows Off The caller audio is passed without modification Fixed This is a simple manual equalizer mode Adaptive Digital Dynamic This is a three band dynamic equalizer We ve chosen frequency breakpoints time constants and other characteristics to optimize the tonal quality of varied telephone callers The user entered EQ values are used as target levels This allows you to customize the caller spectral characteristics but still get a consistent spectral characteristic from caller to caller Ducking System The ducking function occurs primarily on the caller audio This function is important for several reasons Provides an aesthetic control over the caller that many programmers prefer When the announcer speaks the caller is ducked or reduced in volume dynamically This is particu larly useful when the caller is carrying on and the talent wants to interrupt to move the show along Reduces feedback when a loudspeaker is necessary In the loudspeaker case reduces audibility of the caller signal that couples through the acoustic path into the microphones and is returned the caller resulting in a more natural sound for the caller The duplex system inserts a controlled loss ducking into whichever audio path send or receive is not active at the moment When the caller is speaking this loss is inserted in the announcer path when the announcer is speaking the loss is inserted i
201. ter the correct Service Profile ID SPID numbers for each ISDN channel normally two per physical line The telephone company should tell you this number when they confirm your order Often it is found written on the line jack or punch block Usually the SPID is the area code phone number 0101 However we have seen some that include a two digit prefix and a two digit suffix and many other variations are possible See Ap pendix 1 for a list of known working SPIDs If this works consider yourself lucky if it doesn t work there is no substitute for getting the correct SPID from your Telco DETAILED CONFIGURATION amp REFERENCE 55 ISDN Tip ETS 300 users should disregard all references to SPIDs Euro ISDN does not have SPIDs you lucky souls IMPORTANT If your line status shows Wait at least one of your SPIDs is incorrect or your ISDN Protocol setting is incorrect In this case check the SPID For a list of known working SPIDs by tele phone company see the Appendix Notes on SPIDs 1 Incorrectly entered SPIDs are the most common problem USA users have placing ISDN calls Compare the SPIDs you have entered with the SPIDs provided to you by the phone company to be certain that you have entered them correctly Do not add anything to your SPIDs 2 Note that there are no hyphens dashes or spaces in SPIDs so the system will not allow you to enter any non numeric characters If your installer has included them in the
202. termi nal signals the off hook condition by allowing DC current to flow See Ground Start Trunk See also Glare Loop Start Trunk A plain old telephone line connected to a PBX switch See Loop Start Line The PBX signals the off hook condition by allowing DC current to flow Ground Start Trunks are generally preferred for use on PBXs to prevent glare See Ground Start Trunk See also Glare The Telco may call this a Loop Start Line LOS Loss Of Signal An LED or other indicator that illuminates if a signal is absent This terminology is commonly used with T 1 equipment LT Line Termination The electrical and protocol specifications for the Central Office end of an ISDN line If you wish to connect an ISDN terminal such as a Zephyr Xstream to a PBX the PBX must support LT ISDN See also NT and Line Card Lucent Technologies Company which now makes the former AT amp T 5ESS switch as well as various other piece of Telco gear and semiconductors Lucent was split off from AT amp T in 1996 As of approximately 1999 Lucent sold their telephone set manufacturing business and the right to use the AT amp T name on telephone sets to V Tech V Tech is using the AT amp T name and line as their high end line Lucent s PBX division was spun off in 2001 as AVAYA MCLD Modifying Calling Line Disconnect The parameter on the Lucent 5ESS switch that determines if CPC is active Should be set to Yes if CPC is required
203. th the clear plastic protector inserting the plastic tabs in the slots on the Director to hold it in place Should you decide to change things later simply lift one end of the plastic and slide out the strip Mounting Options The Desktop Director legs can be mounted in an alternative vertical orientation To do so remove the legs by removing the two Phillips head screws on each leg The legs can then be mounted in the alternative orientation using the same screws The Desktop Director can be mounted on a wall or other vertical surface First remove the legs by removing the two Phillips head screws on each leg The unit can then be hung from two screws on 5 inch 127 mm centers Care should be taken that the screws do not protrude more than 1 8 inch 3 mm from the mounting surface 2 9 Ethernet The Ethernet 100Base T port is used for Web based configuration connection to Telos Assis tant Producer and other software applications and interface to Livewire networked studio audio components The Talkshow System fully supports Livewire For more information on using the Talkshow System with a Livewire Audio over IP network see Section 5 8 IMPORTANT If you are not using a Livewire ready network do not enable Livewire streams or you will flood your Ethernet network Only when a properly configured Livewire compatible Ethernet switch is present should Livewire streaming be enabled The Audio menu includes LW enable disable as the first it
204. the Feedback Reduction function Adjust Caller Ducking to a higher setting more ducking When mic processing is used connect the hybrid in such a way that it gets an unpro cessed mic signal The problem here is that the mic processing combines with the Talkshow Systems internal AGC to increase gain in the feedback path Depending on the processor used the feedback margin can be reduced by many dB The Talkshow System s internal AGC has a internal adaptive smart gate function to prevent inappropriate gain increase but it is thwarted by this additional processing If it is not possible to get an unprocessed mic signal try to set the processor in such a way that room noise is not sucked up during pauses You can also try reducing the output the mic processor or set the Nom In Nominal Input Level selection in the Audio menu to the next higher setting thereby reducing levels into the unit The level to the caller should be OK since the Send AGC will compensate however you will force that AGC towards its maximum gain and thereby reduce how much gain it can add to the feedback path Try repositioning the mics or speakers It also helps of course to use directional mics Customers have reported good results with cardioid mics such the EV RE 20 and the Shure SM 7 Add equalization to the monitor path Acoustic resonances usually result in pronounced peaks in the feedback response Since there are likely to be only a few pron
205. the ISDN connection is on the rear panel near each of the U jacks If the circuit is inactive the LED will remain in the rapidly flashing state This state indicates a loss of the ISDN BRI circuit at the lowest physical level Try another connecting cable or another line If the NT1 can contact the central office the LED will blink slowly about 1 Hz If the LED remains in this state there is an internal ISDN problem or you could be connected to the wrong jack on the back users in USA amp Canada should be using the lower RJ11 style jack The LED will come on solid when all handshaking is completed and the basic low level line connection is good If the LED stays on with the ISDN disconnected or does not come on at all after cycling power there is a hardware problem Condition Can t Get ISDN Ready Indication First try disconnecting the ISDN circuit s in question and waiting a full 60 seconds leave the Talkshow System on and then reconnect the circuit s What kind of ISDN do you have and is the Talkshow System set to match it The telephone company should have given you the ISDN protocol when the line was ordered If you find yourself in the field without this information it is usually possible to piece together what you need If you are given SPIDs you probably have Natl ISDN 1 or DMS 100 for your ISDN protocol One SPID means one channel was activated rather than the usual two No SPIDs and only one phone numbe
206. this ISDN BRI circuit will also be used with an audio codec such as the Zephyr or Zephyr Xstream then CSD should be ordered as it is required for audio codec calls CSV is for interworking with voice phone service and allows ISDN to interwork with analog phone lines and phones You will be ordering one or more ISDN BRI 2B D circuits with CSV required and CSD optional on each B channel NT1s The ISDN standard specifies two reference points the U and the S interfaces The U is the single pair bare copper from the Telco CO A device called a Network Termination Type 1 converts this to the two pair S interface In Europe Asia South America and most of the world the NT1 is always provided by the phone company Therefore Telos TWOx12s shipped outside the USA amp Canada have the S interface In the USA amp Canada the NT1 must be provided by the user and is therefore built in to the TWOx12 s ISDN terminal adapter In this case the U RJ 11 interface is provided Terminals and Terminal Types Any equipment connected to an ISDN line is a terminal whether phone computer or Hybrid Point to point lines support one terminal while multipoint lines can have up to eight in some applications Terminal Type is a parameter sometimes requested by the phone people The appropriate value for the Talkshow System varies depending upon protocol and is included on the order forms IOC Capability Packages provisioning
207. tings but the thinking is the same Too many times we fall in love with our suppositions We are quick to overtly or subconsciously assume the problem being chased is due to lousy design the stupid phone company or the managers latest memo Armed with a healthy skeptical attitude the basic philosophy of troubleshooting any system is to follow these steps Observe the behavior to find the apparent problem Observe collateral behavior to gain as much information as possible about the problem Round up the usual suspects Generate a hypothesis Generate an experiment to test the hypothesis Fix the problem Then repeat if necessary to attack additional problems Let us now cover each step of the troubleshooting sequence in detail Step 1 Observe the behavior to find the apparent bug In other words determine the bug s symptoms Remember always that many problems are subtle and exhibit themselves via a confusing set of symptoms Step 2 Observe collateral behavior to gain as much information as possible about the problem Does the LCD problem correlate to an LED flashing Try to avoid studying a problem in isolation but at the same time be wary of trying to fix too many at the same time No one is smart enough to deal with multiple problems all at once unless they are all manifestations of something more fundamental Step 3 Round up the usual suspects At the system level always suspect the menu set up the ca
208. tional information from Telcordia http www telcordia com resources genericreq g1303 index html SLIC 1 Subscriber Line Interface Circuit see Line Card 2 The equipment used with the AT amp T Lucent SLCC Subscriber Loop Carrier Circuit a system used to multiplex a number of subscriber loops onto a single circuit usually a T1 circuit to reduce fixed costs 3 Also sometimes used generically for other brands of similar equipment See also SLC 96 SPID Service Profile IDentifier On the National ISDN AT amp T Custom PMP and Custom DMS ISDN BRI protocols the Telco switch must receive correct SPID s from the CPE before it will allow access to ISDN service Intended to allow multiple configurations on ISDN lines shared among different types of CPE equipment While your SPID may include your area code and telephone number the SPID is distinct from the telephone number For the National ISDN Custom PMP and DMS custom ISDN protocols the equipment requires that the user to program SPIDs into it Custom PTP and ETS 300 protocols do not require 114 Appendix 2 a SPID No SPIDs are required for ISDN PRI protocols SS7 Signaling System 7 The internationally adopted Common Channel Interoffice Signaling CCIS system Previous to SS7 the Bell System used SS6 which did not support the International Standards SS7 does It allows for substantially flexibility and power in dynamically routings calls An SS7 data
209. tputs To use this mode your mixing console must be capable of splitting and combining separate channels of audio to and from the L R pairs of AES signals This is sometimes called the dual mono mode of AES operation Program on Hold is via the analog inputs AES POH AES Program On Hold In this mode there are two caller inputs outputs and two AES Program on Hold inputs In the case when only two caller inputs outputs are needed and the mixing console is capable of splitting and combining separate channels of caller audio to and from the L R pairs of AES signals this mode allows the Program on Hold to also be via AES The caller inputs use AES XLR Caller input 1 both L R channels and Program on Hold inputs use AES XLR input 2 both L R channels The following table shows the relationships of AES to the input and output signal paths shown in the block diagrams above 2 channel 2 channel 4 channel AES POH 1 XLR connection 2 XLR connections AESin 1L input 1 Input 1 Input 1 Input 1 AESin1R Input 2 Input 2 Input 2 AES in2L Input 2 Input 3 POH in 1 AESin2R Input 4 POH in 2 AESout1L Output 1 Output 1 Output 1 Output 1 AES out1R Output 2 Output 2 Output 2 Output 2 AES out2L Output 2 Output 2 Output 3 Output 2 AES out2R Output 1 Output 1 Output 4 Output 1 AES Output Sync
210. ults were revealing Consequently some form of receive equalization is desirable There are three choices for the type of equalization applied to the received telephone audio Off The caller audio is passed through without modification The Rev EQ values are ignored in this mode Fixed This is a simple manual equalizer mode where a fixed EQ is applied Adaptive This is a three band dynamic equalizer The high and low frequency boost or cut is automatically determined and set The Dynamic EQ uses the values set below as target levels Additional Low EQ Amount of low frequency boost or cut applied to the caller audio Settings are in 2dB steps from 8dB to 8dB are possible with OdB representing flat response The default value is 4dB which sounds good across a variety of callers lines and telephone sets Additional High EQ High Amount of high frequency boost or cut applied to the caller audio Settings from 8dB to 8dB are possible with OdB representing flat response The default value is OdB Feedback Reduction This enables or disables the acoustic adaption process used for feedback control when the system is used with an open speaker to microphone acoustic path This setting also increases by 6dB the amount of send to caller audio ducking It should be set to On when it is needed to elimi nate feedback Since the send to caller audio will be colored by this process should be turned off unless needed
211. umber DN These are simply your seven digit phone numbers as would be listed in your local telephone directory and will need to be entered into your TWOx12 in rare cases you may need to enter the full 10 digit number Normally each ISDN BRI circuit will have two DNs and two SPIDs Each DN corresponds to a given SPID and when you enter this information you must enter them to reflect this relationship You may also hear of a Listed Directory Number or LDN This is simply the primary number for lines in a hunt group see below The LDN need not be entered in the unit unless it is also one of your DNs In some cases lines configured to hunt may deliver all calls to a single DN number In this case you must enter that DN for each of the DNs on each of the hunting channels CSD and CSV Recall that each ISDN BRI has two possible B channels It is possible to order a line with one or both of the B channels enabled and each may be enabled for voice and or data use Phone termi nology for the class of service is CSV for Circuit Switched Voice and CSD for Circuit Switched Data In contrast to PSD Packet Switched Data which is possible but irrelevant to our needs USA ISDN ORDERING GUIDE 119 CSD is not required for Telos Talkshow System connections Even though you are sending a stream of data the phone network knows to translate this back to analog at the far end due to the call setup message sent on the D channel when the call is made If
212. use is used to indicate that the original destination is unavailable and to invoke redirec tion to a new destination Cause No 49 QOS unacceptable This cause is returned when the Quality of Service is unacceptable CLASS 1 3 service or option not available Cause No 50 Requested facility not subscribed The cause is used to report that the user cannot use this feature because s he has not subscribed to it Cause No 51 Bearer capability incompatible with service request This cause indicates a user request for action was rejected because the action was incompatible with the capability of the call Cause No 52 Outgoing calls barred This cause indicates that because of call screening provided by the network the calling user is not permitted to make a call Cause No 53 Service operation violated This cause indicates that the user has violated the service operation Cause No 54 Incoming calls barred The user will not accept the call delivered in the SETUP message Cause No 55 Incoming 90 Section 6 calls barred within CUG at destination This cause indicates that incoming calls are not permit ted to members if this Closed User Group Cause No 57 Bearer capability Data voice not authorized This cause indicates that the user has requested a bearer capability which is implemented by the equipment which generated this cause but the user is not authorized to use it This is a common problem caused
213. use the A and V arrow buttons to can change the value of the setting When you reach the value you desire press the OK button again to lock it in and return you to the item selection scroll mode The Web User Interface A Web Browser can be used to remotely configure and administer the system All front panel configuration options are duplicated on the web pages Certain advanced options such as enter ing ISDN DNs SPIDs and Livewire configuration require the Web interface You can connect a PC directly to the Talkshow System with an Ethernet cross cable Or you can connect both the PC and the Talkshow System to an Ethernet switch or hub Important Note If Livewire has been enabled there will be a lot of network traffic generated by the Talkshow System Some PCs switches hubs and routers that are not multicast aware will not be able to handle the volume and Web communication will not be possible Normally you would use the Web interface to switch off the Livewire streams but since you can t communicate with the Web interface you will have to first disable the Livewire streams from the front panel LCD button interface The first item under the Audio menu is what you need DETAILED CONFIGURATION amp REFERENCE 45 The Talkshow System s default IP number is 192 168 0 230 Normally you would change this to a number appropriate for your network using TCP IP Setup in the System menu You would also enter appropriate values for Subnet
214. vel callers The AGC has a range of up to 12 dB The AGC is an advanced dB linear feed forward topology providing a consistent sound independent of drive level Its smart operation normalizes levels while retaining the natural dynamics of the caller s voice The default setting is FULL An important additional feature of this AGC is that it is cross coupled to other sections of the Talkshow System and can therefore reliably distinguish between caller audio and hybrid leakage This allows a more aggressive gain control for bringing up low level callers while still preserving excellent hybrid performance 76 Section 5 Noise Gate Turning on the noise gate enables the built in downward expander The downward expander reduces low level line noise when no caller audio is present And reduces low level leakage thereby improving hybrid performance This function is cross coupled with the AGCs and the ducking system Digital Dynamic Equalizer Telephone audio frequency response varies widely as many different factors can effect it we ve measured the response on a number of calls and the results were revealing Consequently some form of receive equalization is desirable The Digital Dynamic EQ process used on the Telos is the most sophisticated equalizer avail able in a broadcast telephone interface All processing is performed in the digital domain The Receive EQ selection controls the type of equalization applied to the receive
215. vices were available to connect to these ports Desktop Director Extended Desktop Director Console Director Call Controller The Talkshow System has four Desktop Director ports Each supports two devices via simple parallel connection allowing up to eight total devices However the Talkshow System can only power four devices If you have more than four devices you will need to use the external Desk top Director power supply kit available from Telos The Extended Director counts as two devices for power purposes but uses one connection The Talkshow System can control eight Extended Desktop Directors but will power only one of them It does not matter what port you plug your first four devices into since all ports support power You can plug two devices into one port with the Desktop Director power supply kit it comes with a Y cable that allows two devices to be powered by one supply Desktop Director This is the most common control surface It serves as both a phone and call controller depend ing on whether it is set to Talent on air controller or Producer Screener phone It has two telephone line columns The Desktop Director is suitable for a Producer screener telephone in all cases INSTALLATION 19 Extended Desktop Director You have four columns in the Extended Director In Talent mode you can assign a caller to each of the four hybrids by pressing the corresponding button for that column
216. will NOT busy out when the Busy All function is enabled Any Priority line ringing will cause the GPO option Priority Ringing Out on the Sub D connector to go Active low Share with lt ip address gt amp Lines shared with POTS only No ISDN lines This special option allows two Nx Talkshow systems to share bridged POTS lines Both Nx systems must be connected to the same Ethernet subnet and have valid IP addresses For each unit enter the IP address of the Nx system it is sharing phone lines with Lines that are shared between units must have identical check marks per line ie sharing lines 1 through 6 place check marks on 1 through 6 on both units Share with lt ip address gt column Line status of a phone line is passed between Nx systems via TCP IP Ifa line is in use on one Nx system the other Nx system should see an X symbol on the Line display of the Desktop Director indicating IN USE ELSEWHERE Note This option is independent of the SHOW field and does not affect these shared options Settings will be saved when you press APPLY Audio 1 0 Configuration This is a duplicate control found in the main Audio menu web page Choosing your Mix Minus Routing Mode will automatically configure your internal audio routing In addition to selecting this option the web diagrams displayed under this option Audio Connections and Audio Rout ing will change to show the configuration of the rear panel connectors This helps take the gues
217. ybrid itself off the bus NOTE NOTE This must be turned down or AUX 2 will no longer be a mix Minus This must be turned down or AUX 1 will no longer be a mix Minus Mic Preamp OUTPUT FROM Y 1 UTILITY MIXER AUX 1 O we A TT Mic1 Calleri LEFT OUTPUT RI GHT OUTPUT AUX 1 OUT AUX 2 OUT TO HYB 2 IN 2 TO HYB 1 IN 2 Caller2 Phones and Remotes When on remote to save money and hassle calls are usually received at the studio rather than at the remote site In this situation caller audio must be fed to the remote talent so that they can hear and respond to callers Moreover the callers need to hear the talent In many cases the remotes are sufficiently distant that talent cannot monitor the station for the caller feed Even if they could the profanity delay would be a problem since the talent needs to hear the callers pre delay All perceptual codecs such as the Telos Zephyr have too much delay for talent at remote locations to hear themselves via a round trip loop Therefore another mix minus is required to feed a codec The talent hears callers via the codec return path As before you feed this return with mix minus a mix of everything on the program bus minus the remote audio INSTALLATION 15 As for the second half of the equation the callers hear the talent because the remote feed is added to the telephone mix minus bus
218. you extra be sure to request a refund ISDN Worldwide outside the USA amp Canada The S T Interface The S interface version of the Talkshow System connects to the NT1 at one of its terminal jacks The NT1 is sometimes generically called a NCTE Network Channel Terminating Equipment or in some countries an NTBA Nx6 S Interface RJ 45 PIN oN OO UU Bb U N DESCRIPTION N C2 N C 3 Transmit to network 4 Receive from network 5 Receive from network 6 Transmit to network 7 N C8 N C Nx12 S Interface RJ 45 PIN oN DO UU Bb UN DESCRIPTION N C N C Transmit to network Receive from network Receive from network Transmit to network N C N C 9 10 Section 2 ISDN S interface cable Use only an 8 conductor RJ 45 style cable Unshielded twisted pair category 3 or better cable should be used ISDN S Interface Cable RJ 45 PIN COLOR DESCRIPTION 1 White Green PS3 power ground optional 2 Green PS3 power optional 3 White Orange Transmit to network 4 Blue Receive from network 5 White Blue Receive from network 6 Orange Transmit to network 7 White Brown PS 2 power optional 8 Brown PS 2 power ground optional TIA 568A Category 5 cable colors shown Both ends are wired identically not crossover The Talkshow System does not use these power connections When fully wired this cable has 4 twisted pairs straight through just as with cables used for Ethernet Both
219. your unit The AES inputs have the following characteristics AES3 compliant 110 Ohms Sample rate converted so may accept sources at any of the common rates from 32 to 48 kHz AES Input XLR PIN DESCRIPTION 1 Ground 2 Signal 3 Signal So that all four hybrids are accessible in spite of the limited rear panel space for connectors we use one AES input for two hybrids The relationship is as follows AES 1 Left is to Hybrid A AES 1 Right is to Hybrid B AES 2 Left is to Hybrid C AES 2 Right is to Hybrid D INSTALLATION 17 Most mixing consoles with AES outputs are able to deliver the required split If not you might need an external AES router or combiner to provide this input An alternative could be to use a Talkshow System configuration requiring only a single mix minus see Mix Minus Modes in Section 5 4 The role of each hybrid depends upon the configured Show Mode which is described in Section 5 7 POH Program on Hold Inputs This analog input is for whatever audio you want callers to hear when put on hold usually your main program feed This is an XLR female This signal is processed by an AGC within the Talkshow System Output Caller Analog Outputs The analog audio outputs have the following characteristics Active differential Output level 10 to 8 dBu nominal Clip point 24dBu Impedance lt 60 Ohms x 2 Analog Output XLR PIN DESCRIPTION 1 Ground 2 Audio 3 Aud
220. yr Express do not require that Directory Numbers be entered This is not true for the Telos Talkshow System and other newer products You must correctly enter these numbers for the system to work 56 Section 5 Note for the Curious The actual physical B channels of the BRI line are assigned on a per call basis with the first incoming call assigned to the first B channel However the call setup message on the D channel includes what number was dialed and the Directory Number allows the system to sort it all out IMPORTANT After you enter or change your SPIDs or DN s you must restart the system to use the new information Therefore once you have entered SPIDs DNs or MSNs cycle power off and then back on Then go to the ISDN status screen and verify Ready status for all channels POTS Country Affects analog lines only This setting configures the analog Telco line interfaces for proper impedance matching and other characteristics for different countries If your country is not listed try using CTR21 European standard or USA CallerlD Alert Signal Web Only Affects analog lines only There are several different alert signals that are used to signal equipment that Caller Identifica tion is about to be sent also known by CLID and other names see Appendix 2 The signaling used varies by country and sometimes within a country The POTS Country setting chooses a reasonable default for any given country

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