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USER`S MANUAL - Telos

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1. 10000 20000 SIP RTP 8080 HTTP Server A computer on the same subnet as the Network interface example 192 168 0 100 could log into the VX with this URL http 192 168 0 200 while a computer on the same subnet as the WAN interface example 192 168 5 2 could log on using http 192 168 5 200 8080 Default user name is user Default password is lt blank gt Passwords are case sensitive If you change any of these you might want to write the new values into this manual here and elsewhere 16 Section 2 SIP Configuration Telos Y TelosVX Control Center Confi ti n SIP Configuration Patton PRI General settings Cisco Asterisk Default Server 192 168 0 37 v Studios Use SRV Lookups O Shows Call Audio Processing Apply Tones Eaaaing Servers Information Studios Name Server Lines J Refs Calls Patton PRI 192 168 0 37 0 0 Delete Devices Cisco 192 168 0 23 0 2 Stream Statistics Script Manager Asterisk 192 168 0 155 1 1 jelet Script Information Misc OManager Info Add New System Status sever Ade Logs Backup Restore Firmware The SIP configuration page shows the global SIP settings along with a list of all SIP servers VX is configured to communicate with The form at the bottom of the page lets you to add another server VX will automatically create server configuration for any line in the Show configura tion As a consequence it is not possible to delete server entry
2. Configuration Main SIP Studios Log level Custom mw Advanced DSP Trace Shows Hybrid v Log to internal Tones Log to Syslog 3 Host 192 168 0 31 pees udi 7 Calls Devices Stream Statistics Script Manager Script Information Misc lOManager Info System Status Logs Backup Restore Firmware The VX s logging capability is a powerful tool to track down problems even those due to operator error not that that ever happens In the Logging config page you determine where logs are stored and how much detail is recorded You can choose to store logs inside the VX or to an external PC on the network that is set up to handle Linux standard syslogs If you choose the internal option log messages will be kept in the VX s RAM so are limited in size The advantage is convenience You can view internally stored logs with the System page log section by default at http 192 168 0 7 log A typical log is shown in the screen capture below For basic troubleshooting choose Log to Internal then go to the Log page clear all existing messages then place a call or make the operation that s causing trouble Then return to the Log page to see the entries While you might be able to understand the log messages after some effort we don t expect you to do so These are mostly intended for our support people who would advise you as what settings to use for the capture and then interpret
3. We are routing callers who dialed 7605130999 directly to Extension 2222 on the VX Click submit and then apply the changes We could also send those calls to any of the other options such as Misc Destinations Mental Health Hotline or Joe s Voicemail These destinations can be created and added by you in the FreePBX GUI Remember that if the show that uses extension 2222 is not selected on the VX through the VSet or VX Producer software calls to 2222 or the DID number routed to it will receive an All NOTES RESOURCES ADDITIONAL INFORMATION 89 Circuits Busy recording from Asterisk because all circuits are busy as there is nowhere for the call to go Outbound Calls Below we decide to use 9 to dial outside the PBX so we cleverly name the route 9_outside to make it clear later what the thing does In the Dial Patterns box we enter 91 What this does is match the leading 9 from the number dialed let s call it 913115552368 deletes the 9 everything ahead of the pipe symbol I is deleted then the represents the rest of the number dialed 13115552368 and passes that number to the trunk selected at the bottom of the outbound route page in this case our Vitelity trunk Connect your VX to Asterisk We re going to assume that you ve done the basic setup of your VX That means you ve got audio going and a VSet phone connected and talking to your VX engine and it s on the same ne
4. Default IP numbers passwords etc Default Passwords VX Engine username user password lt blank gt VSet Telephone username user password lt blank gt NOTES RESOURCES ADDITIONALINFORMATION 91 Default IP Addresses VX Engine Network interface 192 168 2 200 front panel settable VX Engine WAN interface IP address OFF by default no IP set SIP Parameters Codecs supported as of early 2011 g 711 alaw and ulaw g 722 SIP Trunks SIP Stations TCP IP Ports and protocols used by the VX System 80 HTTP TCP Network Port only 22 SSH TCP 5004 used internally by DSP engine UDP 5060 SIP UDP 4 8080 HTTP TCP WAN Port only 20518 Livewire ports 62000 through 62513 RTP UDP Port numbers may be changed on the WAN interface for security and flexibility VSet telephone VSet Telephone default IP address 192 168 0 201 front panel settable Default username and password username user password lt blank gt telnet requires no password TCP IP Ports and protocols used by the VSet Telephone 80 HTTP TCP Network Port only 23 Telnet TCP 20518 Livewire port Power Over Ethernet PoE The VSet uses standard 802 Power Over Ethernet POE and consumes 15 4 watts Please consider this relatively high power consumption rate when choosing your POE source 92 Section 5 VX FAQ The Telos VX uses VoIP What does that mean to me Let s define what uses V
5. Support user mobility across networks and devices Support for multipoint conferencing 4 Support presence information Inform users as to call progress Communicate requests for QoS to various network elements such as IP routers While the various servers above could run on individual machines and could even be physi cally separated by thousands of kilometers in usual practice they are often just software elements of an application running on a single machine As we ll see many small IP PBXs include the Gateway as well making a one box solution that includes everything needed for a small office installation An example a SIP Server being used in the broadcast world comes from the Telos Z IP codec family which uses a Telos developed enhanced SIP Server called naturally Z IP Server The server is provided as a service on the Internet but may also be installed by users who prefer to maintain their own In addition to basic SIP functions registration address look up the Z IP server offers additional services 4 Allows display and dialing by simple text name Keeps a database of names and per forms DNS IP look up upon a dialing request from an endpoint codec 4 Maintains group lists created by users Upon entering a group name and password the list is displayed on endpoint codecs so that users aren t burdened with having to enter or upload lists manually 4 Provides geolocation services by associating IP numbers with physica
6. The request includes the address of the caller and the address of the intended called party In more sophisticated scenarios users register with a registrar server using their assigned SIP addresses The registrar server provides this information to the location server upon request From time to time a SIP user might move between end systems The location of the user can be dynamically registered with the SIP server Because the end user can be logged in at more than one station and because the location server can sometimes have inaccurate information it might return more than one address for the end user If the request is coming through a SIP proxy server the proxy server tries each of the returned addresses until it locates the end user If the request is coming through a SIP redirect server the redirect server forwards all the addresses to the caller in the Contact header field of the invitation response NOTES RESOURCES ADDITIONALINFORMATION 73 When communicating through a proxy server the caller sends an INVITE request to the proxy server and then the proxy server determines the path and forwards the request to the called party Since we usually need to reach phones that are connected to the PSTN gateways will be involved in real world systems These translate SIP signaling to the PSTN s requirements loop current DTMF and ring detect for POTS lines set up messages for ISDN etc 26 IP Phone SIP Server SIP POTS Gateway
7. Your ethernet interface eth0 should now be up and working If you re connected to the Internet type ping yahoo com lt enter gt and if you re successful you ll get your ping returned from yahoo com To stop the pinging hold down control and hit c Ctrl c You should get the prompt back again This little test proved two things that you have internet connectivity and that your DNS name resolution worked it looked up the IP address for yahoo com and gave your box the ip address for them If you reboot the machine the eth0 interface will come up automatically from now on Please be aware that this tool only configures a single Ethernet card If you have more you can configure them later TIP Is your system really slow at certain points or does it seem to stop DNS is used for many things in Linux If it isn t available many services will be VERY slow Minutes long pauses waiting for name resolution timeout are not uncommon and some things will appear to stop working entirely So if you have long pauses while working with your system lack of DNS is a probable cause If you plan to run your system without DNS you ll want to fill in the hosts table with any addresses that might be logged or talked to by the system It is found at etc hosts You can edit the file using mc Midnight Commander or other editors like pico and vi Follow the examples in the file Set the passwords for various features
8. gt Example Call 311 555 2368 called from a cell phone in Barstow California 760 256 8463 Cell carrier routes the call to the SIP provider over the Public Switched Telephone Network as the number translates to the SIP provider Asterisk Switch Asterisk SIP Trunk Digits sent include Caller ID and 10 Digit DID number Asterisk Zaptel or DAHDI type trunk Digits sent include Caller ID and 4 10 Digit DID number Asterisk SIP or IAX2 type trunk Digits sent include Caller ID and extension number that calls to POTS line are auto forwarded to Asterisk Extensions May be SIP or ZAPTEL or DAHDI hardware such as expansion cards or a Channel bank Most often used to create analog extensions Asterisk Trunk or trunks May be SIP IAX2 ZAPTEL or Dahdi Hardware such as expansion cards gt gt Ext 2368 rings and shows 760 256 8463 as calling number on Vx The Vx receives 2368 and ringing commences while the Caller ID data is displayed along with the ringing indication A ringing line lamp flashes triggered by a GPIO interface as set in the Vx engine Studio definition Vx Engine or Telephone sets Vx Engine or trunks to other PBX s 311 555 2368 with Caller ID of 760 256 8463 is received The SIP provider sends the call to the Asterisk SIP trunk because 555 2368 is registered to the trunk belonging to the subscriber Caller ID and the 10 digit
9. soned that if they could pass the full 64kbps rate of the network to users they d be thrilled with the amazing speed And with two channels you could simultaneously talk and look up recipes YouTube had apparently not entered their imagination In the modern networked world The Telos VX gets with the program It is the next generation multi studio phone system from Telos providing a powerful simple and cost effective way to share phone lines across a number of studios using standard IP technology It offers a number of advanced features to enhance production of talk shows and active DJ use of phones For years you have been asking for an uncomplicated way to share phone lines around a facility to multiple studios And we are always hearing that you need a way to better integrate the studio system with your office phone PBX Finally the VoIP based Telos VX lets us do this We re the studio phones company We re also the IP Audio company With the introduction of the Telos VX we are marrying the two creating an exciting new approach to broadcast studio phones The Telos VX is a scalable system that provides a solution for stations with modest needs while being able to grow to support facilities that have large numbers of Telco lines and studios The standards based SIP IP interface makes all the things we ve wanted to do possible Passing calls between your office PBX and on air system is easily accomplished Sharing Telco service
10. device was chosen as a measure to optimize audio quality Fortunately the NEC PBX provides disconnect supervision via a momentary break in loop current The gateway was configured to respond to this if that hadn t been possible the gateway would have to be configured to respond to the return of dialtone In the other direction two dialtones are provided from the VoIP side to the Nortel PBX via a Linksys two line ATA Analog terminal Adapter an FXS gateway that converts VoIP to analog including talk battery generating ringing etc WKSU has a number of phones on the PBX and this puts a couple of the VoIP numbers on phone buttons An Asterisk PC based PBX is installed functionally in front of the VX to provide a basic automatic answer and IVR Interactive Voice Response function Callers are greeted with a welcome message and a choice of leaving a message going on the air or speaking to the receptionist When a caller chooses the option go on the air the call is passed to the VX where it may either be directly answered on air or be fielded by a producer This IVR function is created through a web GUI for Asterisk called FreePBX 98 Section 5 Calls are routed from Asterisk to the VX via SIP VoIP over the LAN The Asterisk is configured to provide a SIP extension for each line that connects to the VX SIP trunks could have been used but extensions are more capable for example they can pass Caller ID The
11. www patton com voip We have tested and support Patton gateways such as the model 4940 for T1 E1 and ISDN PRI The Patton gateways for POTS FXO and ISDN BRI are also satisfactory www grandstream com www audiocodes com www cisco com www digium com www quintum com VoIP SIP equipment suppliers www telephonydepot com www rockbottomvoip com 66 Section 5 Telephony Discussion and VolP news sites www sipforum org www telos systems com www voipforums com www voipuser org forum_index html www broadbandreports com forum voip forum voxilla com SIP Information and suggested reading A Request for Comments RFC is a memorandum published by the Internet Engineering Task Force IETF describing methods behaviors research or innovations applicable to the work ing of the Internet and Internet connected systems The IETF adopts some of the proposals published as RFCs as Internet standards These are available at www ietf org rfc Among those relevant to VoIP and SIP are RFC 1889 Real Time Protocol RTP original version RFC 3550 Real Time Protocol RTP latest version RFC 1890 Real Time Protocol Audio Video Profiles RTP AVP original version RFC 3551 Real Time Protocol Audio Video Profiles RTP AVP latest version RFC 2327 Session Description Protocol SDP original version RFC 4566 Session Description Protocol SDP latest version RFC 2543 Session Initiation Protocol SIP original ve
12. 2 2 ee ee 61 SPOCiNCALIONS so e iaa a UR APE aw aed 62 O edad Se ee ates ae Sek ee Be a aces 62 Audio Performance o o o s seeren waro nren nreno raau 62 Controller s s sa va hee eae howe eee a ee 63 VAENQING 3 5 542 Goo we oR a eo wre ba oe 48 63 IP Ethernet Connections 2 ee 63 Processing FUNCIONS 2 2s amat a ae e e hee Ge oe 63 Studio Audio Connections o oo aa 64 Telco Connections iee e ae a ee ee ee 64 RESQUICES oe y rra al a e Bea e aad ete ee 64 TELOS VX MANUAL vii Internet Speed tests o ee ee ee 64 Internet VOIP JitterTest gt ooe ee ee 64 Packet sniffe co ora peas wea 65 VolP Soft phone SIP PC clients o o o o o o o o o ooo 65 Open Source PBX Distributions o o 65 Commercial PBX Products s sus e enega ienanc ooo 65 Gateway products and suppliers ooo aa 65 VolP SIP equipment suppliers ooa aaa 65 Telephony Discussion and VolP news Sites 66 SIP Information and suggested reading 66 BOOKS OR N a 66 SIP Providers o ote Rae Oe abe eA oa See ed 67 Local Number Portability in the USA and Canada 68 Introduction to SIP s insides forthe Curious 68 The PartsofaSIP System o o 69 Address s a tota ae dia ooh aia us 71 How SIP Works s a ooh a4 nei ae ee a EY ew ace a 72 The State of SIP and its Future 74 lAXasaSIP Alter
13. Asterisk PC is equipped with three Ethernet ports One is connected to the LAN and nor mally used for all input and output Another is directly connected to the networks that deliver VoIP calls as a backup pfsense a software router is installed on the machine to perform the routing to select the active VoIP path and provide the automatic backup The VX Producer application is used for call screening and communication between the producer and air talent Two Cisco 2960G Ethernet switches provide the networking One is on the Livewire LAN side of the VX and the other is on the VoIP WAN side NOTES RESOURCES ADDITIONAL INFORMATION Incoming Call flow to VX via Asterisk Open Source PBX DID numbers from SIP Provider 311 555 2368 311 555 9467 311 555 1067 DID Numbers from ISDN PRI 311 555 7200 7299 311 555 1300 1399 Numbers from POTS Gateway 311 950 1022 311 976 1234 311 853 1212 311 958 1114 gt gt Asterisk Switch Asterisk SIP Trunk Digits sent include Caller ID and 10 Digit DID number Asterisk Zaptel or DAHDI type trunk Digits sent include Caller ID and 4 10 Digit DID number Asterisk SIP or IAX2 type trunk Digits sent include Caller ID and extension number that calls to POTS line are auto forwarded to Asterisk Extensions May be SIP or ZAPTEL or DAHDI hardware such as expansion cards or a Channel bank Most often used to create analog extensions As
14. DID number are sent to the Asterisk SIP trunk belonging to the Subscriber 311 555 2368 is routed to extension 2368 on the Asterisk switch All info sent is passed onto the Vx The Asterisk switch receives the digits and attempts to match the DID number to an Incoming route If matched the call is sent further to either and extension trunk or special treatment in the system such as voice mail or a busy signal if there are no free trunks or buttons at the far end Ext 2368 rings and shows 760 256 8463 as calling number on Vx The Vx receives 2368 and ringing commences while the Caller ID data is displayed along with the ringing indication A ringing line lamp flashes triggered by a GPIO interface as set in the Vx engine Studio definition WARRANTY 101 1 Warranty Telos VX 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
15. Dec 01 2010 11 26 04 Slot 5 ver 0 4 3 0 r8165 k2 6 17 14 pce7197 Thu Mar 03 2011 11 48 57 Active Note You have to reboot the system for changes to take effect New firmware for the VX Engine is first obtained from Telos via email Internet download etc It is then uploaded to the VX Engine using this page You can have as many as five stored in the Engine at one time On this page you choose which of them will be active Reboot the Engine after activating new firmware Backing up and restoring Engine configuration TelosVX Control Center Configuration Main SIP Studios Shows Backup Call Audio Processing Tones Save settings to an XML file Backup Settings Logging Information Restore Studios Calls Restore previously saved Browse Restore Settings Devices settings E Keep current network settings Stream Statistics Configuration Management Script Manager Script Information Reset Misc gt lOManager Info Reset to factory defaults O Reset network settings Reset to Defaults System Note The system will automatically restart after the configuration is loaded reset Status Logs Firmware 32 Section 2 Backup restore configuration settings The backup file for configuration settings is stored on your local machine or on a removable drive attached to it VX also works with Axia iProbe the automated IP Audio network management software VX Engines appear in iProbe just
16. Disable Stream Statistics Default Script Manager Reorder Tone AA Upload Disable Script Information Default Misc Error Tone i lOManager Info Browse Upload Disable System Status Call Disposition Logs y 40 Backup Restore A Custom fx_answer au Tue Jun 28 2011 14 10 23 98 8 KB Firmware Browse Upload Reset to Default Delete Default Caller Hang Up Browse Upload Disable Custom fx_switch au Fri Jun 10 2011 21 23 19 293 5 KB ial Browse Upload Reset to Default Delete r Custom fx_alert au Mon Jun 27 2011 16 30 40 374 87 KB Caller Alert Browse Upload Reset to Default Delete Played only to the caller INSTALLATION AND CONFIGURATION 27 SIP signaling is via a text message not audio This means the VX has to make its own sounds in response to various line status conditions We provide default sounds which are similar to the traditional tones generated by the PSTN but you can upload your own to create a unique on air signature You can also use any sound you like to signal that a new call has been taken This is actually a cool new feature made possible by modern technology Rather than a new call appearing silently the event can be accompanied by a signature sound A story Years ago we installed a fancy new Telos system at Z 100 New York replacing an old key phone setup The new system had a mute function to remove the loud and annoying clunk that banged out when each call was taken to air On the f
17. In the former case a stereo file will be produced Enabling Auto Record causes the recorder to automatically make a new recording each time a call is taken File names are automatically created by the system but can be changed by writing over the default text after a right mouse click The file will appear in the tabbed list box determined by Settings Normally a VX Producer in producer mode will put recorded files under the Producer tab and in studio mode under the Studio tab naturally You can move files among the tabbed lists by either dragging them to the destination tab or using the Send button and drop down list The meter to left of the tabbed file list window shows the audio level as a recording is being made Playing Select a file by clicking on it Press the Play button The audio will be sent to the PC s output configured in Settings Depending on the Settings configuration the playback will be available on PC connected loudspeakers or headphones or it may be routed to a studio mixing console fader Editing Select a file by clicking on it Press the Edit button The audio editor configured in Settings will open and the selected file will appear in the editing window Audacity is the default Audio input and output will be according to the editor s configuration Depending on the configura tion the payback will be available on PC connected loudspeakers or headphones or it may be routed to a studio mixing console fad
18. Managing Calls ocio aa 43 Handset 6 ab 4 tea aed eal b be ca lead 43 AA Be eee eee Re Ee ote Ses 44 The line nfo Field arrasada Oe Re a E 44 Drop Hold and Hold Ready Buttons o 44 Selectable Lines aa 4 fac eS a e e a a 45 Lock Unlock Button and Function 2 2 2 eee eee 45 Fader Assign Buttons 2 ee ee 45 Next Button and Function aaa a 46 Block All s oag ou a e be era A A a Bek 46 Fixed Lines s in aa a ae Wk RO OR ee 46 Recording Editing and Playing Calls aaaea 46 Playing e e e Fe bee a Mee eee es 47 FICS a eh Bye eres A Re wy a AS Hd ee et 47 Host Producer Text Chatting 2 ee 48 5 Notes Resources Additional Information 49 The Acoustic Echo Canceller 2 2 ee ee 49 Telco Services and Interfaces 2 ee ee 51 VoIP SIP TRUNKING a on a a ta 51 IP Centrex and Hosted PBX Services o o o o o a 52 Number portability in the US and Canada 52 Circuit Switched Interfaces o o o o o o o o ooo ooo 52 ESO nd a eles tee Oe Ones e E 52 ESMTTINKSS oso aora aran AAA 4 53 TYE bie ib ee haw a ees 53 ISON PRL rs atte oe Beet tae SOR a See we 53 ISON BR cocoa iia naa leido oat WES 54 SIP Compatibility s e e e aio ra a 54 Axia Element Console as VX Controller 2 2 eee eee 54 Installation 1 42 bs a a ebb E 55 Configuring the Element to Control the VX 2 0 0 2 0004 56 MO A bch ot a Seb a Walaa deed 60 Beyond Edison s Legacy
19. NOTES RESOURCES ADDITIONALINFORMATION 79 ISOLINUX 3 11 2095 89 92 opyright C 1994 2885 H Peter Anvin a Flash Version 1 7 5 5 3 32 bit 183115 ou MUST be connected to the internet in order to install this program ARNING This install HILL FORMAT ERASE ALL DRIVES attached to this system INCLUDING USB DRIVES For a default install just press enter For an LUM based install type kslum For a network install type ksnet and remove CD at keyboard prompt For an auto install no prompts type ksauto The password is passHorm For a raid based install type ksraid ASTERISK VERSION IS CHOOSEN AFTER CENTOS INSTALLS Fi Main F2 Options F3 Generall F4 Kernel F5 Rescue oot Select the US keyboard or the keyboard of your choice when prompted then press the lt TAB gt key to advance to the OK prompt After this selection the GUI starts up Move your mouse to verify that it works Keyboard Type What type of keyboard do you have sg latin1 sk qwerty slovene sv latini trq ua utf uk The system asks how you want to use the disk It points out that You have chosen to remove all partitions ALL DATA on the following drives lt lists drives gt If you re sure that this is OK and it should be select Yes You ll be asked your time zone Select the correct one and click Next Next you ll be asked to enter your root password It is case sensitive Enter and confirm it as asked Then
20. Producer identifiers are added automatically The colors for them can be configured in settings Tech note The VX Engine works as the chat server 49 5 Notes Resources Additional Information The Acoustic Echo Canceller You can learn how to connect the Acoustic Echo Canceller to your studio equipment in the Studio Configuration section Here we provide more detail on how it works and how to use it A common annoyance in radio studio and TV studio operations is the feedback that often results from using a loudspeaker to listen to telephone calls This comes from the acoustic coupling of the audio from the loudspeaker to the studio microphone that feeds to the caller As well echo might be heard by the caller When a caller talks the phone audio bounces around the studio and gets sent back Due to the time dispersion caused by the room and the round trip transmission delay this acoustic echo is distracting to the caller In the past when analog Telco lines were the norm and mobile phones were not yet attached to everyone s ears feedback was the usual problem not echo per se In earlier Telos hybrids we addressed that with a combination of ducking frequency shifting and a basic canceller That worked pretty well in its day but Mobile phones and VolP connections impose much more delay and the problem has become more challenging With the multiple hundreds of milliseconds latency that can occur these days an effecti
21. VOIP Hacking O Reilly Asterisk O Reilly SIP Providers Something that takes some time to get one s head around is that SIP providers usually do not provide the IP network connection The main service they perform is giving you a gateway to the PSTN That is they give you telephone numbers The gateway need not be local to you Indeed you may be in California and find that your provider s gateway is in New York Providers also give you access to a SIP server that helps with obscure things like traversing NATs and taking care of relocation services Because the SIP provider and network vendor are independent it s sometimes hard to know who to blame for troubles or praise for good service It s a good idea to discuss network vendors with the SIP provider since they have built up a lot of experience with clients around the country and often have insight into who best can deliver reliable Internet connectivity in your area Visit the Telos Website for the latest information The short list below was created in December 2010 and the marketplace is changing rapidly These are companies we or our clients have had direct experience with and report positively on their experiences Most of these providers will allow you a free period to test their service in your environment your ISP with your switch etc Consider that some services work better in some areas than others Usually it s wise to choose a provider close to you p
22. VSet or activate that feature from VX producer In this example that s what we did EA Active Fixed LW Channels Logging Information ld Studios LW Channel Name Calls 1 in 3198 out 3198 S1 Fix1 2 in 3199 out 3199 SL Fix2 Test Calls Devices Stream Statistics Script Manager Script Information Misc OManager Info Active Selectable LW Channels ld LW Channel Name Calls 3 in 3111 out 3111 S1 Sell 4 in 3112 out 3112 S1 Sel2 Program on Hold LW Channels System Status LWChannel Direction Calls Logs 0 From Source Backup Restore Firmware Acoustic Echo Canceller Disabled Devices No devices attached Lines State Comment Position Name Local Number Remote Number LWChannel Device SID Status OK Registered IDLE 1 2222 2222 2 2222 2222 OK Registered IDLE 3 2222 2222 OK Registered IDLE 4 2222 2222 OK Registered IDLE 5 2222 2222 OK Registered IDLE 6 2222 2222 OK Registered IDLE 7 Hot l 5600 3198 3198 OK Trunk IDLE 8 Hot 2 5601 3199 3199 OK Trunk IDLE 2009 2011 Telos Systems Make a test call on your VSet Pick up a line using your VSet and dial 1234 and press the GO button If you are using PBX in a Flash you should hear a congratulatory recorded message from the Asterisk Then try dialing 2222 One of your 2222 lines should ring and you ll hear a ringback tone in the handset or on the studio output Congratulations VX Tech Cheat sheet
23. VX and just to check for signs of life You deserve to experience some action after all that work loading and configuring Click on Extensions and then click on Add Extension The Device field will default to Generic SIP Device Accept this by clicking on the Submit button Then Fill in the extension number you d like to use then the display name and outbound caller ID This is what callers will see when you call them from this extension Interestingly it can be just about anything Some phone companies screen to be sure that you re sending a properly formatted area code and phone number while others will take anything a 4 digit extension a single digit your lucky lotto numbers whatever E O 9 nuvnz 168 0 248 s260 meru chpTid admin rv a amen Getting Rated Latest Headlines y JSlashiot EjWeather El 4U Server Short De Ania shared GStumbie Aly 4194 qiliteis Y A El Shrev info Favorites Stumblers Tools v FreePBX as PRA ete SA cn A E _ _ _ A_ _____ ___ _ __ ___ _ _ _ gt gt gt Tools AS Add SIP Extension FreePBX System Status Add Extenson Mette ponen Joe lt 5600 gt 27 sde Tos Joe VM lt 5655 gt ob hy SP Phone 3 5656 lt 5656 gt l erasers Foaure Cons User Extension 2222 A lt 5666 gt Gereral Cetirgs Detplay Nane Wat Extension Sent SERA Outacurd Rates CID Num Ales Trunks SIP Alas Scrolling down the page some more we see so
24. Zap Channel DiDa Tak Uno Announcements Alar Into 7605138255 any Siden CID name prefix co The DID number for our example Talk Line is 7605138255 We leave the Caller ID Number blank because we want it to work from ANY number and entering a number would cause it to ONLY work from the number in that field Rumor has it that a FreePBX programmer added this feature so that his X Girlfriend could get special treatment like being sent to a special ring forever extension We don t need this right now but it could come in handy later pos sibly depending on your personal situation Leave the rest alone and scroll down further e Eh View Higtery Reokmarks pols tep O Q mew inten geamoo com s0comens pro ra admir JEA a D Mest Visited Boeing Rated latest Headlines gt BSieshdct ColMeil Login New usenet group Flowroute DIOS _ Damiel Aly 6195 bilikeki P Ej Sue info Favorites Dstumtiers Tools y FBS Tools ois Engish x CARs Edit Route FreePSX System Salus Module Asmin Delete Roule 9_cutside Add Route 27 Toot a5 Route Name 9_outsido tesne ES SIPSTATION Route Password Extensions Emergency Dialing gt a extensions Festaro Codes Inta Company Route 2 AAA Music On Hold imit aa Tate Dial Pasems a ol Arama E inoound Roses dd Zap Coarne DOs 23 Arrourncemens Oeni emer asians 5 tandem net Blast Dial pa ems wizards pick one
25. access the line for an outgoing call In many other countries an audio tone is sent on the line to indicate the end of the call This lack of disconnect supervision results from the idea that there was no reason for a central office to hang up a phone by remote This was a physical human action performed by someone who knew that the conversation had ended and put reacted by replacing the receiver back in its cradle The design was never intended to work for machine to machine connections NOTES RESOURCES ADDITIONALINFORMATION 53 Actually there are two types of off hook signaling On a loop start line when a phone goes off hook the circuit is closed and the central office detects the change in current This is the common residential format Ground start signaling is a small modification to the scheme to permit disconnect supervision and remove the possibility of glare where a PBX mistakenly takes a ringing in line for an outgoing call In an idle circuit the central office provides 48v on the Ring wire and an open on the Tip wire From the PBX side Ring is grounded first then the central office circuit must respond by grounding Tip The PBX senses this releases its ground and maintains the connection by drawing loop current E amp M Trunks E amp M trunks use two extra wires for signaling the so called Ear and Mouth connections These solve the problems with glare and disconnect supervision This scheme is nearly obsolete but oc
26. also decide to eliminate walls full of couplers for pre delay IFB dial up lines and Transitioning your Telco service might save you a lot of scratch We seen stations saving thousands of dollars a month no kidding by eliminating POTS lines with their taxes and fees Anything else cool about the VX Did we mention the color LCD user interface on the new VSet phone control surface Produc ers and talent love it Fancy gear like this has to be trouble no VX is simpler than a multiple box approach With fewer components it s more reliable At the time of this writing no VX engine at a radio station has crashed ever We ll probably have a contest to see who has the longest uptime Right now it s 6 months but that system was installed 6 months ago NOTES RESOURCES ADDITIONALINFORMATION 97 Application Example WKSU WKSU is a non commercial FM radio station affiliated with Kent State University It features NPR APM and PRI programming classical music regional news and on weekends folk music The station serves the Akron and Cleveland radio markets with its main FM service and six repeaters The main FM and four full power repeaters each host four full time HD Radio chan nels a simulcast of the main service a folk music channel a classical service and The WKSU News Channel All of the programs are also streamed for Internet listeners The office and studio facility is housed in a stand alone building at t
27. and use a digital connection either IP SIP ISDN PRI or T1 E1 All are four wire maintaining send receive isolation ISDN PRI is the usual choice in both the USA and Europe Following is a step by step guide to setting up Asterisk With the hardware in place you are ready to move on to getting the software installed and configured 78 Section5 1 Get your distro The most popular are at www digium com pbxinaflash net www trixbox org www freepbx org Download your distribution We ll proceed assuming that you re using PBX in a Flash hence forth PIAF It s a few hundred megabytes Once you have it saved on your hard drive you ll need to burn it to a CD It ll have a file name that ends with iso and you ll have to use your CD burning software to burn an ISO image This procedure varies depending on what burning software you have installed on your machine Nero etc so if you haven t done this we sug gest googling the search term burn an iso image with and add your burning software name at the end Many people have had difficulty with this process in the past and you ll find clear instructions there It s important that you burn the image and not just copy files The CD that you create will boot up by itself and will wipe out anything on the installed hard drive Before booting check the bios in your machine to be sure that CD ROMs are bootable If the machine is recent no action shoul
28. are designed so that this does not happen For example almost all phones gateways and SIP Telco services have G 711 as a supported codec so this is an insurance policy that two endpoints will find common ground Within a PBX system designers usually choose one codec as a standard for the system and stick with it for all connections For example the Telos VX studio system uses 8kHz sampling rate 16 bit uncompressed PCM internally for all calls that connect to the PSTN 74 Section 5 The State of SIP and its Future SIP is not often used as it was intended by its developers Most PBXs that have SIP interfaces don t use SIP servers at their core All use their own rough equivalents designed independent ly So what went awry The SIP schemers were certainly far ahead thinkers who wanted their protocol to support rich media mobility portability sophisticated endpoints etc The problem seems to have been a lack of a certain practicality For example consumer PC to PC VoIP products needed to solve the problem of firewall and NAT traversal which has been addressed quite slowly within the SIP working groups Meanwhile Skype s developers solved it quickly and effectively Then there is the problem of supporting all the features a vendor wants to employ to differentiate its product It s unsurprisingly faster to just implement it your own way rather than waiting for the idea to make its way through a committee who might well not se
29. be connected to an existing Livewire network or to a new dedicated network that you create to serve only the VX system which could be as simple as a single Ethernet switch For audio to work there must be at least one Livewire device capable of supplying clocking on the Livewire network Normally this would be a Livewire audio Node The Axia Element console Powerstation can supply this clock More on the Network An appeal of the Ethernet IP approach to building studios is that you can make them as simple or elaborate as your needs require The components of a VX system the Engine VSet phones console controllers and PCs are networked together using standard off the shelf Ethernet switches A small VX system might have only a single Ethernet switch as the network infra structure while a large full facility setup could have dozens of switches and an IP router or two The Ethernet switch on the Engine s LAN port must be Livewire capable That is it needs to support multicast Switches that are not multicast capable usually flood multicast traffic such as Livewire audio streams to all ports potentially overwhelming devices like PCs and printers So you don t want to plug the VX LAN port into an office network after Livewire outputs are enabled Multicast capable switches will not propagate the Livewire traffic to ports that are not subscribed to a particular audio channel Thus blocking the high volume traffic from places it i
30. connections are made In a non Axia studio route the AEC connections to a LW Node by assigning the appropriate LW channels at both ends and then connect the audio to from the mixing console via analog or AES3 to the Node To check if an AEC is working call into the system and switch on the mic and loudspeaker Adjust the mic level to normal and the loudspeaker to reasonable volume Talk into the phone listen for echo and assure that none is audible Telco Services and Interfaces VoIP SIP Trunking While it remains a niche in early 2011 SIP trunking is growing rapidly in support from both PBX vendors and carriers Over time this will almost certainly appreciably reduce the use of the older POTS and T1 trunking Eventually it may replace it completely Whether the gateway to the PSTN is at your physical location or at another site should make no difference as long as the IP path between you and the gateway has guaranteed QoS with sufficient bandwidth to support the maximum number of active connections you expect to have In the case that the IP link is to be used for both telephony and data the system must be designed so that phone calls have priority In order to ensure this there must be only one IP vendor between you and the PSTN and this vendor must guarantee QoS in a properly written Service Level Agreement Any time that IP service crosses from one vendor to another all bets are off as to both the probability of achieving consiste
31. es 9 The Engine Network o o o e 2 9 The front panel OLED knob interface 9 Network Connections lt s osso cresci nunata tewas 10 MoreontheNetWwoik s i a 0 ee ee ns 10 VSet Installation s sceso ee a 11 System ConfiguratiON o o o o ooo ee 12 Selectable and Fixed Lines o 12 Studios and Shows ssor tanana gean A 12 Configuration Web Pages 2 aaa 13 HomePage coord E ER on ee 4 13 Main page General Configuration oaa 14 Default Username Password and IP settings 15 SIP Configuration e 00 Serene wy wg alae OR a 16 RegardingTheVXandSIP 2 ee ee 16 Server Configuration o o ooo o 18 SIP and Network Address Translation NAT 19 Overall Studio Configuration Page o o 19 Individual Studio Configuration Pages Part 20 Individual Studio Configuration Pages Part2 21 GPIO t c ral dd 22 Overall Show Configuration o o o 23 Individual Show Configuration Pages 23 VX in the News and Production rooms o o o 24 Using the VX to replace Couplers 2 ooo oo 24 Call Audio Processing page Audio Processing and metering 25 Assigning sounds 8 Tones The Tones page 26 SIP and DIME p eaei cla a We ee 28 LOGOO e 6 ne oo o Gia le a de bo a Be bd 2
32. hybrids faders 2 Call Control module for 12 or 24 lines 4 hybrids faders 56 Section 5 Note that a Call Control module is required even when no selectable lines will be configured The fixed lines rely upon software linked to the Call Control Module Configuring the Element to Control the VX Element configuration is via web Do the VX configuration first and then move on to Element configuration Things you will need to know about your VX in order to configure an Element as a controller are The VX Engine IP address User name and password to access the VX Engine The VX Studio name you will connect to VX hybrids Livewire output names and numbers you want to use in this studio VX Show name if you want to load VX Shows from Element console profiles In order to correctly control your VX your Element and or PowerStation must be be running the following software versions 4 PowerStation v1 1 3a or higher Element v2 5 0 3 or higher To obtain and install this software visit www AxiaAudio com downloads and follow the update instructions contained in the accompanying Release Notes There are three steps to configuring the Element for the VX as follows 1 Setup the VX to Element connection Open the Element s Module Manager Phone Channels web page If you have the Caller Controller Module installed you will see the following at the top of the screen Phone channel configuration ID Module Server Ad
33. if it is referenced from at least one show The Lines Refs column shows the number of lines which are explicitly configured and the total number of line positions in shows using this server Clicking on the link with the server name will bring you to the server configuration page As of VX firmware version 1 1 0 there are only two global options The Default Server changes the server address which will be used for new lines in Show configuration Note that this will affect only new lines existing ones will keep their old settings use the S P Server option to change the server address for existing lines Enabling SRV Lookups makes the VX to strictly follow the SIP standard when resolv ing domain names This feature is not widely used and can be left unchecked unless required by the SIP provider Regarding The VX and SIP Some details regarding the VX s SIP implementation As of the time of writing the VX supports the g 711 and g 722 codecs g 711 is the usual PSTN codec so ISDN or T1 connections will not suffer any transcoding loss SIP auto INSTALLATION AND CONFIGURATION 17 matically negotiates and selects a codec that is supported by both ends Implements RFC 3261 standard but see below The 9 722 codec is the same codec known to broadcasters from the pre MPEG days of ISDN remotes It has 7kHz audio bandwidth opening the door to much better than usual speech quality especially from mobile handsels When there is
34. is OFF Talkback will be the source Alternatively Feed to Source can be fixed to P1 P2 P3 P4 SA SB SC or SD You can assign a line to a specific fader by pushing the Set OFF ON Hold or PVW button on the fader channel before selecting the line The fader channel alpha display will toggle between source type and LINE N after a line is assigned 4 The VX Producer Application Introduction The VX Producer application integrates 4 Call screening and production communication Recording playing and editing calls A softphone for talking with callers off air Normally it would be used by Producer call screeners to communicate with callers and the on air talent Each would have a copy running on a PC near to them Installation The standard Windows installer is used to install the VX Producer application and the Net framework if needed It also can install the Audacity audio editor Remember that since the VX uses Livewire audio to carry the handset audio to your VX producer PC that the PC needs to be on a LAN subnet that carries Livewire traffic the same subnet as the VX engine Network LAN port You ll need at least Windows XP Service Pack 2 TS General Recordings Audio Chat Name Studio Chat Color Black VX Engine IP 192 168 2 101 Editor Path C Program Files Audacity audacity exe Choice of recording Phone only or Phone plus local mic 42 Section 4 a One Micropho
35. like any Axia device and can be backed up and restored in the same manner When restoring either from backup or to factory defaults it is possible to keep the current network settings intact Note that VX will automatically restart to apply the new configuration VSet Phone Configuration VX Phone gt D amp mm Pager Safety y VX Phone 0 9 7 181 build Nov 27 2009 17 59 10 Take screenshot png Screenshot Warning System will pause while sending screenshot Screenshot Backup config xml Restore config xml Browse Restore Notice If the new config has different IP address you ll have to reopen the web page from the new IP Firmware Upgrade Firmware image Browse _ Upgrade Notice Firmware will be uploaded to inactive bank Firmware Version Bank 1 ver 0 9 7 164 build Fri Oct 16 11 14 20 2009 Bank 2 ver 0 9 7 181 build Fri Nov 27 15 31 06 2009 active Warning System will reboot after changing active bank User password Old password New password Repeat new password 3 Operation VSet Operation Select Studio and Show If these have not been already selected this should be done as the first step before using the VSet Press the Menu button to access the menu functions The LCD will show the various items that can be changed Select the studio and show you want and then exit the menu by pressing the Menu button Select studio 1 Studio 1 Sele
36. reliable and sufficient IP bandwidth audio quality from mobile phones can be much improved compared to the usual fuzztone We ve tested the iPhone application Media5 FON with satisfactory results but only the upgrade version includes g 722 The Acrobits app should also work On Windows and Linux the free Ekiga VolP softphone works fine Chat Edit View Help sip 500 ekiga net Contacts Dialpad Call history Local roster Y Ekiga 2 4 e Fabrice Alphonso Do Not Disturb Please t home Y Office PBX 2 2 Phone 91 Connected with 500 ekiga net O Phone 92 Call Duration 00 00 20 Y Services 1 1 Echo Test Network neighbours At home wil 4 7 8 8 0 V 6 7 0 0 FPS 29 0 For both of the above you need to force the use of the g 722 codec by enabling it and disabling the others on the VX side everything is handled automatically during normal SIP codec negotiation Following the usual procedure the codec choice is made by the caller and conveyed in the SIP Invite message and procedure SIP connected g 722 audio is via IP so the VX Engine needs a public IP address that is accessible to the calling side G 722 does not pass via the PSTN and gateways which means that ISDN g 722 codecs will not work with the VX RFC 1890 the original standard governing audio codecs in SIP erroneously listed the clock rate of G 722 as 8kHz the actual sampling is 16kHz When the error was discov ered it was too late to
37. s the ultimate You start with the most flexible console audio platform and then add smoothly integrated phones with the IP network powering it all Sweet The network delivers any of your Telco lines to any of your studios in any combination Any line is available in any studio at any time 96 Section 5 notice the VX engine has both a LAN and WAN connector why is that It s a built in firewall isolating the VoIP connection from your studio network We use this same approach in the iPort Livewire WAN MPEG gateway What about call screener and database functions A basic Call Screening app VX Producer comes with the system Other networked PC based apps such as Broadcast Bionics PhoneBOX VX or NeoGroupe s applications put information about your callers in front of your producers without the need for caller ID boxes serial cables or other hassles What about SMS messages and chat Can they be integrated into my phone system Using Broadcast Bionics Phone Box VX yes Telos has always built open systems to allow others to create their own visions around our gear OK so the catch has to be the price The VX lets you leverage cheap networking to serve your entire facility Since you don t need hardware boxes for each studio cost is surprisingly reasonable You ll use the VX in your on air studios to replace older multi line systems and you ll use it to replace hybrids in news rooms and production studios You might
38. 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 incidental 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 equipment 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 Sy
39. such as for T1 line interfaces Choose a server grade motherboard We usually go with on board video to reduce costs The CPU need not be the biggest baddest fastest thing out there but it should be in the sweet spot with regard to price performance We often put an extra Ethernet card in the machine or buy a motherboard with two Ethernets on the board One is configured one for the inside LAN and one for the public WAN similar to the VX 2 4G of RAM is fine but more is better Hard disks should be server grade and kept well cooled for longest life The goal is to assemble a machine that can run for 5 years or more We are aware of Asterisk servers that have run continuously for three years or more and hardware that is still in service after 15 years Server grade doesn t mean exotic RAIDs or multiple processors a low end basic server will be fine You ll find that the more common the hardware the better supported it will be by Linux and the fewer problems you will have with compatibility Stay away from Bleeding Edge hardware but spend a few extra bucks on a better CPU heatsink fan or power supply We ve learned over the years that the most common failures are fans power supplies and optical disks If you treat hard disks well mainly keep them cool they ll be more likely to last Also many newer power supplies have an 80 logo This means that they are better than 80 efficient which means l
40. telephone When wideband audio is fed directly into a telephone connection there are often complaints that it is hard to understand because the low frequency energy is masking the higher frequencies that are needed for intelligibility A Sample Rate Converter is an essential part of the signal processing The SRC is automatic and there are no controls or configuration The SRC between the telephone processing and the studio side audio adapts the telephone sampling rate to the 48kHz Livewire rate and allows the switching and mixing to be performed synchronously within the VX Engine The SRC adapts a number of rates on the telephone side The usual telephone rate is 8kHz but newer wideband codecs making their way into telephony will have higher rates The output to Telco rate of the send side SRC is locked to the receive rate on the same line Assigning sounds amp Tones The Tones page 4 MA E htteyi92168 0 9 sFuntonectg G w TelosVX Control Center Telos VA TelosVX Control Center Confi ti telas seh Oo o Tone Configuration SIP Studios Shows Call Audio Processing Note Uploaded tones must be in AU file format 48kHz linear PCM Call Progress Default Tones USA s Change DTMF Ringtones Dial Tone penu Logging Browse Upload Disable Information Default Studios Ringheek Tane Browse Upload Disable Calls Default Devices ee Husar Browse Upload
41. the audio processing features of VX such as AGC EO ducking etc all at once Enabling this option will let you adjust the individual options below Ducking The purpose of this is both to improve the performance of the system with regard to feedback and echo as well as to provide an effect that many DJs and talk show hosts prefer that the caller level is reduced when they speak The ducker reduces the gain in either the send or receive path depending on who is talking at a given moment There is a smoothing filter with time constants tuned to make the effect as natural as possible The amount of ducking is a user adjusted variable value and may be defeated entirely if you choose 26 Section 2 Caller AGC EQ Normalizes the level coming from telephone calls A wideband AGC is used for consistency followed a multi band frequency selective processor that provides spectral control for the best caller audio balancing intelligibility and warmth The Duck Control module communicates with the AGC module so that gain is not increased when the studio side is sending audio Leakage from the hybrid in this case could look like low level telephone audio that needs to be boosted If this happens the effect of the hybrid and ducker would be undone Send EQ This is a simple EO that rolls off the low end and has a peak at high frequencies to improve intelligibility This is designed to make the audio similar to what a caller would receive from a normal
42. the result 30 Section 2 With Log to Syslog log messages are sent over your network to the Syslog server Enter the IP address for your syslog server Syslog servers need not be Linux machines You can install a syslog server on a windows PC One simple Windows app is at http www kiwisyslog com A typical setting for diagnosing SIP problems would be to set the log level to Debug extended Fewer log messages will be generated if you set level to Debug basic Easier to read but maybe missing detail that you need If you want to adjust logging in detail go into the Advanced page and select which modules to log then select the DEBUG level for the module you want to trace For example if you want to trace what VX Producer or VSet is sending to the Engine you might chose to select LWCPBE module log level to DEBUG The three most interesting modules to log are AIF VX front end communications with VX DSP part CPSIP SIP SIP messages LWCPBE LWCP protocol for VX control Internet standard Network Time Protocol NTP is used to provide an accurate timestamp to the events captured in logs For accurate timestamps you need to have an NTP server accessible on your network and its IP address properly configured on the System page gt TelosVX Control Center C fi 192 168 0 7 log Telos VA TelosVX Control Center Configuration Main SIP Studios 6e787b90 Mar 8 07 22 21 493 WARNING tact_receive R
43. with very little of the caller audio making it back to the other end They work with up to 20kHz audio bandwidth so are ready for the wideband VoIP codecs now coming online And they solve a longstanding problem Older time domain AECs depended upon the acoustic path remaining invariant and could quickly degenerate into feedback when a microphone was slightly moved or the acoustic path changed from some other cause The frequency domain technology used by the new AECs works just fine with moving microphones and other echo path changes This new AEC technology is particularly useful for TV studio applications where it can be impractical to have talk show guests using earplugs Today s high performance AECs let talent and guests listen to phone calls on foldback loudspeakers AECs are provided within the VX Engine but they are not connected into its internal signal paths This is because they need to be placed in the signal paths between the studio loud speakers and microphones With Axia mixing consoles the insert and return points are provided with Livewire channels and are installed by configuring the appropriate channel numbers The AEC can be used with Non LW consoles via LW Nodes connected to the appro priate audio signal paths AECs are possible from a single VX Engine Additional Engines could be used to provide more Please contact us to discuss your options should you want to do this There are three connections for
44. 2 1 gt Content Length 126 This is how Steve s SIP client would signal to Skip s that he wants to connect and speak with him SIP works together with several other protocols and is only involved in the signaling portion of a communication session SIP is a carrier for the SDP Session Description Protocol which describes the media content of the session e g the codec being used the bitrate etc SIP provides the following capabilities Determines the location of the endpoint SIP supports address resolution name map ping and call redirection Determines the media capabilities of the endpoint i e which codecs are available and supported During a negotiation SIP determines the best codec that can be used by the parties on the call Determines the availability of the called endpoint If a call cannot be completed because the target endpoint is unavailable SIP returns a message indicating this and why Establishes a session between the originating and called endpoints if the call can be completed Handles the transfer and termination of calls SIP supports the transfer of calls from one endpoint to another During a call transfer SIP establishes a session between the transferee and a new endpoint specified by the transferring party and terminates the session between the transferee and the transferring party The Parts of a SIP System Like most things based on IP SIP was designed to be modular Implementers c
45. 9 Firmware updates 2 o ee 31 Backing up and restoring Engine configuration 31 Backing restore configuration settings o 32 VSet Phone Configuration o o o o o o oo ooo 32 Operation 33 VSet Operation oa aa A 33 Select Studio and ShOW o 33 Set Talent or Producer Mode o o o o o o o ooo oo 34 The Line Info Display 2 ee 34 Next indicator sas bie eg a ale eed th baw ok ae hw ee A 34 Fader numbei s aoi ace oe RI A oes 34 LING NAME even a hoe ees SG Pa Se 34 Gallet oia wae be eA wee ee Ee eR RS 34 TME ss ae BE NS ww are eee eR Wa hak ae eR 35 TELOSVX MANUAL v Line Status icons ss 4 4 6 4 is ee be ee Sd ee wee ee 35 Line Button Columns y er e les ek a a e 36 HOMMIBUON Scat eb Ae ee ae ech A oh ek a leg 36 Drop DUTTON geesi o o 37 LOS a kias a a e E a Eea TE Ka N on ae eee eS 37 Next Button and Function 2 aaa 37 BIOCK Allie ihc aea mat acted e a ea a a At a 37 Numeric Keypad lt oo aoee eeen temeer eee 38 Re dialFunction is sorse a a a oa a a ns 38 Fader ASSION cor moata bh EADS do Cee 38 Firmware Upgrades 2 ee a 38 Tak it Easy ora ead ae eee se Peas 39 Operation with the Element Console 39 4 The VX Producer Application 4 NtrOdUCtION as wea eee ee REED A AR EEE OR EO 41 Installation ooo on a ara ee sr wa a 6 41 Set up Studio and Show oo oo oo 43
46. At the shell command prompt type passwd master and then lt enter gt You ll be prompted to enter the password twice If they match the password is set Save these or choose passwords that you won t forget Do the same for the following commands passwd maint passwd wwwadmin you ll use this one most often passwd meetme passwd webmin 82 Section 5 Turn some services off for now At the shell prompt type disable iptables and then lt enter gt then type disable fail2ban and hit lt enter gt These are security tools that could get in your way until you re fully set up If you run your machine behind a router or firewall you won t need these anyway 3 Configure Asterisk Asterisk needs to be configured for your application There is plenty of information on this topic on the Asterisk site In brief the steps are 1 Set up trunks 2 Set up outbound routes We ll work through a simple example in the following pages We prefer to use SIP extensions rather than SIP trunks to connect Asterisk lines to the VX because they are more flexible For example station SIP signaling conveys caller ID which the VX can display and use SIP trunks on the other hand do not pass CID The VX supports the required SIP registration when you enter the authentication password on the show configuration pages By default Asterisk uses SIP Reinvites This causes the VX or VolP phone to make a direct connection to the Telco b
47. Contest sem A We enable the Vmx Locater This is useful as it allows callers to be transferred to number inside or even outside of the system if 0 1 or 2 is pressed Here 0 is programmed to send callers to the station operator 1 send the caller to our imaginary cell phone and 2 transfers the caller to our imaginary personal assistant at extension 5667 If you do use these options be sure to mention pressing 0 for the operator in your outgoing voicemail message You probably don t want to use these optional features for request lines or public call in numbers but they re there and work well if you have the need If you re happy with all of the above click on the Submit button The entry will be created and stored but not activated until you click on the orange bar at the top of the screen that appears when ever you have unapplied changes Don t worry applying changes won t drop any calls or interrupt any conversations in progress Trunks Since we now have an extension let s make it so you can call somebody or get called To do this we ll need to way to and from the Public Switched Telephone Network or PSTN The pro vider of SIP here was Vitelity who provides cut and paste trunk set up data on their website for subscribers Most providers do this The next screenshot shows how Vitelity suggests that their trunk be set up NOTES RESOURCES ADDITIONAL INFORMATION 87 COQ M
48. DEBUG Via SIP 2 0 UDP 127 0 0 1 5101 branch 9hG4bK 131 31 515 DEBUG Content Length o 31 515 Prom 11 lt sip 11 195 13 179 146 gt tag 3131013326 Accept application sdp User Agent friendly scanner Firmware 6e787b90 Mar 6e787b90 Mar 6e787b90 Mar 6e787b90 Mar 6e787b90 Mar z To 11 lt sip 11 195 13 179 146 gt z Contact sip 11 195 13 179 146 z CSeg 1 REGISTER ACK z Call ID 1592768955 Max Forwards 70 6e787b90 Mar 6e787b90 Mar 6e787b90 Mar 6e787b90 Mar 6e787b90 Mar String to IP Address OK XPORT ADD received z received 81 94 239 150 6e787b90 Mar 8 07 33 31 515 DEBUG TACT_RECEIVE method serv_cb INSTALLATION AND CONFIGURATION 31 Firmware updates TelosVX Control Center Hardware and Firmware Firmware Upgrade Hybrid New firmware image Browse Upload Tones Logging Note All available firmware slots are used Please free one before uploading new software Information Studios Calls Firmware Version Devices Firmware Information Control Stream Statistics Script Manager Script Information Slot 1 Misc TOManager Info System Slot 3 ver 0 4 1 0 r7789 k2 6 17 14 pce 166 Fri Dec 03 2010 12 39 08 Status Logs Slot 4 ver 0 4 3 0 r8165 k2 6 17 14 pce7197 Thu Mar 03 2011 11 48 57 Backup Restore ver 0 4 0 0 17716 k2 6 17 14 pce 166 Wed Now 17 2010 16 00 34 Slot 2 wer 0 4 1 0 r7771 k2 6 17 14 pce 166 Wed
49. DP RTP ENUM and UDP We know that engineers are lifetime learners and encourage that However just as you prob ably don t know much about SS7 or IUP in the telephone network understanding these details is optional We do have White Papers on our web site to educate you on these and other terms Start with the one here You could read Steve and Skip s AolP book too for a fun and comprehensive coverage of this stuff Here s a paper on the adaptive IP codec http telos systems com techtalk Advanced 20 Tech 20for 20IP 20Remotes_Church_Final pdf Does Livewire technology come in to the VX picture Yup The VX takes VoIP on the Telco side and Livewire AolP on the studio side This makes integration with Axia consoles and networks easy and efficient If you don t already have a Livewire network you would use Axia analog or AES audio nodes to provide I O in either for mat Each node provides eight stereo inputs and eight stereo outputs to and from the system Each Axia GPIO node provides 8 groups of 5 inputs and 5 outputs covering the needs of 8 studios Telos Support is always available to help you specify exactly what you need If you are new to Livewire Technology you may wish to skim through our Primer here http www axiaaudio com manuals files IntroToLivewire2 1 pdf So the two can live together side by side on the same LAN Yes they can So let s talk caller audio quality What does VX offer compared
50. One step up would have you pressing the right column line buttons to select the calls you want one at a time Again when you are finished taking calls you would press Drop You could use the left column line buttons to talk with callers on the handset At the next level you would use either the lock function or the fixed line capability to confer ence calls Some Common Questions Question can t place a call After dial nothing happens What s up Answer Don t forget to press the green Go button after dialing Question Why don t hear any audio after pressing a line button in the handset column left column Answer Double check that the handset cord is plugged into the handset jack not the headset jack on the VSet Question Why can t put a call on the air When press the right column line key nothing happens Answer Put the VSet into Talent mode When the VSet is in Producer mode calls cannot be placed on the air from that VSet so as to prevent accidental call drops or placing the wrong caller on the air Your opinions and ideas are always very welcome at Telos Please share them with us at sup port telosalliance com Operation with the Element Console The Axia Element console can be used to control both fixed and selectable lines from the VX Entering the VX Engine IP address connects all the needed logic functions line selection is the main one but muting for t
51. PSTN aK gt Calis 2417225 Loop Current Analog audio Talking 2 Taking Hangs up Loop drop Loop drop Disconnect A SIP call set up to the PSTN via a SIP server and a gateway to POTS lines SIP messages may be carried by UDP or TCP SIP has its own built in reliability mechanisms so it doesn t need TCP s reliability services Most SIP devices such as phones and PC clients use UDP for transmission of SIP messages PBXs on LANs almost always use UDP because LANs don t drop packets and there is no need to incur the overhead of TCP Transport Layer Security TLS protocol is sometimes used to encrypt SIP messages TLS runs on top of TCP This is the protocol used with HTTP to make the secure HTTPS used for secure Web transactions Not shown in our transaction examples is the media negotiation that is part of the INVITE 200 OK ACK sequence Through this process endpoints decide which codec to use The Session Description Protocol SDP defined by RFC 2327 is the way codecs are offered and hopefully accepted by the other end Usually the caller sends an SDP message along with its INVITE listing the codecs it is prepared to use The far end chooses one of them and tells the caller which it prefers in the 200 OK response The caller can let the far end propose a codec by not sending an SDP message in its INVITE It is possible that the two endpoints have no codec in common and the connection is unable to proceed but systems
52. Support Telos Systems com Inquiry Telos Systems com Notices and Cautions This symbol wherever it appears alerts you to the presence of uninsulated dangerous voltage inside the enclosure 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 equipment 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 cause interference If it does the user will be required to eliminate the interference at the user s expense NOTE Obj
53. Te los VA Multi Studio IP Phone Interface System Telos Y USER S MANUAL Version 2 0 1 May 2014 Telos VX Manual 2011 2012 TLS Corporation Published by Telos Systems TLS Corporation All rights reserved Trademarks Telos Systems the Telos logo and VX 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 year limited warranty the full text of which is included in this manual Updates The operation of the VX 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 VX or this manual In the past many good ideas from users have made their way into software revisions or new products Please contact us with your comment
54. VX using Ethernet audio remains clean Many channels of audio and SIP signaling are interfaced over a single Ethernet link Telos has people on staff and access to consultants who are experienced with Asterisk and who can help you with regard to integrated Asterisk VX installations Feel free to call us with your questions and suggestions Here are some ways our clients are using Asterisk To automatically play a disclaimer or informational message to callers before your producer or talent answer As a programmable ISDN PRI gateway Use with the PRI T1 that feeds your office PBX and redirect some of the DID numbers to your VX or to a voice mail Interactive Voice Response system running on Asterisk To block unwanted callers by implementing a blacklist based on caller ID 4 To create your own off site extensions over the Internet It s great for news bureaus sales offices or even reporter s homes To create a private tie trunk network between co owned stations to reduce costs To have a backup Telco service by ordering a few inexpensive SIP lines or trunks to use in case of loss of a T1 To get rid of expensive toll free numbers or foreign exchange lines Order SIP lines from a distant city that your station covers SIP lines can cost as little as 1 50 a month Also SIP lines don t have all the extra charges for city taxes and other junk fees 4 Use the Asterisk call detail reports as a resear
55. an be used to secure the Engine from being controlled by unauthorized devices The VXset for example uses this password By default this is set to lt blank gt INSTALLATION AND CONFIGURATION 15 WAN Services chooses which services are available on the WAN port HTTP is for web access LWCP is for controllers and SSH is shell access to the Linux OS in the Engine SIP VoIP are always enabled on the WAN port when the port is enabled NTP Network Time Protocol is the Internet standard for keeping devices synchronized to the correct time The VX uses this to have accurate timestamps for logged events Enter the IP number for the NTP server here An NTP server must be accessible on either the LAN or WAN port The VX automatically finds the correct port If the NTP server is outside of both the LAN and WAN local network segments gateway must be set You can use a domain name if a DNS server is configured and accessible Under NTP Config are the settings for your location Default Username Password and IP settings We put this info here under its own header just to make it easy for you to find when surfing the table of contents User user Password lt blank gt that is no password Network interface IP 192 168 2 200 netmask 255 255 255 0 state for the WAN connection is OFF by default with no IP set Active ports on WAN interface by default are 5004 DSP Engine 62000 through 62513 RTP 5060 SIP
56. an example of URIs In SIP a Request URI is defined to indicate the name of the destination for the SIP Request INVITE REGISTER etc URLs Universal Resource Locator describes the location of a resource available on the Internet For example http www telos systems com is the URL for a Web home page It is resolved by DNS to a concrete IP address PSTN telephone numbers are sometimes called E 164 numbers a designation applied in an ITU T standard that describes the format of telephone numbers to be used worldwide ENUM 72 Section 5 E 164 NUmber Mapping is the Internet service used to look up the URI associated with a particular E 164 telephone number It s part of the DNS system SIP can use ENUM to locate the VoIP system associated with a telephone number that accepts incoming calls How SIP Works As we ve seen SIP is a simple text based protocol It uses requests and responses to arrange for communication among the various components in the network and ultimately to establish a connection between two or more endpoints User B User A IP 18 18 2 4 Es s Calls 18 18 2 4 INVITE sip 18 18 2 4 180 Ringing Rings Answers Talking Talking Hangs up SIP at its simplest An IP phone calls another directly But this is not the real world Almost always there are SIP servers of various kinds in the picture When a user initiates a call a SIP request is sent to a SIP server either a proxy ora redirect server
57. an pick and choose among the following elements to build the system they need SIP Clients Sometimes called User Agents or endpoints These can be hardware phones or softphones phone applications running on PCs Gateways When needed gateways translate between the IP network side and the switched circuit Telco side providing physical electrical signaling and audio interface Proxy server Receives SIP requests from a client and forwards them on the client s behalf Basically proxy servers receive SIP messages and forward them to the next SIP 70 Section 5 server in the network Proxy servers can provide functions such as authentication autho rization network access control routing reliable request retransmission and security Redirect server Provides the client with information about the next hop or hops that a message should take and then the client contacts the next hop server or client directly Registrar server Processes requests from clients for registration of their current loca tion Registrar servers are often co located with a redirect or proxy server While SIP enabled endpoints are able to connect directly to each other SIP servers provide a number of valuable services including the following Register SIP client devices Register individual human users for access to their services Perform authentication authorization and accounting when needed Look up the address of the far endpoint
58. any many local exchange areas and toll free and foreign numbers as well Local number portability and SMS messaging are also offered 68 Section 5 SipStation with connectivity provided by Bandwidth com they ve proven to be very reliable and low cost 8x8 Based in Silicon Valley they deal mostly in hosted PBX service Reliability is very good Flowroute Based in las Vegas offers very attractively priced DID numbers and special deals on blocks of 20 50 and even hundreds of numbers Local Number Portability in the USA and Canada Telephone number porting is fairly common these days mainly due to the large number of wireless customers changing carriers and plans frequently and wanting to keep their phone numbers Each carrier has its own rules and policies for number porting with the average port taking from 10 days to 3 weeks It is important to note that high volume or choke numbers cannot be ported You can try but we haven t seen it work yet Usually the order will be taken a due date given and then on the day of the port you ll get a call from the utility or provider inform ing you of the bad news Porting is usually done in the morning but special arrangements can be made with the carriers to port at a specific time Once ported your old POTS lines are automatically canceled by the old company We have seen situations where CLECs and Wireless companies have some prob lems routing calls for the first few ho
59. call and data connec tion With DSL providing much higher data rates ISDN BRIs are moving ever closer to obsoles cence Nevertheless when they are available they could be useful for small installations that need only a few lines Most VoIP gateways have cards to interface with these lines As with PRI lines take care to set the gateway s configuration to match the type of signaling that your line uses SIP Compatibility VX implements the basic set of SIP functionality described in RFC 3261 together with the related standards RFC 4566 and RFC 3264 for describing and negotiating supported audio codecs RFC 3550 and RFC 3551 for audio transmission RFC 2833 codec for sending DTMF tones separately from audio RFC 2617 digest authentication We have tested VX with different SIP providers gateways PBX s soft and hardphones and other gadgets to make sure it works out of the box You will find a list of some of these in this very manual and on our web site We strongly recommend using VX together with a tested device but sometimes it might not be an option Telco provided SIP trunks are one such case Chances are that these will work just like that However it is not possible to test every SIP talking thing out there and different vendors tend to interpret some parts of the standard differently SIP started as a simple text protocol to set up phone calls but over it has grown quite complex While we strive to do the righ
60. casionally E amp M interface cards are used to connect music on hold to VolP PBXs They are convenient for this purpose because the audio path is transformer isolated and there is no need to supply talk battery from the MOH source T1 E1 These are basic digital interfaces to the switched voice network and are widely used This is especially so in the USA where T1 is nearly standard for large PBXs In Europe ISDN PRI is more widely employed T1 transports up to 24 voice channels while E1 supports as many as 32 T1s are common in the USA and Japan while E1s are provided by Telcos in most of the rest of the world In addition to the audio these digital circuits also carry basic signaling in CAS Channel Associated Signaling bits This signaling emulates loop start ground start or E amp M depending upon configuration Over a T1 23 speech channels are offered while an E1 provides 30 They actually have 24 and 32 channels respectively but one or two of the chan nels are reserved for signaling communications T1s can also be used for IP connections In this case all or part of a T1 s 1 544kbps capacity is used as a transparent pipe from the local IP router to the ISPs equipment The phrase channelized T1 is sometimes used to distinguish a T1 that is intended for circuit switched voice application A fractional T7 is a service that uses a portion of the line s full capacity It is sometimes pos sible to order a T1 that is divided into a chann
61. ces In the USA this is is the fast busy a dual frequency tone of 480Hz and 620 Hz at a cadence of 0 25s on 0 25s off Error Usually caused by an incorrectly entered number but can be from other problems in call setup Call answered Played whenever a call is put on air whether directly from handset or from hold Off Hold The sound that is played on air when a call is taken to air from Hold by pressing a line select button Line switch The sound that is heard when there is a call on air and a new call is taken to air by pressing a line select button This is the one Scott Shannon wanted Caller hang up A brief sound that is heard when an on air caller disconnects Caller Alert tone Sent to caller when call is answered or when caller is taken to air from hold to inform the caller that they are now on air The Caller Alert sound is sent to the caller not played on the air lts purpose is to let a waiting caller know that it is his turn to talk Since he has usually been listening on hold he will have heard the host say hello you re on the air a 28 Section 2 few times before it really is his turn Since this sound is played only to callers being switched to air it clears the confusion The Producer call screener needs to explain this to callers DTMF on air Choose the DTMF menu to set these tones Dialling keypresses send call setup messages digitally to VoIP lines or gateways Though they are not sent to
62. ch tool The reports tell you who called when they called where they called from and how long the call was The data can be exported to a csv formatted file that is imported into a spreadsheet or database applica tion for further processing Asterisk software Distributions Asterisk runs on Linux You can download Linux from many websites and then add Asterisk to the operating system Or you can load The OS and Asterisk from an all in one distribution disk You can also buy an Asterisk appliance pre loaded from companies like Fonality Trixbox and others If you have one of these Skip on to the next section titled Log in to the web GUI for the first time A distribution is an organized and maintained collection of open source programs these are available on an ISO image file that is burned to a CD and booted automatically formatting the hard drive and creating the entire system from bare metal Distributions can include the Linux OS the Asterisk components GUI tools and specialty modules that the creators of the distribution deem useful Choose one that has what you value If this is your first experience with Asterisk we suggest keeping it simple If you know a Linux guru his assistance may be helpful but not essential Linux has come a long way over the past few years and is now perhaps even easier and faster to install than Windows You ll also find that you can go years without a reboot due to its exce
63. cing much in the way of high frequencies why bother carrying them over the radio links Thus the radios were designed with narrow 4kHz carrier spacing When the first digital T Carrier systems where invented it must have seemed perfectly natural to stay with the 3kHz or so audio bandwidth enshrined in the microwave link technology A sampling rate of 8kHz with 8 bits compressed depth had a nice symmetry and delivered a satisfactory 4kHz Nyquist response limit so on with the show The Electret capacitor microphones ubiquitous in today s phones have frequency response well beyond 3kHz IP networking combined with modern codecs allow the better fidelity to be transported Some mobile phone applications such as Apple s facetime are beginning to take advantage of this The VX is ready for high fidelity phone calls and includes support for g 722 a codec rapidly gaining favor with VoIP providers and users g 722 uses about the same amount of bandwidth as g 711 but samples audio at 16kHz double that of g 711 It delivers very clean clear natural caller audio Future VX software releases are likely to includes support for more high fidelity codecs 62 Section 5 Specifications System Maximum number of phone lines 48 when used with aLaw or uLaw codecs for VoIP lines Higher quailty codecs such as G 722 consume more system resources and result in a de creased number of total lines available Maximum number of SIP nu
64. click Next TIP Be sure to remember your root Password You ll need it later to continue the system setup Centos installation commences this is the operating system The drive will be formatted and the installation and program files are copied The operating system will be installed and 80 Section 5 then Asterisk and the PBX in a Flash files Enjoy a sip of your favorite beverage as the process continues over the next few minutes Eventually you ll see a message congratulating you and that Installation is complete Not so fast The operating system is installed but there is the second part of the installation the Asterisk and PIAF Don t worry the system is running from a kickstart script that that will do all of the ugly stuff for you Keep that beverage handy Remove the CD ROM and click reboot The system will shut down and restart It will then offer you Please choose one of the fol lowing options Choose A recommended option This will begin a download of the latest version of PIAF At this point the system expects that you have it plugged into a network with a router that has DHCP and it will attempt to grab an IP address netmask gateway address and Domain Name Server DNS address If your Internet connection isn t working the install will appear to bomb at this point This is not a disaster But more than likely it will just work then you ll see it download the updates and continue to co
65. ct show 1 Show 1 33 34 Section 3 Set Talent or Producer Mode The VSet should be set to Talent mode when it is located in the studio and is used to put calls on the air Producer mode is for a VSet that is being used by a producer to screen calls In Producer mode calls cannot be put on air and are protected from being dropped The function of the right line button column is different in the two modes to serve the needs of each type of operator as described below Mode Talent Press the menu button to access the menu functions The LCD will show the various items that can be changed Select Talent or Producer and then exit the menu functions by pressing the menu button The Line Info Display Each line has a corresponding portion of the LCD display to the right of the line button that shows status and info about that line Line 20 ONAIR 101 E Next indicator The gt gt symbol Marks the line that will be taken when the next button is pressed Fader number The yellow orange rectangle near the status icon Line name The first text line This is defined in configuration using the VX Engine s control center web GUI Caller ID Caller ID will be displayed on incoming calls if it is available Outgoing calls will show the num ber dialed The green arrow to the left of the text points left for incoming or right for outgoing Calls that are blocked or that come from lines without caller ID
66. d be necessary 2 Get loaded Install Asterisk Put your freshly burned PBX In A Flash disc into the CD ROM drive and start the machine You may need to press the space bar when the machine asks if you would like to boot from the CD ROM If you have trouble check the boot menu in your BIOS Sometimes you have to set a BIOS menu option to allow CD ROM booting Most newer boards allow CD booting because Windows and Linux are distributed on CD ROM Have an Ethernet cable plugged into your new machine as the system will attempt to download the latest updates at the end of the installa tion You should have the machine connected to a router that has a DHCP server so that the machine can configure itself with the parameters that your router gives If you are using two or more Ethernet cards the new system will only attempt to use one of them for the automatic updates usually the onboard one or the card with the lowest MAC address lt may ask you to choose one It s no big deal If you guess wrong because you can change it later and get the updates or configure it all manually The procedure will start by copying all of the files needed for the basic operating system into a RAM disk that is used for the install You ll be asked a few questions along the way From the text menu select an install type Take the default by pressing the ENTER key This selects the Default installation It will remind you that the machine will be wiped completely
67. d the audio entertainment industry Only a year before a gentleman aptly named Bell had been the first boss to interrupt an employee with a telephone call birthing speech communication by wire Until the 1980s sound reproducers continued to work pretty much the way Edison s did and telephones the way Al Bell s did wiggly Grooves for the former and wiggly electrical currents for the latter Why did these technologies remain in stasis for over a century before eventually entering a wildly innovative phase only a couple of decades ago The answer is to be found in Moore s law which predicted a half century ago that silicon processing power would double every year Modified to 18 months it has been on course since and is expected to continue to for at least a few decades into the future Think of all the ways this has touched both your professional and work lives Digital audio workstations automation systems and mobile phones are all beneficiaries of this remark able progress The Internet too since processing chips are at the heart of network routers switches servers PCs and smartphones Notwithstanding The Great Moore s Law Compensator in desktop PCs which suggests that bloat in PC applications has taken much of what Moore s Law has given Fortunately we can avoid this in our embedded designs So why are we still mostly using telephone technology scarcely different from Bell s ancient prototypes A nodding acquaintance w
68. dress Login Password 34 If you don t see this or if the ID or Module information is incorrect you will need to Capture your current module configuration This is normally done at the factory but if you have added the phone module yourself you will need to execute the Capture so that the Element can see the newly added module NOTES RESOURCES ADDITIONAL INFORMATION 57 To Capture press and hold the Help key on Element s Navigation Module for 5 seconds until CAPTURE appears in the source name displays On the phone module you will see the assigned ID number in the upper left line icon position You can change this by pushing the adjacent line button End Capture Mode by pressing the Enter key on the Navigation Module Show profile VX9 Phone configuration Phone Module 1 Phone Url vx lap gt user pass G host 192 168 0 9 Show Name Host Studio Name Studio 1 Show Password F Reversed Hybrid 1st on right Mode Auto 12 Lines 24 Lines Enter your VX s IP address Use your Web browser to navigate to Element s Sources and Profiles gt Configuration gt Show Profiles web page Select the Element Show Profile you wish to configure and click on Phone In the Phone URL field enter the IP address of your VX In the screenshot above a VX IP address is shown as vx 192 168 0 9 where 192 168 0 9 is the IP of a VX Engine You can also specify a username and password if needed If you use the default
69. e things your way At Telos we faced this problem in the design of the VX system We needed a lot of things specific to the studio environment that are not supported in SIP s structures So we designed our own protocols for use within the boundaries of our system But we use SIP at the border of the system to connect with other vendors products and eventually to the Telco network This is just the strategy Cisco Microsoft and almost all PBXs vendors have followed And now this process is emerging as SIP s great value It s the glue that ties systems together Studio systems can talk with PBXs for the first time PBXs can talk with each other And eventu ally they will all be able to talk to Telco networks smoothly and fluently SIP s inventors got something right IAX as a SIP Alternative SIP is not the only game in town The Inter Asterisk eXchange IAX protocol is an alternative to SIP for interconnections between both VoIP servers and for client server communication IAX2 as the current version is named uses a single UDP data stream usually on port 4569 to communicate between endpoints both for signaling and data The voice traffic is transmitted in band in contrast to SIP which uses an out of band RTP stream for audio IAX2 supports multiplexing channels over a single link When trunking data from multiple calls are merged into a single set of packets meaning that one IP datagram can deliver control and audio for
70. e Ethernet RJ 45 Livewire is both an elegant technical solution and an unprecedented value Only one Ethernet RJ 45 connects dozens of bi directional audio channels and rich control Step by step There is a lot of information in the pages that follow Here s the very abridged version of what you need to do to get your VX system up and running 1 If you will not be using an existing network to support the VX install the Ethernet switch or switches depending upon your needs 2 Mount the Engine and set the IP address or accept the default Make the network connections 3 Install and configure your gateways if you will be using them to interface to POTS ISDN or T1 PSTN lines If you will be using VoIP trunking for your Telco connection get it connected and gather the IP and registration password info 4 Get a web browser going and do the system configuration 5 Move on to install and configure the VXset phones and VX Producer software applica tion Then to any other controllers you have in your system such as integrated console controller modules The Engine and Network The front panel OLED knob interface 10 Section 2 The Engine front panel display gives at a glance status information The main screen is pictured above It shows the IP number for the LAN Livewire port its connection speed and status The overall status and CPU temperature are also indicated The knob lets you scroll through the menu pag
71. e Internet Going beyond email academic researchers were imagining on line audio video whiteboard conferences where ideas could be shared live It became clear that the Mr Watson come here want to see you function had to be done more efficiently than by shouting across the college quad or sending invitation mails Thus was the IETF s working group MMUSIC Multiparty Multimedia Session Control born There had already been work within the telephone world that had resulted in an ITU standard but Internet types didn t much like it Too complicated they averred Too limited they sniffed Too Phone Company they huffed So off they went to do it the Internet way The document describing SIP was eventually published as proposed standard RFC 2543 in 1999 Work has been ongoing with the latest version of the specification as this is written being RFC 3261 NOTES RESOURCES ADDITIONALINFORMATION 69 The SIP message protocol is similar to the Web s HTTP Hypertext Transfer Protocol and shares some of its design principles It is human readable and request response structured SIP even shares many HTTP status codes including the familiar 404 not found Here is a typical SIP message INVITE sip skip there com SIP 2 0 Via SIP 2 0 UDP 4 3 2 1 5060 To Skip Pizzi lt sip skip there com gt From Steve Church lt sip stevec here com gt Call ID 4678995554545 4 3 2 1 CSeq 1 INVITE Contact lt sip stevec 4 3
72. e checkmark to put lines in ready hold mode and assign priority for the Next function See Next Button and function Ready hold works like normal hold except the line status icon has the checkmark to indicate to operators that a call has been determined to be ready for air normally by a producer screener The Vset must be in Producer mode in order to use this key Tip You can re order the next priority queue at any time Put lines in Ready Hold even those already on Ready Hold in reverse order from the order you wish to air them First Screen hold the call to be aired last then the call to be aired second to last and so on You would screen hold the call that you want to be aired next last At first blush this may seem confus ing but it makes sense when you remember that the call held the longest goes first to air Hold Button e Holds a call that is ringing on the handset or on air If more than one line is in this category there will be a small hold icon near the status icon on the lines that could be held Press the line that you want to hold OPERATION 37 Drop button Drops a call that is active on the handset or on air If more than one line is in this category there will be a small drop icon near the status icon on the lines that could be held Press the line that you want to drop Lock Normally taking a call to air causes any others on air to be dropped If you need to conference two or more call
73. e for a reasonable default usually one hour The other two fields allow to set the password and change the user name used for authentica tion You will almost always want to set the password for lines that get registered with the server but it is also possible to use it with trunks there are some providers that require it for outbound calls even on SIP trunks Auth User if unspecified defaults to the extension It is only needed to be filled in if required by the SIP provider For quick access to the configuration you can use the Config link in the Show configuration page Each position with correct configuration has a link marked next to it Clicking on it will bring you to the corresponding server configuration page with the matching line configuration highlighted In case the line is not configured the extension field for adding it will be filled in from the show SIP and Network Address Translation NAT As SIP messages list IP addresses and ports used to transmit audio RTP it doesn t work well if the client is in a private LAN but needs to communicate with a SIP provider outside of the LAN As messages pass through the router it translates addresses in IP headers but not the SIP message itself giving the provider wrong connection info Many SIP providers use clever hacks to work around this limitation without any additional sup port from the client If you are connecting VX to a SIP provider that doesn t have such se
74. e pages and how to use them is detailed below 4 gt amp Q BD httpyns2168 07 y O soog TelosVX Control Center Telos Y TelosVX Control Center Configuration Main de Configuration Studios Shows Call Audio Processing Main Tones Logging Network configuration NTP configuration Web and LWCP passwords Information Studios Calls Devices sip Stream Statistics Script Manager Global SIP gateway configuration Script Information Misc lOManager Info f In Studios Status Logs Studio configuration Livewire channels Show selection Backup Restore Firmware Shows Show configuration Line configuration Fader assignment SIP gateway per line assignment Call Audio Processing 14 Section 2 Main page General Configuration Moers Minutes OST B on Time Zone Offset 9 0 o Apply Tame Zone Cont NTP Config Engine Name Assigns the text name for this VX Engine It becomes the Linux network name for the Engine If you will not use the defaults for IP Netmask and Gateway enter them here IP addresses can be entered for both the LAN and WAN ports Web Username and Password Used for web access for configuration and monitoring User name is always user and the default password is lt blank gt LWCP Password LWCP stands for Livewire Control Protocol the way controllers communicate with the VX Engine A password c
75. e we use most often is the ISDN PRI T1 E1 Tormenta 2 or 3 type card available from a number of vendors ISDN PRI and T1 or E1 use the same card since ISDN employs T1 or El as the underlying transport A four port PCI card is available from Phoniceq inc www phoniceq com for about 350 Quality and service from the company have been excellent They also offer a number of other cards including for ISDN BRI The company offers examples of use and driver packages that simplify installation Asterisk creators Digium also offer a wide range of card types and support options www digium com en products digital We don t recommend installing analog interface cards within an Asterisk server While there s nothing wrong with the cards we think it s better to use external gateways to interface POTS lines and Telephones such as the small appliance gateways from Cisco Sipura Grandstream AudioCodes Patton and others that go straight to SIP Ethernet Another way to go for more than 8 lines is to repurpose a Channel Bank with a T1 interface and connect it to A T1 card in the Asterisk Channel banks are available at reasonable cost on ebay and from used equipment suppliers It should be equipped with an FXO Foreign Exchange Office interface if you plan to terminate analog lines from a Telco Channel banks use different interfaces so be sure that you get the right hardware configuration Better is to avoid analog lines altogether
76. each AEC Microphone input from the studio mic Microphone output feeds the console mic input that goes to the phone system Reference input parallels the signal that feeds phone audio to the studio loudspeaker If multiple mics are used there must be an AEC dedicated to each of them The AEC cancels audio that is put on the reference input so that it does not appear on the output Without canceling the output would be a mix of the studio mic and the delayed echoed phone audio After cancellation only the mic audio will be present That is it creates a replica of the acoustic path then passes the reference audio through it and subtracts the resulting filtered signal from the microphone feed T received signal from phone echo A Reference in y ici echo free mic Y O at Be AEC Gut gt to phone local speech The AECs are full range and low distortion so they can be put in an air signal path if need be Nevertheless we recommend that they be inserted only into the mic to phone path if possible avoiding causing any trouble to mic to air fidelity There could be subtle frequency response alteration depending upon the echo path and other factors Noise might also be added NOTES RESOURCES ADDITIONALINFORMATION 51 In a studio equipped with an Axia console the needed signal connections are available via Livewire Configure both the console and the VX Engine with LW channel assignments so that the correct
77. ectable and fixed lines are supported For the former The Element s Call Control Module includes line select buttons and NOTES RESOURCES ADDITIONAL INFORMATION 55 status icons For the latter Phone Fader modules have a couple of extra buttons to manage the Telco lines Livewire makes it easy to connect the multiple inputs and outputs that the VX s multi hybrid capability affords A gt PSTN CU aoe Element Console T1 ISDN or POTS Gateway Telos VX Engine j Audio 1 0 GPIO ME Axia Powerstation ws a Telos VX Assistant or 3rd party applications Installation The setup in the block diagram shows the Axia Powerstation as the core for the studio provid ing the Ethernet switch function The VX Engine LAN port attaches to one of the gigabit ports and the VSets to PoE powered ports The Engine WAN port could have its own switch to connect the VoIP lines and gateways Or the VX can be configured so that the LAN port could serve everything with no additional switch The Powerstation could also link up with a central switch to pass audio to from other locations in the facility In the Element Install or confirm the presence of the call control module and the faders you will be using for telephone operations Some possible Element configurations are 4 1 Call Control module for 12 or 24 lines 2 hybrids faders 1 Call Control module 1 Phone fader module for 12 or 24 lines 6
78. ection for a list of SIP VolP providers that we ve tested with What s a Line While we are talking about Telco lines and VolP services we should take a moment to consider just what is a line these days Use of the word line is becoming bothersome at this point in telecom history Back when an analog line was associated with a single telephone number the word had a clear meaning When you ordered ten lines you received ten physical wire pairs and had ten telephone numbers Yes roll over rotary service exposed only one number to the public but the others were still there ISDN was the first step on the path to trouble We engineers began to speak of a BRI line with two voice channels But producers and hosts still communicated with each other as if channels were lines saying things like The tree hugger is on line 3 Do you want to take him now We knew that meant BRI line 2 channel 1 or T1 channel 8 but users had no clue ISDN PRI made matters yet more knotty since the telephone number and physical channel were divorced Now we are faced with SIP trunking and other IP based services where a single link which might be connected via one two or more physical copper pairs or an optical cable or via a wireless system might carry any number of voice channels And what is going to happen when phone numbers become obsolete as they already are for VoIP applications such as Skype Users blis
79. ectionable 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 appa reil numerique n emet 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 TELOSVX MANUAL iii Table of Contents W support YOU ce e o ed ee aa a a he A aE do c de i 1 Welcome 1 Note from Steves ee ee ee 1 What s the Bigidea oso s o ns 2 ThE VENDIDO ser e e eed eo ae a Ew Hee cd 3 The VSet Phone Controller o o o o o o o o o o ee 4 VX Producer Windows Software Application 4 Console Controllers o o o o o o o o o ee 5 3rd Party Producer Applications o o o o o o ooo 5 The Acoustic Echo Canceller o o o o o o o oo ooo 5 2 Installation and Configuration 7 Connecting to PSTN Lines Gateways amp PBXs 7 Using VoIP to Connect to the Telco Network 8 Whats ailing Saad o AA 8 Livewireforaudiol 0 o o 9 Step by Step aio ac a e a a we
80. elized portion and a data transparent part for IP connectivity ISDN PRI ISDN PRI Integrated Services Digital Network Primary Rate Interface uses the same underly ing circuits as T1s and E1s However it employs sophisticated out of band signaling which allows transfer of information such as calling number codec type clearing causes and such It s rather strange that non ISDN T1 sends caller ID modem encoded in the audio channel no The speech paths are called B bearer channels while the signaling is carried in D data channels Almost all large VoIP gateways and PBXs support ISDN PRI lines The signaling in the USA is a slightly different protocol than that used in Europe and other parts of the world Your gateway will need to be set to match the protocol on your line Normally in the USA this would be NI 1 National ISDN 1 while Europe would use the Euro ISDN standard You might hear the term OSIG in the context of ISDN PRI gateways This is a signaling protocol that is yet more sophisticated than ISDN s usual 0 931 protocol and is layered on top of it With the ascendancy of SIP OSIG looks to be yet another valiant attempt falling by the wayside 54 Section5 ISDN BRI ISDN BRI Basic Rate Interface lines offer two B channels supported by one D channel These were intended as a residential replacement for POTS lines or for small businesses One application envisaged by its inventors was to allow a simultaneous voice
81. equest not in dialog send 481 resp 6e787b90 Mar 8 07 33 21 492 WARNING Unexpected event in cpsip_tact_receive_msg cpsip_ Shows 6e787b90 Mar 8 07 22 21 493 DEBUG gt gt gt NET_SEND Hybrid 6e787b90 Mar Modus IPv4 Tones 6e787b90 Mar MsgDelete msg 6e787b90 Mar 0x87e8520 Logging 6e787b90 Mar Sending message to 81 94 239 150 5101 from 195 14 F 6e787b90 Mar SIP 2 0 481 Call does not exist 6e787b90 Mar Via SIP 2 0 UDP 127 0 0 1 5101 branch 9hG4bK 121 From 55 lt sip 550195 13 179 146 gt tag 25250132373 To SS lt sip 550195 13 179 146 gt Call ID 2743542086 CSeq 1 REGISTER g E A E 8 8 8 8 8 3 8 Studios 6e787b90 Mar 8 Calls 6e787b90 Mar 8 6e787b90 Mar 8 pz 6e787b90 Mar 8 Stream Statistics 6e787b90 Mar 8 Script Manager 6e787b90 Mar 8 A 6e787b90 Mar 8 Script Information 6e787b90 Mar 8 Misc 6e787b90 Mar 8 6e787b90 Mar 8 Recv rom OK IPy4 2sssssssssacccccsccs JOManager Info 6e787b90 Mar 8 31 515 MagCrezte msg System 6e787b90 Mar 8 07 33 31 515 DEBUG 0x87e61 8 8 8 8 8 8 8 8 8 8 8 8 8 8 a 8 8 Sendto OK 24ssssssscsscscccacacce MsgDelete msg z Ox 72618 Status 6e787b90 Mar 731 515 DEBUG Received message from 81 94 239 150 5101 on 195 1 6e787b90 Mar 31 515 DEBUG ACK sip 11 195 13 179 146 SIP 2 0 6e787b90 Mar Backup Restore 6e787b90 Mar 6e787b90 Mar 6e787b90 Mar 6e787b90 Mar 31 515
82. er A stereo file will be passed to the editor when both a mic and phone audio are recorded A mono file will be passed when phone only record mode has been selected Files The recorded audio files are physically stored on the local PC or on another PC or server accessible over the network The location depends upon the file path configuration in the VX Producer settings menu Ready Studio Producer Import Export Name Length Comment Show Seventeen way 1 45 Recorded incoming call 1 m aar By default files created in the studio will go the list under the Studio tab and files created by the producer will be saved under the Producer tab The Ready tab holds files that are ready for air The Import and Export tabbed lists are used to accept and send files to other people on the network Again the tab names and file locations depend upon your settings 48 Section 4 The files are in standard wav format that may be accessed by a variety of audio applications Standard Windows file sharing allows access to the files across the network at the various machines Host Producer Text Chatting The window at the lower right lets people within a Studio communicate with each other Type text into the lower field and it will appear on any other VX Producer applications selected to your studio Producer Caller on line 5 want s to talk about taxes Studio Put him on hold Producer Ok Type your message here a The Studio and
83. er might ask Why was such a low fidelity considered to be satisfactory for digital telephony Microphones loudspeakers and earphones have all had much better fidelity for many decades Indeed when the g 711 codec was standardized in the 1960s FM radio was just getting going with its much superior 15kHz bandwidth Mostly the coding choice resulted from the legacy of the carbon button microphones that were ubiquitous in telephones throughout most of the early history of telephony Edison invented the carbon microphone and licensed it to the Bell System which had only an impractical liquid based microphone in its own portfolio This microphone works on the principle that acoustic pressure squeezes the carbon granules more closely together thus lowering electrical resistance in proportion to the pressure As you can imagine it has a non flat response curve with a 5db peak at 2kHz and hefty roll offs below 300Hz and above 3kHz They were mostly abandoned for all but telephone use in the late 1920s but were standard in phones up to the 1980s Old analog long distance lines also had a lot of high frequency attenuation owing to capacitive effects When microwave radios were introduced to long distance telephony a decision had to be made as to what frequency range to accommodate in their FDM Frequency Division Multiplex scheme There was as always a trade off more frequency response meant fewer channels Since the micro phones weren t produ
84. es Among the menus is one that lets you change the IP address and another that shows the status of the Livewire audio channels The main screen shown at power up is pictured above To see any of the other pages menus push and then rotate the knob to show each in turn Pushing the knob enters you into the menu that is displayed Editing items inside menus such as the IP address is similar navigate to the specific item you want to change by rotating the knob then push it to choose the item such as the IP address In the case of the IP address select from among the digits by rotating the knob then pushing Finally rotate and push to enter the value you want yeah this isn t the most convenient of user interfaces but you shouldn t have to use it the web interface once available is much more friendly The reason for the IP address entry here on the front panel is to avoid the circularity that you need the IP address to connect a web browser in order to enter the IP address should the default not work for some reason We think you ll also find it convenient at times to have the IP address displayed on the front panel Network Connections The VX Engine has two RJ 45 Ethernet jacks on the rear panel Both are 100 1000 auto speed setting The upper LAN port is for Livewire audio and by default everything else such as PCs used for web based configuration VSet phone controllers producer PCs and console control lers This can
85. ess power consumed and less waste heat As a general rule buy a larger power supply than you think you ll need If you are the extra careful sort consider building two identical boxes or at least keep a spare identical motherboard and set of peripherals such as Telco interface or Ethernet cards on hand The convenient availability of a given motherboard and some kinds of PC hardware will usually be less than a year at the current pace of development NOTES RESOURCES ADDITIONALINFORMATION 77 Keep any filters clear and inspect the machine periodically to see if dust bunnies are col lecting anywhere Do a full backup using the Linux DD full disk image bit copy or clone command once in a while quarterly annually or after you are satisfied that any major change you ve made should be retained after a trial period Consider that the hard drives might hold users voicemail asterisk configuration and call records There is a good backup utility that will automatically make regular backups of the unique data and whatever you specify Think of this as an incremental backup There will probably be spare capacity so you could use your Asterisk server to provide other services you need such as centralized logging using Syslog or a Network Time Protocol NTP server Expansion Cards Because Asterisk has become so popular there are a variety of expansion cards available that can be used with it The on
86. ess that you are assigning to the phone and press the green Go button 2 Do the same for the netmask value Most often this may be kept at the default 255 255 255 0 3 Do not enter anything in the Vset Gateway field unless directed to do so by Telos Support The Vset will operate on the Livewire network and will not need to be routed anywhere unless you have an unusual or custom installation After selecting the engine you might see Connected to for a moment at the bottom of the set screen 12 Section 2 System Configuration Selectable and Fixed Lines The VX supports the concept of both Selectable and Fixed lines It is possible to have a mix of both Selectable lines are used for an operation style like traditional Telos systems where there was a line selector before the hybrid or hybrids Hybrids were expensive which meant that we were not able to have one dedicated to each line This limitation is now over and we can pro vide a hybrid and other processing functions for each line Yeeeaaa Progress with a capital P Nevertheless operators are used to the operating paradigm that the old 1A2 key phones made so widely known where you have a column of buttons and pressing one takes a line and drops the one that was active before For that reason we have kept this operation style as the default for the VX We have also kept the lock function that lets an operator conference
87. fix it and it was decided to keep it that way Nevertheless makers of some SIP devices decided fix this mistake on their own breaking compatibility with most other devices that stick to the standard including VX As the result VX will use G 711 to communicate with such devices Because it is essentially a pro grade SIP phone that includes g 722 the Telos Z IP One also works as the other end of a VX g 722 call Note that VX s call audio processing is bypassed when using g 722 18 Section 2 RFC 2671 MD5 digest authentication is supported RFC 2833 DTMFs is supported Supports SRV address lookups Has basic NAT support Unlike some other SIP devices the VX supports non numeric extensions Server Configuration TelosVX Control Center Configuration Main Asterisk SIP Patton PRI General Settings Cisco SIP Server 192 168 0 155 Studios Name Asterisk Shows Call Audio Processing External IP Tones Logging Apply Information Studios Lines Extension Register Expires Auth User Auth Password Shows Studios Devices Stream Statistics 3101 a DE peee Fixed Delete Script Manager Script Information Misc ARAN lOManager Info System Add New Te Extension Register Expires Auth User Auth Password Backup Restore o Firmware Add When the line details are entered in Show configuration VX creates the server configuration a
88. fixed hybrids in the Primary Source field In the VX Engine studio configuration you configure both fixed and selectable hybrids enter a Fixed Hybrid number in this field if you wish to associate this source with a Fixed Hybrid The other fields in the Source Profile page are non phone specific and can he configured as you choose See Chapter Two Configuring Inputs in the Element manual for info on these options Repeat this procedure to create a source profile for each hybrid in your studio You ll now need to load the hybrid sources onto the fader channels This can be done either manually or automatically using Show Profiles We recommend that you make this assignment in each Show Profile which will use the phone system since leaving manual assignment to air talent is asking for trouble For normal operation you would assign Hybrid 1 to the left fader channel of the Call Controller and Hybrid 2 to the right fader channel Additional hybrids if you have them may be assigned to any channel on a 4 Fader Phone Module Note Within Element each phone type audio source has both an incoming from caller and associated backfeed to caller for automatic mix minus generation There are several configuration options that affect which audio sources are fed to the hybrid under various conditions Refer to the Source Profile Options and Feed to Source Mode sections in Chapter 2 of the Element manual When you are done entering config
89. h no compression Low delay and synchronized channels are other distinguishing characteris tics Another way the two differ is that AolP uses an advertising discovery protocol for receiv ers to find sources instead of the Session Initiation Protocol SIP that VoIP employs AolP uses a system wide clock mechanism to support low delay and tightly synchronized channels Finally AolP often takes advantage of IP s multicast capability to permit multiple receivers to listen to an audio source efficiently AolP is intended for managed guaranteed bandwidth networks such as with an Ethernet switch as the core of a local area network And what about IP codecs Now you are very close to VoIP These use SIP for call setup and various codecs for compres sion so are similar to VoIP telephones In fact they actually are VoIP telephones and can sometimes interoperate with them They have better codecs than VoIP phones though AAC ELD in particular Advanced IP codecs such as our Z IP employ sophisticated technologies to overcome the Internet s deficiencies Dynamic buffering error concealment and more clever stuff have read that some VoIP PBXs use IAX trunking Can I use the VX with these IAX is a protocol invented by the Asterisk people It provides functions similar to SIP but with more bandwidth efficiency The VX doesn t support IAX trunking at this time But you can connect the VX to an Asterisk with SIP trunking or as multiple SIP e
90. h no noticeable feedback or return echo If you ve already experienced the Axia intercom system you know what we re talking about Installation and Configuration Audio I O GPIO AA see e LAN port WAN port Ms gt gil Gateway Kiai Al vv s 4 T1 ISDN el yo or POTS pa ES a J H pm il Call Screening S 3 VoIP Software Application PSTN a eee The diagram shows a generic installation that could be used in a typical studio Two Ethernet switches are employed to keep the isolation between the WAN and LAN ports on the VX Engine The LAN port carries the Livewire audio and everything else but the VoIP which is on the WAN port Telco lines enter two ways From the PSTN via a gateway and from a SIP VoIP provider over an IP network VX Interfaces provide analog AES and or GPIO connections to traditional studio equipment They could be thought of as another kind of gateway And just as with the Telco gateway they are not needed when the studio equipment can accept AolP natively For an example of this sort of installation see Axia Element Console as VX Controller in Section 5 Connecting to PSTN Lines Gateways amp PBXs PSTN is an acronym for Public Switched Telephone Network the traditional telephone network that includes POTS ISDN and T1 as last mile connection technologies The VX system connects to Telco lines using industry standard SIP Session Initiation Pro
91. have made echo the trouble du jour The canceller needs two inputs and produces one output The Mic input is fed from the studio microphone The reference input is the audio that needs to be cancelled the received phone audio that is going to the monitor or preview loudspeaker The output of the canceller goes to the VX phone feed input s Why didn t we just internally connect the canceller Because the canceller works best when the reference input is after anything that is in the phone to speaker path such as the volume control and mute In a Livewire equipped studio it should not be too difficult to tap the needed signals The canceller is low distortion and full fidelity so it may be used with wideband codecs More information on the Acoustic Echo Canceller is in Manual Section 5 GPIO GPIO General Purpose Parallel Input Output control is possible The Livewire channel assign ment for this is done on this page The electrical connection is made via Livewire GPIO Nodes using the standard 5 inputs and 5 outputs per channel You tell the system how many Livewire channels you will use with GPIO Channel Count After you press apply you will see entry fields for each of the channels Specify the Livewire channel number you want to use for each and choose the functions you want to assign GPIO Actions are inputs to the VX from some external source which could be nothing more than a pushbutton GPIO Indications are outputs from the VX For examp
92. he VX sets is also handled Any VX set assigned to a studio and show shared with an Element will mute its ringer when the Element tells it to Nice eh Selectable lines use the Element s Call Control Module as the line selector lt can be used in either 12 or 24 line mode depending upon configuration In the former two columns faders are used in the latter one column fader The same module is used for either 12 or 24 line 40 Section 3 operation When 12 line mode the left column of Status Symbols displays the line number and the right displays status When configured in 24 line mode each active line is numbered with the number being shown when the line is idle replacing the idle dot In the 12 line mode the left row of buttons assigns lines to the left fader and the right row to the right fader In the 24 line mode both rows assign lines to the left fader To assign to the right fader push on off or preview on the right fader which activates flashing and displays PRESS LINE on the alpha display Selecting a line assigns it to the right fader To cancel the mode before selecting a line press the right drop button or the right channel off button When you need to be able to assign selectable lines to more than one or two faders the Call Control Module can be expanded using one or more Phone Fader modules The additional Phone fader channels will work like the right fader on the Call Control Module You can quickly build multip
93. he edge of the Kent State University campus The two largest studios are equipped with Axia Element consoles One of these is the main on air control room while the other is used for producing a variety of long form programs including an evening interview and call in program The VX currently serves these two studios The main station call in numbers are delivered via a SIP service offered by the VoIP provider bandwidth com IP network service is from the local Telco and is delivered over a T1 circuit with the University s Internet connection being a backup A router located on the WKSU premises is configured for automatic backup 20 telephone numbers are delivered over the SIP service which may be used for both incoming and outgoing calls Despite the leading edge nature of SIP Telco service CE Chuck Poulton and network administrator Dan Kuzinsky report that it has proven reliable and has delivered excellent audio quality perhaps because call audio is maintained 4 wire independent send and receive audio paths and digital from the PSTN to the studio The PSTN g 711 coding is maintained until it is converted to the high fidelity Livewire format in the VX system Two analog lines from the local Telco connect to the studio IP system via a Grandstream 4 line FXO gateway Another two lines are brought into the gateway from the University s NEC PBX An FXO interface card could have been installed into the Asterisk but a dedicated external
94. hould be set to no It remains to be seen if and how well Asterisk will support wideband codecs Until this is clarified and when the VX supports wideband codecs it might be required to keep Asterisk out of a wideband call path Let s get started Log in to the web GUI Point your browser to your Asterisk server address being sure that your computer is on the same subnet For example if your Asterisk box is on 192 168 0 248 with a netmask of 255 255 255 0 and your browser computer is on 192 168 0 11 or any IP with the same first three bytes 192 168 0 and the last byte between 2 and 254 accompanied by the same netmask as the Asterisk box s you are good Supply the login credentials and you should see the screen below NOTES RESOURCES ADDITIONAL INFORMATION 83 Click on the Apelications Maces System o ES iel E Muar 7 B12 Oje De Gat Yew Wgory Beokmaks pois teip 00 reno gt gt avo alte In Mont Valtes Getting Started latest Headines EBSashect CaM Login New weret grupn E flearote 010s Osumi ANY cise pikem Pv p n AR Omo Paroles Ste Tok 070 W Mongan 19 43206 ONN 24 1230 Goolrey C wr nto KEY i a x I v s mensi fie o Doe file bouner VeMeraa TO bo 20502 un 35H Erne Man L ie admin virtual slide switch at the bottom left of the page and the menu will change to the one below You may be asked for your username and password to access thi
95. hysically Having said that some Telcos and other companies offer bundled IP network and SIP service Generally these are more expensive than the unbundled services with the pitch that having a single vendor delivers higher reliability and one stop troubleshooting We have seen this approach more often outside the USA It remains to be seen if the promise is fulfilled We ll keep our eye on it BYOD Bring Your Own device Providers are recommended for broadcast application Com panies such as Vonage that supply an ATA Analog Terminal Adapter for each line do not provide SIP credentials for standards based equipment such as the VX and are not oriented to serve customers with sophisticated requirements such as broadcast stations They tend to be more expensive as well Broadcast users are better served by wholesale providers who deal with Telephony and IT professionals VoicePulse is a smaller company that excels at providing tech support They have proven themselves excellent at solving obscure problems that many providers will just ignore They know Asterisk and can be counted on to give good advice They don t have phone numbers everywhere but they probably won t disappoint you East and West Coast gateways are available Local Number portability LNP is available Bandwidth com is used by at least one station running a VX with solid performance being reported Vitelity offers reliable and very cost effective DID numbers
96. inked to this port The VSet is powered over the Ethernet connection This is 48 volts DC according the PoE Power over Ethernet standard The Vset s power consumption is 15 4W Power may be supplied by the standard PoE injector we include with the Vset or it can come from an PoE equipped Ethernet switch If a PoE switch is used ensure that the Ethernet cabling is not so long that it causes too much voltage drop Some Axia products such as the PowerStation and IQ console have ports that provide PoE Connect the Ethernet cable and confirm that that the LEDs near the jack are indicating correctly The green LED link should be on solidly showing a good connection The yellow LED data should be flashing showing data flow 4 After a few seconds the LCD screens should come to life as the Vset boots up If they do not the likely problem is the PoE supply or an ethernet cable issue Press the Menu button and then navigate to the Setup Menu On the Vset 1 6 press the more button to display additional options including Setup Press and hold the button to the left of Setup Menu key for 3 5 seconds and then release it to access the setup menu for changes This press and hold feature will make it less likely for unauthorized users to make changes that could disrupt normal operation Just pressing it and releasing it without holding it will only show you the IP address and will not let you change it 1 Enter the IP addr
97. ions with a simple mixer off to the side You can set up a hybrid for each of your workstations or production rooms You can use the VX Producer software or VXset for the user interface Or you could program the VX to automatically answer a call transferred to it Using the VX to replace Couplers If your station uses Pre delay IFB Interruptible FoldBack Couplers or listen lines the VX can replace an entire wall of hardware and by using DID Direct Inward Dial numbers instead of POTS lines you can also save a lot of money each month by replacing those expensive analog lines Create dummy studios for each coupler bank and shows with the phone number or extension you want to use and tick the auto answer box next to it Choose the audio feed from among the Livewire sources The VX lends itself to creative solutions As you become more comfortable with the system we re sure that you ll find more of them Please share them with colleagues at our website www telos systems com forums and check in to see what s new INSTALLATION AND CONFIGURATION 25 Call Audio Processing page Audio Processing and metering Telos Y TelosVX Control Center Hybrid Settings Configuration Main SIP Devices Stream Statistics ore nateve Studios Enable Call Audio Processing ona Y Shows Receive AGC al Y Tones Noise Gate Ala Y La Receive EQ Mode laaie ooo Information Studios Additional Low EQ rra Y calls Aastra High EQ lan
98. irst day star morning DJ Scott Shannon called Frank Foti the Chief Engineer to complain he was missing his kerchunk kerchunk He liked the clunk saying it created a kind of dramatic yer outta here punctuation when he dismissed a call and took another He would have loved this feature of the VX We ve included a key phone button punch sound in his memory For line status sometimes called call progress tones The Tones configuration page lets you choose either USA or Euro standard for the default tones You can upload your own sounds Files to be uploaded must be mono in the signed au file format linear PCM 8 16 24 or 32bits or 32 bit normalized floating point Ringtones MUST be 8kHz 16 bit mono other tones 48kHz You can use the free Audacity audio editor to convert whatever you have to this format Use the Export command and choose the required values The files you wish to upload must be available on the PC that hosts the Web browser you are using to access the VX Your tones can be any length but should normally be kept short to conserve memory and to avoid annoying listeners Dial tone This is only heard when a line button is pressed before a call is dialed Ringback tone Heard when dialing is completed and the called phone is ringing Busy Heard when the called phone is busy Reorder Signals that there are no call paths available in the PSTN usually caused by conges tion in links between central offi
99. ited e Getting Started latest Headines BSachdet Feather E 4u Server Short De PEER Deta a Aintaade asto host sutbourd y tel ss met nateyes extersapeYoncestermalcP contect tros internal mare senal Incoming Settings USER Contert trorr intemal USER Dotals type fraend stafeage aut0 howtesnbuned watelity ret For incoming calls this is done with an inbound route that matches digits sent be the provider the Direct Inward Dial number itself and telling it where to go in our case to an extension 88 Section5 Ble Edt Wew History Bookmarks Pois Help e C 2 ietpslow gorroa com W80mern php id adrrin e Me o a Pz Mosx visited GH Oetung Santed Later Headines Bsiestdot CalMad Login New uenet group n f Flowrowe 0905 Dune Aly 198 ile P e El Shire info Favorites J Stumbiers Tools O Sasna Piccogie lt Googie Geogle Ne FAHowa big E 070 W Mow 4220 Wer 18 1230 Geofr T WINIFRED Koogie vo Freer g F y FreePBX PRZ Pi as 0 MIA Route Talk Line FreePBX System Status Mocule arar Delete Route Talk Line Ade Incoming Route une Tool View All DIDs SIPSTATION Lat incoming Route View User 010s Extensions S View General DIDs Feaxre Codes reat ok Use View Unused DIDS A DID Number 7005130255 Trews Caller ID Number any DID any CID Sezer CID Priority Route any DID 1772 7608411118 any Inbound Routes Opens co
100. ith history shows this often to be the case A technology is good enough and good enough suffices Traditions grow roots and the incremental improvement offered by an innovation is not sufficiently enticing to displace people from their comfort zones But eventually the capability of a new technology or constellation of technologies reaches beyond incremental enabling a fundamental re thinking of possibilities Then POW that s when the world changes That is just what is happening now Robert Lucky of Bell Labs observed a couple of decades ago that this ever increasing power of digital processing was going to affect communica tion more than computation and that there was a much stronger driver for change in the former The Internet and smartphones have proven him resoundingly correct Such is the situation exactly now with regard to on air phones We now have the tools to achieve what was only imagination until recently As you might know I ve been doing this stuff for a long time so it is particularly satisfying to see it all coming together With the VX we ve wedded the capability of modern networking to the remarkable power of today s digital processing to bring the benefits of the result ing synergy to broadcast facilities As you discover what the VX can do trust you will appreciate how it can enhance the appeal of your listener interaction segments It does everything older systems did smoother ea
101. l location Allows display of a routing map on the codec LCD display Upon request keeps a record of network performance in order to assist in troubleshoot ing problems caused by QoS impairments Many products that support SIP for its standards based interconnection capability do not have an internal architecture corresponding to the SIP specifications so you would not see these SIP NOTES RESOURCES ADDITIONALINFORMATION 71 server components listed therein or they might be included but not labeled according to the standard names bulleted above Instead these functions would just be provided as part of the system black box Cisco and Microsoft VoIP products fall into this category for example Addressing SIP addresses also called SIP URIs Uniform Resource Identifiers are in the form sip user host The user portion of the address can be a text name or a telephone number and the host portion can be a domain name or an IP address The address resolution process normally begins with a URI and ends with a username at an IP address Just as with email the sender needs no information about the physical location or IP address of the receiver This is one of the powerful features of SIP it automatically implements portability and mobility Examples of valid SIP addresses The usual form is an email address prefixed by sip sip joesmith company com You can call a PBX telephone at a business this way sip 123ftel
102. l page status screen that shows version numbers services running and an OK or other summary Then at the bottom of the screen you ll see root pbx _ NOTES RESOURCES ADDITIONAL INFORMATION 81 Type help pbx and lt enter gt you ll get a list of commands First Let s set up your ethernet IP address and get this box on your network so that you can get it going fully At the prompt type netconfig and lt enter gt You ll get a blue screen not like in windows don t worry asking if you d like to set up networking Press the space bar if the YES button is highlighted the lt TAB gt is used to toggle between yes and no The next screen has a place to enter an IP address netmask default gateway and name server or a check box that you can select with a spacebar tap that enables DHCP so that your router or DHCP server can assign an address for you We suggest that you assign a static address as this machine is a server that will always be online and running Below is our configuration change this for your network On our lab network the router is on 192 168 0 1 We use a class C netmask of 255 255 255 0 and the DNS server at 8 8 8 8 works almost everywhere and is fast so it s used often Use the lt TAB gt key to advance to the OK button and lt enter gt to select it At the prompt root pbx _ Type ifup eth0 and hit return There will be a short pause and then you ll get the prompt back
103. l ways The diagram shows a typical installation hinting at the possibilities inherent in the system ANS PSTN 5 T1 ISDN or POTS Gateway or PBX Telos VX Engine VoIP from telco via SIP trunking Livewire Network Ethernet Switch Telos IP Assistant Producer or 3rd party applications The VX Engine A 2U rack mount device with enormous processing power the VX Engine provides all the call control and audio processing needed for the system It supports dozens of telephone lines and many studios Its two gigabit Ethernet ports provides a cost effective interface to both telephone lines and studio audio via Livewire AolP 4 Section 1 Call processing is sophisticated and flexible Lines may be readily shared among studios A web interface allows easy assignment of lines to shows which can then be selected by users on the studio controllers Each studio can provide its own Program on Hold Audio processing features also have taken a leap forward There is a hybrid per line allowing multiple calls to be conferenced and aired simultaneously with excel lent quality Each line is also equipped with AGC automatic EO processing and override ducking All connections to the Engine are via two Ethernet jacks that are extended with an Ethernet switch to support a wide variety of peripherals telephone lines Livewire studio audio VSet phones VX Producer PC applications console integrated controlle
104. lar G 722 AMR WB is part of the new ITU standard for mobiles so should grow over time Meanwhile some IP based apps for mobiles are starting to use wide band codecs such as MPEG ELD in Apple s facetime app Note that VX s call audio processing is bypassed when using g 722 OK am starting to see the light Cool stuff but where s the catch Is VX hard to install and configure Setup is via web It may be little different than what you re used to or not but it s not difficult and some customers never crack open the book to set it up Power and flexibility do come with a little complexity but we ll always be at your side should you need us There is no such thing as a free lunch it must be hard to use then know there s a catch really don t have time to explain a new system to the air staff We know Rest assured it s easier for your talent not harder We recognize that any time you change anything in a studio there can be some transition time While there are a lot of new features in the VX your staff can use the basic stuff immediately because it works just like familiar and comfortable Telos gear The color hi rez LCDs and seamless console integration to Axia Element and iQ enhance the user experience As you read this systems around the world are screening calls and putting them on the air without drama Jocks and Talk hosts alike praise the VX If use an Axia console it gets even better Yes that
105. le there s a Ringing Line could be used for a wall mounted lamp and Delay dump was pressed to trigger a profanity delay unit You can find the details regarding the corresponding pins of the GPIO connectors in the various GPIO Node manuals INSTALLATION AND CONFIGURATION 23 Overall Show Configuration Telos Y TelosVX Control Center Configuration t Main Show Configuration SIP Studios Show Name Lines parte Joe Joe 6 New Studio 1 Delete ope 1 coupler1 1 coupler1 Delete coupler2 Production 12 None Hybrid coupler2 1 coupler2 Delete Tones Logging Information Studios Calls Devices Stream Statistics Script Manager Script Information Misc IOManager Info System Status Logs Backup Restore Firmware Individual Show Configuration Pages Telos Y TelosVX Control Center Configuration Main ag Show N Li Studios codi qe Shows Fixed B Address Book Sel l Lines Call Audio Processing Tones Position Name Extension Server Config Fader Busy All Ringer Fixed Configuration Logging 1 6715501 6715501 m 19513208106 Feas gt M Defaut a inrormation 2 Cisco 41 u 192 168 0 23 Fied 2 a Default Studios isco E 168 0 al Calls 3 Ast 3101 3101 m 192 168 0 155 Selectable O Defaut Devices Stream Statistics Apply Script Manager Script Information Misc lOManager Info System Status Logs Backup Re
106. le line conferences on the fly while keeping the usual one call after another auto drop style operation on the left fader In all cases the keypad for dialing is on the Master Module Pressing an options key on a fader channel with a phone line assigned activates the keypad in the master section and opens up some other phone related options such as the auto dial list The keypad is also automatically activated to the fader channel that last seized a phone or codec line However when options is active on a phone or codec module the keypad remains locked to that channel until options is deselected Usually a VSet phone will be used in the studio along with the console to allow handset conversations Taking a line on the VSet will disconnect it from the Element and vice versa The call will not be dropped Fixed lines can be configured directly to Element faders Phone Fader modules are preferred because the set hold and drop buttons are useful Note that using only fixed lines and Phone Fader modules would give you Euro style operation where each line has its own hybrid fader and there is no line selector There is no limit to fixed lines vs selectable in the VX so you can have as many as you have available console faders Mix minus is generated automatically in all Axia consoles In Element mix minus AUTO mode the feed to source will be generated by the PGM 1 bus minus the caller audio when the channel is ON When the channel
107. lines to accept calls When block all mode is selected as lines become available they stop ringing or drop off they will automatically go busy Pressing Block All again will release the lines and allow incoming calls Numeric Keypad and GO Button For dialing out either on the handset or on air The keypad works during an active call to generate tones for voice mail and other services While a number is being composed the digits will appear in the line info field After a number is entered the GO button initiates the connection in similar fashion to a mobile phone Re dial Function After selecting a line press the GO button before any digits are dialed The last dialed number appears in the line info field Pressing GO again causes dialing to start Fader Assign Normally when a call is taken it goes to fader 1 but you can assign it to any of six console faders The number of possible faders depends upon your specific installation and configura tion When a call is on air or before it is taken the round fader assign buttons to the right of the LCD can be used to move the call to the desired fader The number in the yellow rectangle in the line info field shows you which fader is or will be used Fader assign is configurable to be enabled or disabled to keep the LCD uncluttered for people who will not use the feature In our older on air phone systems we were limited by the cost of the day s technology to one o
108. llent stability We ve had success using the PBX in a Flash distribution It is feature rich well maintained and documented and tends to be kept more current than other distributions It s more maintainable than some other distributions and tough to break Basic configuration is done through a built in web server and a powerful happy face web front end called FreePBX 76 Section 5 Another choice that looks good is the Digium product called Phonevox This is a ready to go software and hardware VoIP PBX from the main developers of Asterisk Because it is a com mercial product it has support from the vendor Hardware Unless you are experienced and confident we advise you to buy a pre built configured s w loaded and supported Asterisk machine such as the one Digium sells Nevertheless we offer the following gratuitous advice for those of you who decide to go it alone You are building a machine that will need to be more reliable than most of your regular file servers You don t want to be rebooting your phone system We like to use a 4u Rack Mount Chassis with generous cooling and a higher end power supply 4U cases are inexpensive roomy usually have good cooling with low noise fans have good filtering options and can ac commodate just about any mother board We like to use SATA drives in removable caddies for easy full system backups Choose a mother board with PCI slots if you intend to use expansion cards
109. may display anonymous OPERATION 35 Time The length of time the call has been ringing in or on hold H r R Action hint None active in the graphic Shows what line will be affected by drop and hold Choose the line you want to drop or hold after pressing either of those buttons This is shown only when more than one is possible this step is not needed when only one line is possible when the line would be dropped or held immediately Line Status Icons Dialing Shows when you are using the keypad to dial an outgoing call Outgoing call in progress The icon will be animated with the white highlight moving around when a call is in the process of being connected Ringing in On air and locked On handset o 36 Section 3 Hold e Ready Hold Screened Hold e Used Elsewhere in another studio O Line is blocked S Line Button Columns Each line has 2 associated buttons to the left of the LCD Pressing a left column line button puts a held or ringing caller on the handset Press the right column button to put a held or ringing line on air dropping any other unlocked calls If a call is already on the air pressing the right button locks the line or if already locked unlocks it That is it toggles between the two states Not to worry we ve made it difficult to accidentally drop a call Call screeners or producers should use the Ready Hold button the one with th
110. mbers 250 Maximum active on air calls 30 Number of hybrids one per each active line Maximum number of studios 20 Maxiumum number of Livewire input output channels 20 systemwide Audio performance Analog Line Inputs Input Impedance gt 40 k ohms balanced 4 Nominal Level Range Selectable 4 dBu or 10dBv Input Headroom 20 dB above nominal input Analog Line Outputs Output Source Impedance lt 50 ohms balanced Output Load Impedance 600 ohms minimum Nominal Output Level 4 dBu 4 Maximum Output Level 24 dBu Digital Audio Inputs and Outputs Reference Level 4 dBu 20 dB FSD Impedance 110 Ohm balanced XLR Signal Format AES 3 AES EBU AES 3 Input Compliance 24 bit with selectable sample rate conversion 32 kHz to 96kHz input sample rate capable 4 AES 3 Output Compliance 24 bit 4 Digital Reference Internal network timebase or external reference 48 kHz 2 ppm Internal Sampling Rate 48 kHz Output Sample Rate 44 1 kHz or 48 kHz A D Conversions 24 bit Delta Sigma 256x oversampling D A Conversions 24 bit Delta Sigma 256x oversampling 4 Latency lt 3 ms mic in to monitor out including network and processor loop NOTES RESOURCES ADDITIONALINFORMATION 63 Frequency Response Any input to any output 0 5 0 5 dB 20 Hz to 20 kHz Dynamic Range Analog Input to Analog Output 102 dB referenced to 0 dBFS 105 dB A weighted to 0 dBFS Analog Input to Digital Outpu
111. me more options Most of these are optional except for secret By default the nat field is set to Yes be sure to change it for use with the VX or any extension to be used inside your firewall This is another name for password Scrolling further NOTES RESOURCES ADDITIONAL INFORMATION 85 ei E O 9 gt nuvans 168 9 243 9080menu phprid admin NE Y ab a Oeting Soted EJLatest Headlines JUastiot Greater El 4U Server Short De Paria pared El emanate Aly 5194 Qilkeit Y r E sharev Info Favorites Durbies Tools v x Outbound CID 3115552222 prosa Ring Time Defaut Y CID Supertecta Cal Wi Ee Cater Loop Sources man Y Dayteyt Corera Cal Scanno A oo a lt Fotow Me Emergency CID ma Ping Grupe Assigned DIC Time Conations Tire Oros DID Description Comes Add Inbound DIO utes Add Inbound CO piga Mie Apptcahiom Misc Destinations Music of Moo Paging and intercom This device uses sip technology Gyuem sacret rd Srrimode 2833 ba M Custom Contents Custom Coments Times foet casero We see Recording Options and Voice Mail settings You can record all calls or the extension user can dial a code while on the line to initiate recording of that call Recording is optional of course Below the field marked Secret is another called DTMFmode which should default to RFC2833 Leave this as is The default is the preferred method of handling DTMF tone trans mission RFC2833 uses SIP
112. ment and services 52 Section 5 IP Centrex and Hosted PBX Services Just as it shouldn t matter whether the PSTN gateway is on your premises or not it also shouldn t matter where your IP PBX is located This is the principle that allows IP Centrex services or hosted PBX services to happen These locate the hardware at the service provider s site and remove the need for phone system equipment at your location In the pure case you would have only IP phones at your site which would be plugged into an Ethernet switch which would connect to the Internet via a router The main advantage is that someone else is responsible for installation and maintenance of the back end equipment It might also be that a vendor of these services has invented a suite of applications that would be difficult to replicate at individual business sites Number portability in the US and Canada Number portability regulations allow you to move your existing numbers to the service pro vider of your choice Unfortunately our experience is that choke exchange numbers cannot be ported We ve tried with various carriers all with the same disappointing result a call on the day of the port explaining that it can t be done Circuit Switched Interfaces When you order or configure a gateway you need to know what kinds of interfaces you will be using to connect to the Telco network FXS FXO These are designations for the two ends of a standard anal
113. messages to pass detected DTMF tones along to the other end of a call rather than generating audio tones locally The Voicemail system in Asterisk offers all the usual capability You could enable it on our about to be created extension 2222 here so that you could see some of what it can do Set the password We set up this mailbox to automatically send us a wave file of any messages left to the email address in the field below The audio quality of the recordings is impressive and usable for on air playback This is also optional scrolling further down e O 9 nupins 168 0 243 9080 menu chprid sdmin Ba Most Visited Y SH Oeting Rated latest Headlines JU Ewah El 4u Server Short De ASA pared al Stumble Aly 194 ie Y vr El Sharer info Favorites LSMumbies Tools Y 520 Emai Address Mmenogpubic com Pager Emai Actress Emal Atactenent tye wo pa Play CD ys Sm Play Envelope ys m mw We are finally at the bottom of the page for extension 2222 86 Section 5 e E O D Min 109 020 90r0mena ponosan r a O PA F Gething States EUes Meadires JSiesior Elwevrer il 4U Server Shot De Ana shared T Oaume Aly 6196 bilte v E sheer Oto 4Fevodtes Stumble Tools y Fre M a Ic Status maes Y Volcemal Passeoro 520 Eras Adcress Meno poe com Pegar Email Address Eras Attachment tyes no Ply CID yes n0 Play Ervelope yoo no Delete Vexco as yen no VM Optors 4 VM
114. mix minus matrix of some kind within the studio mixing console Studios and Shows The VX system can support multiple studios and can share lines among them A studio is a collection of controllers and other devices that are used together usually to serve one studio VXsets VX Producer applications on PCs Audio input and output Nodes if any GPIO interfaces if any and console controllers if any A show is a pre configured profile that can be selected by users It assigns the lines to buttons and such This is the mechanism that lets you move lines from studio to studio Studios and shows are configured via the web interface to the VX Engine Interfaces such as the VSet and the VX Producer let operators select the studio and show they want to use INSTALLATION AND CONFIGURATION 13 Configuration Web Pages To access the configuration Web pages find or connect a PC to the network that the VX Engine is connected to Start a web browser and open a connection to the VX Engine by entering http 192 168 2 200 into the browser s address field assuming the default IP number has not been changed and that your PC is on the same subnet Home Page This is the page you will see upon initial connection Select the pages you want by clicking on the menu at the left side of the page If this is a new installation start with the Main page then move to the SIP page the Studios page and then on to the Shows page The purpose of each of th
115. more than one call reducing the effective IP overhead without creating additional latency As IAX s name indicates it was invented by the Asterisk people as a way to trunk calls between one Asterisk server and another It has escaped from Asterisk and is now supported in a variety of softswitches and by a few VoIP carriers Its main advantage is its bandwidth efficiency and simpler firewall configuration since all traffic flows through a single port Using VX with the Asterisk PBX Asterisk is a great way to provide extra flexibility and power to your VX installation The software is inexpensive or free and runs on a standard PC It s an open source project having been created by teams and individuals from around the world often volunteers Despite this we ve found it to be reliable At Telos we count on it daily to network our sites throughout the world A few VX installations are already using it You ll find that it s a useful tool if you want to do Interactive Voice Response systems support on or off site VoIP telephones call detail recording failover systems ISDN PRI channel splitters and such With appropriate hardware Asterisk can also be used as a gateway For example you may need to peel a few channels off of a PRI to send them over to the VX while the others feed a PBX An ISDN PRI card in an Asterisk box and some configuration would get this done NOTES RESOURCES ADDITIONAL INFORMATION 75 Because it connects to the
116. mpile the program over the next 20 minutes or so depending on your machine and connection speed Don t interrupt the process It may look like not much is going on but watch the activity of your hard drive LED It should blink periodically Messages on the screen will update every once ina while The system will firmly suggest that you do not press any keys to interrupt this process Upon completion the system will reboot by itself and you ll then see the colorful PIAF splash screen followed by a normal boot up that should leave you with a login prompt TIP You can always choose the defaults unless you ve gone off the reservation and are trying to do something unusual or complex in which case you re probably smarter that your writer In other words you re on your own Google and your Linux geek are your best friends The good news is that many people have done what you re doing and have had the same problems that you have Fortunately they often share solutions on the web Log in to the Console for the first time No not the mixing console Console is the Linux term for the command line interface You ll be greeted with a friendly if spartan login screen that looks like this CentOS release 5 5 Final Kernel 2 6 18 194 8 1 e15 on an x86_64 pbx login _ Enter root and hit lt enter gt Password _ Enter the password that you set for root earlier in the setup and hit lt enter gt You should get a ful
117. multiple lines and keep VIP callers on air while coming and going with others This also allows selective holding and dropping of conferenced lines Multiple calls that are assigned to a single fader have individual hybrids processing and are actively conferenced within the VX Engine using an internal mix minus matrix Calls assigned to different faders would normally be conferenced via a mix minus matrix of some kind within the studio mixing console With fixed lines there is a one to one correspondence between Telco lines and console faders A particular telephone number is always associated with a particular button on the VSet phone and a particular fader on the console It is as if each line had its own hybrid This allows VIP and hotlines to have fixed and dedicated console faders Perhaps this will be an easier operat ing paradigm for stations that often have multiple lines conferenced together on air You could configure a system to have only fixed lines and no selectable lines Each line would have its own dedicated fader This is in fact the way most large European stations have oper ated their phone interfaces for years This approach has become more practical as we move away from analog lines since the lines are virtual and the cost for a telephone number added to the ISDN T1 or SIP trunk pool is much less than the cost of an analog line usually less than 1 per month Fixed line calls would normally be conferenced via a
118. native o ooo ooo 74 Using VX with the Asterisk PBX 2 ee ee ee 74 Asterisk software Distributions 2 2 ee ee ee 75 Had WAS i aes hae Aw A RE ae Oo we ae 76 EXpansiOn Cards o tana Cake ds eG Ba dee 77 TGetyo distro arenosa a 78 2 Get loaded Install Asterisk 2 ee ee 78 Log in to the Console forthefirsttime 0000008 80 3 Confirgure Asterisk nanala 82 MUNKS atado AA ee Aan a 86 Outbound alls 2 ee ee 89 Connect your VX to Asterisk ooo ooo 89 viii Make a test call on your VSet o o o o o o o oo ee 90 VX Tech Cheat sheet Default IP numbers passwords etc 90 Default Passwords e 2 4 each wa ba eho Pa Reda g ae whee 90 Default IP Addresses 2 2 2 ee ee 91 SIP Patametels ce sich Git eG a ale wanes ah a OE oe T 91 TCP IP Ports and protocols used by the VX System 91 VSettelephone is g e ga eaaa es 91 Default username and password oaoa ee 91 TCP IP Ports and protocols used by the VSet Telephone 91 Power Over Ethernet PoE 91 A 92 Application Example WKSU o o o o o o oo ooo 97 Warranty 101 1 Welcome Note from Steve 133 years ago this past summer a tinkerer from Ohio scratched a kid s rhyme into tin foil wrapped around a cylinder Despite the less than hit grade content the warbly tone and the lack of bass response Tom Edison s demonstrations astounded guests and founde
119. ne 2 Logitech USB Hea Left Right Outputs O Two One Speakers 2 Logitech USB Heads Left a Two Remove OK Browse Cancel THE VX PRODUCER APPLICATION 43 Set up Studio and Show File Help Studio Show P an OD Line 10 102 Managing Calls The main window shows a list of the lines that are available for the selected show It is divided into selectable and fixed line groups if both are configured for the selected show Near the bottom of the screen is a tabbed list box controls and an audio level meter that are used for the recording and play functions At the lower right is a window used for text chatting between the producer and talent Handset l Press the handset button to the left of any line to take the call to the handset that is connected to the PC running the VX Producer application Often this will actually be a headset mic If no call is ringing in the system assumes you want to dial out and opens a window to let you do that Y Dial o B X Wl Contacts D History we L 1 2 3 L 4 5 6 7 8 9 L x 0 4 call Close Contacts History Dialing 44 Section 4 On air Pressing this button takes a call to the studio mixing console Your operation of the console determines whether the call is actually on air or sent to preview or to somewhere else Usually the studio mic or microphones are used to talk with a caller
120. notes etc It also reduces cost as no hardware phone need be installed It has a built in audio recorder editor A producer can record and edit a phone call without leaving the application The resulting audio files can be easily sent to the talent PC for quick and convenient airing We imagine that this simple but powerful feature could add an interesting new element to many call in situations Callers who don t have time to wait their turn could be quickly recorded and carefully edited Perhaps montages could be created to open segments Removing the need to wait might encourage comments from higher quality callers Because Livewire audio is available at the PC via standard LW Driver software a producer can readily record calls for later play These could be edited with a PC application running on the same machine When a file has been produced it can be sent to the on air studio over the network WELCOME 5 Thus the one Ethernet cable is used for Telephone audio for the softphone via Livewire 4 Livewire audio for the recording of calls Transfer of recorded call files from the producer to the studio Data messages between the PC and the VX main box for line selection etc Data messages such as call notes and IM between the producer and on air studio Possibly database lookup of caller information such as how many times they have called the quality of their contribution whether they have won any contes
121. nt good quality and having any chance of getting problems resolved The other thing to look out for is what codec will be used For calls that ultimately are carried by the PSTN only the native g 711 codec is acceptable for broadcast applications Anything else would involve transcoding and an unacceptable reduction in fidelity especially audible when mobile phone calls are involved These already have poor quality due to their low rate 14 4kbps codec Passing this through g 711 within the PSTN and then yet another codec on the way to your studio over an IP link is asking for aural trouble The VX natively supports the g 711 A Law g 711 u Law and g 722 codecs Almost all PBXs and gateways support these formats Finally you need to be sure that your equipment and the carrier s gear can properly communi cate While SIP is a standard vendors often enhance it with extensions that are not universally supported One development that could help is a project called SIPconnect undertaken by SIP Forum a consortium of SIP vendors The SIPconnect Interface Specification was launched by Cbeyond Communications in 2004 with support from Avaya BroadSoft Centrepoint Technologies Cisco and Mitel It attempts to detail the interconnection specifications between IP PBXs and VoIP service provider networks It specifies a reference architecture required protocols and features and implementation rules It calls for the g 711 codec to be provided on all equip
122. oIP means The VX uses it in two distinct ways One it can connect to Telco services using standard SIP VoIP You benefit from having options connecting to PBXs digitally to ISDN and analog lines via gateways etc etc With VX you can finally integrate your on air phones with office phone systems from a variety of vendors Getting Telco service from VoIP dial tone providers means that your audio quality and hybrid null will be much better as VoIP dial tone is delivered 4 wire without hum noise and loop loss Building on ubiquitous VoIP standards means a variety of third party hardware can offer flexibility And you might save a lot of money getting service this way Two VX system components connect to each other over standard IP Ethernet networks with all the advantages that brings For example in Livewire equipped facilities one RJ 45 jack connects dozens of audio channels and rich control to phones like controllers PC applications integrated console controllers etc So I can use the VX with my regular 1MB POTS lines How Can I use ground start lines for incoming calls only You can do that and it s not difficult You only need a POTS gateway device However we encourage using digital delivery for the best sound quality What about ISDN BRI and PRI lines If you have ISDN now and want to keep it again there are gateways available However it is often cheaper to port these numbers to a VoIP dial tone provider All of these
123. og POTS line Most often these lines are used for basic home telephones But they can also be used to link a PBX with a Telco central office An FXS Foreign Exchange Station interface emulates a circuit supplied by a Telco Central Office An FXS supplies talk battery and detects an off hook condition It gener ates 100Vac for a ringing indication It provides dialtone and other call progress signals such as ringback and busy It responds to DTMF Dual Tone MultiFrequency tones for dialing and may send caller ID information in modem encoded audio A telephone and anything that looks like a telephone is an FXO Foreign Exchange Office device An FXO device signals an off hook condition by drawing loop current It responds to ringing voltage It provides dialing either by old fashioned pulsed loop interruption or by DTMF It may detect caller ID An interesting limitation of FXS FXO interfaces is that signaling from the FXS that a call has ended is sometimes not signaled and the type of signaling varies around the world In the USA most Telco central offices interrupt the loop current when the call has ended but some do not And many PBXs do not Eventually dialtone will return though and this can be used as a disconnect signal but there will be a many second delay This could cause glare the condition where there is confusion between the CO and the PBX as to whether the line is free A call could ring in just when the PBX attempts to
124. oice over other man aged IP networks If you have been keeping up with the transition to IP Codecs you probably have noticed the same terminology issue there IP based STLs over an IP T1 are just as reliable as traditional STLs over traditional TDM T1 circuits You are completely right to be concerned about a VoIP trunk or an STL over the Internet as that is not at all the same thing and perfor mance in that case could be variable Inside the facility on a LAN all problems disappear since you have plenty of bandwidth and full control over the network So shared bandwidth is the only problem with VoIP Well there s audio quality In the early days VoIP used a lot of compression with bit rates being as low as 6kbps Needless to say the resulting audio was not impressive These low rate codecs have mostly fallen by the wayside The lowest grade codec the VX supports is g 711 the standard for digital audio in the telephone network pre IP And you will eventually benefit from higher fidelity codecs as these proliferate in the VoIP world How does AolP relate to VoIP Despite the similar names and underlying technologies they are very different with regard to performance and application An analog phone line and a balanced 6002 studio audio circuit are pretty much the same tech but the applications and performance are very different AolP has come to mean professional studio grade audio networking full fidelity and usually wit
125. on a T Script Manager Script Information Misc lOManager Info Backup Restore Firmware SY isa 20 20 Ena Ey The VX has dynamics processing on both the send from studio to caller and receive caller audio directions Adjustments and meters are on the hybrid page We recommend as a starting point 8 on the high EQ 6 on the low EO 16 on the AGC full the caller ducking to 6dB and the noise gate off The send processing consists of a limiter and EO The purpose of the limiter is to protect the Telco line from the distortion that would result from clipping due to an audio overload Livewire studio audio has a much larger dynamic range than telephone lines We use a limiter rather than an AGC because we don t want the system to increase low level signals which could cause feedback and sound unnatural to the caller The receive processing includes ducking an AGC and dynamic EO Hybrid refers to the the traditional hybrid cancellation function the removal of the send audio from the received caller audio required due to the nature of analog Telco lines where the two signal directions are combined on one circuit In the VolP world this should be done in the gateway and is not required in the phone system itself but we use the term hybrid here because it s instantly understood Some PBXs refer to the hybrid as an LEC or Line Echo Canceller The Call Audio Processing button allows you to enable or disable all of
126. options are worth considering So can experiment with SIP VoIP trunks Why would want to do that Actually we recommend it We think that you ll find that they work better than you may have expected as many of the VoIP problems we have seen are caused by limitations of analog terminal adaptors IP to POTs gateway devices Since these are not needed with a VolP based system such as the VX that class of problem is eliminated There are a number of inexpensive ways to try VoIP without risk You can also get VolP delivered numbers from distant area codes and exchanges If you re paying mileage for foreign exchange lines or have national toll free numbers you ll definitely want to consider this option l can t put a flasher across a VoIP line so how can I flash a light when the hotline rings This same issue arises with ISDN so beginning with our TWOx12 we included a GPIO output for this function The Not to worry The VX has this capability In fact it has multiple oututs which can be assigned to any of your VoIP lines have been watching VoIP with interest But reports have heard about services such as Von age are that sometimes they work well but other times not Frankly surprised to see Telos advocating it We get this concern often and understand why you ask The term Voice over Internet Protocol NOTES RESOURCES ADDITIONALINFORMATION 93 VoIP does not distinguish between VoIP over the Internet versus V
127. os systems com extension phone 123 at Telos Systems If you don t have a name or extension you might want to contact the receptionist sip receptionist telos systems com Here s an internal machine to machine message such as from an on air phone system to a PBX or gateway to initiate a PSTN call sip 12162417225 168 123 23 1 Note that in this case an IP number is provided to identify the concrete machine that is to receive the message You usually don t want to use DNS for this because it takes time for the look up step and because there may well not be a DNS name associated with a machine being used as a telephone server To assist readability SIP lets you use and separators It removes them prior to pro cessing sip 1 216 241 7225 telos systems com As you can see SIP bridges the telephone and Internet worlds Both Web type and PSTN telephone number addresses are possible and users on either network can reach those on the other Often address resolution involves multiple steps and SIP message hops A DNS Domain Name Service server a SIP proxy server and a SIP redirect server might all be involved in a single name resolution for example A few other points of interest regarding SIP Some servers associated with SIP systems can accept unformatted text names but this is not part of the standard URIs are not URLs URIs are independent of the location of the named object Email addresses are
128. ost Visitedw e Getting Started latest Headlines JGanda Ejvienther El 41 Server Short De BWA BAN Pp 192 166 0 248 9080RnENU phohd admin FreePRX System Sims Meca Aci 27 ade Toot SIPSTATION Extensions Fearre Codes General Settings Ortbound Routes Treks Arista Inbound Routes Zap Chennai OIDs denouncemerts mous CID Supertacts CalleriD Lookup Sources ODay Nigh Control Foton Ue Dore Edit SIP Trunk Dolote Trunk vitel In use by 2 routes Gereral Settings Outbound Caller D Never Override CalleriD Maximum Channets amp Disable Trunk w09 Moritor Trunk Failures Outgoing Dial Rules Dial Rules Add Trunk Trunk ZAPIQO Trunk SiPAncenpet Trunk SiPigrandstroam Trunk lAXZihore Trunk AX2 hgo net Trunk AX2itairhit Trunk AX2ANOS Trunk SIP vite Trunk SPa Trunk AX2icnet Trunk AXZKMI In our example here we only use Vitelity for incoming calls so the Outgoing Dial Rules are irrelevant We could insert a period in the Dial Rules box that would simply pass whatever digits were dialed down the trunk to Vitelity Keep in mind that this is just the trunk setup and that there still has to be inbound and outbound routes to get calls into and out of the switch O gt tetera 168 0 248 paoman phe ide a rmin Misc Appicatons Mac Cestnatons Musc on old Paging and intercom len Custom Corterte Custom Cortexts Times Se Calero Vocemal Adan Ba Most Vis
129. pages are to configure each studio For each you can choose how many faders you want to devote to selectable and fixed lines and you assign the Livewire channel numbers to each Note that when you assign Livewire channels to outputs Livewire streams will start to flow on the LAN network port With a proper Ethernet switch this is no concern but should this port be connected to an office LAN or non Livewire switch there could be flooding from the audio traffic Recall that Livewire channels carry audio in two directions so both send and receive are covered at once Live Stereo should usually be chosen as the Mode to keep the delay as low as possible One LW channel is used to feed Program on Hold to each studio Since only one of the two Livewire audio directions is needed you can choose which is appropriate for your studio set up From Source is the usual INSTALLATION AND CONFIGURATION 21 Individual Studio Configuration Pages Part 2 IOManager Info System Status Logs Backup Restore Firmware GPIO Indications There s a Next line There s a line that can be put On Hold There s a line that can be Dropped Block All state indication Auto Hold state indication Mute On Off NONE Advanced receive opens a ne
130. r two hybrids so we only used one or two console faders With the VX system this limitation is removed That means that you can assign a fader to each call if you want to do so allowing independent control of each call s volume Firmware Upgrades You can upgrade firmware for your VSet by using a web browser and downloading the firmware from Telos Support may give you a link in email or you can find the latest versions at the Telos website After downloading the firmware image point your web browser to the IP address of the VSet that you wish to upgrade the default password is lt blank gt and the username is user Then browse to the firmware image location on your PC desktop or downloads folder is common and select Upload To activate the new firmware select it in the browser and the VSet will reboot When it restarts it will be running the firmware version that you selected OPERATION 39 Take it Easy We ve built a lot of flexibility into the VX system to support all the sophisticated things you might want to do But you don t have to use all the fancy pants stuff when you don t need it and a simpler subset will do In the simplest case when you are a talent supported by a producer you could just use the Next and Drop buttons You press the Next button to take calls in the sequence determined by your producer and you press the Drop button when you are finished with your last call and you want to stop taking more
131. rmined by install time configuration Normally VIP and hotlines are excluded Fixed Lines These are usually used for VIP and hotlines They have a different operating style from select able lines and are independent from them Here each line has its own fixed fader These will have been assigned when the system was installed Taking a line will not drop others that are active Drop and Hold need to be explicitly applied Lock has no effect on fixed lines since they are effectively always locked Block All is usually configured to have no effect on fixed lines Fader assign has no effect since faders are perma nently assigned Note that a system could be configured to have only fixed lines This would result in the operating style that many large European broadcasters favor Each line would have its own hybrid and associated fader Fixed lines also offer a selectable auto answer feature This can be used for automatic feeds news recording or filing lines or listen line coupler replacement Recording Editing and Playing Calls Pressing the Record button starts a recording of the currently active phone call THE VX PRODUCER APPLICATION 47 In talent mode this will be the on air call Usually the audio will have been routed via the studio mixing console In producer mode this will be the call on the handset You can choose if both the local mic audio and the phone are recorded or only the phone
132. rob ably better for you to use than trunks First SIP lines each have their own outgoing caller ID programmable per line in Asterisk Trunks will only send the extension number for caller ID or the same the caller ID for all of the numbers on the trunk which can be confusing People have become used to getting accurate caller ID on their mobile phones etc You may wish to hide or substitute caller ID on Private lines or hotlines Another feature of SIP lines is the registration feature With it the system knows that an extension is ready and available for calls 90 Section5 After adding the line you can assign the extension to line buttons in show configuration Once the show is loaded the information page on VX will show the current status of your lines and trunks Notice below that extension 2222 and all of its multiple appearances is OK and registered The others say OK and trunk This is because the trunk is assumed to be present Extension 2222 has been verified as present Now we ll go to the VX Shows page and look at the show that we built earlier It has a 6 line hunt group for extension 2222 and a couple of back lines or hot lines It s a hunt group because there are 6 line keys set to extension 2222 Optionally we could have the busy all box checked only for extension 2222 our public caller lines Only these lines will be busied out when we press the block all key on the
133. rs etc The VSet Phone Controller While you can control the VX with PC applications and mixing consoles Most systems will include one or more Telos VSets These are the phone like controllers that have handsets for off air conversations IP based Telos VSet phones have large high contrast color LCD panels that provide line status and caller information Caller description text that is entered into compat ible call screening applications will show up on the LCD near the line select buttons For the comfort of familiarity the VX can work like a traditional Telos controller with calls being selected held and dropped in the way to which operators have grown accustomed But because the VX system has a hybrid per line it is often desirable to spread multiple calls over a number of faders using one for each call so that operators can control each line s level individually The VSet offers this possibility It is also possible to hard assign individual lines to fixed faders such as for VIP calls VX Producer Windows Software Application The VX Producer application takes studio phone operations to a higher plane It provides the usual call screening functions for phone active broadcasts but with a number of enhancements enabled by the IP nature of the system The integrated softphone uncomplicates the producer s life since the PC interface is used for all operations including answering and making calls assigning priority writing
134. rsion RFC 3261 Session Initiation Protocol SIP latest version RFC 3264 Codec negotiation A H Inglis Transmission Features of the New Telephone Sets Bell System Technical Journal 17 1938 358 380 Just for fun Waaay back Mr Inglis proposed high fidelity phones noting that cost was the only barrier to better audio quality in the telephone network and predicting it would become common in the future Took awhile Books Henry Sinnreich and Alan B Johnston Internet Communications Using SIP John Wiley amp Sons Inc New York 2001 Gonzalo Camarillo SIP Demystified McGraw Hill New York 2002 Steve Church and Skip Pizzi Audio over IP Building Pro AolP systems with Livewire Focal Press Burlington MA and Oxford UK 2010 Mighty fine fellas these guys Good book too Jonathan Davidson and James Peters Voice over IP Fundamentals Cisco Press Indianapolis 2000 J Alexander C Pearce A Smith and D Whetten Cisco Call Manager Fundamentals 2nd Edition Cisco Press Indianapolis 2006 NOTES RESOURCES ADDITIONAL INFORMATION 67 D Au B Choi R Haridas C Hattingh R Koulagi M Tasker L Xia Cisco IP Communications Express Call Manager Express with Cisco Unity Express Cisco Press Indianapolis 2005 Oliver Hersent David Gurle and Jean Pierre Petit P Telephony Packet Based Multimedia Communications Systems Addison Wesley Reading MA 1999 Switching to VOIP O Reilly
135. rtunities for enhancement of its capabilities When the connection from a station s listeners to its studios eventually evolves to become via IP as it is near certain to do things could change appreciably We ve already touched on the possibility for higher fidelity But that s only the immediate and obvious next step With an un constrained pathway for data along with voice a talk show s producer or host could text chat with a caller prior to his being on the air for example There could be an automatic updating of the time a caller is planned to be on air There could be instant voting A window could be kept open on listeners PCs that would deliver supplementary text or visual information Peering yet further into the future should video streaming catch on we might want both see and hear callers Here are some other things we have in mind A gateway to Skype allowing callers to connect via computer with higher quality than usual phones It could be possible to exchange chat messages to update callers on how long they need to wait etc Build in a texting SMS gateway to allow this form of communication with mobile phones Make a simpler VX phone for applications that don t need the fancy LCD displays We invite your suggestions and creative ideas NOTES RESOURCES ADDITIONAL INFORMATION 61 Beyond Edison s Legacy The PSTN uses the g 711 codec Its audio frequency response is limited to 3 4kHz A modern read
136. rvice don t worry all is not lost as VX has basic NAT support built in First forward the ports used by SIP 5060 and 62000 62512 from the router to the VX This is necessary to allow outside traffic to reach VX Then enter the external WAN IP of the router in the External IP configuration field for the server Setting it will cause VX to use the external IP for all SIP messages it sends to the server while still leaving room for the server side hacks The setting will take effect only for the particular server Overall Studio Configuration Page Telos VX Control Center TelosVX Control Center Studio Configuration Fixed Selectable Active Studio Name Channels Channels Show Studio 1 2 4 Show 1 Studio 2 0 2 Show 1 The Studios page lists all of the studios that you have configured for your system It lets you add new ones And it lists the Shows that studios are using 20 Section 2 Individual Studio Configuration Pages part 1 y R TeowWX Control Center LO C fi 192 168 0 7 SFun StudioCtg 4 Telos VA TelosVX Control Center Co Sel 1 Configuration SPR y Studios Studie a using how Lattelecom Feed 1 az a Gn Address Book Fed 2 Selectable Anger Fwad 3 Sudoneme o Sane Fani ias Coupler1 Coupler2 Shows Hybrid Tones Logging Information Studios Calis Devices Stream Statistics Script Manager Script Information Misc JOManager info The individual Studio
137. s 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 accessories 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 1 216 622 0247 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 30 AM to 6 00 PM USA Eastern Time Monday through Friday Our fax is 1 216 241 4103 By E Mail Technical support is available at SupportO 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 sup port The URL is www Telos Systems com 1098765432 Telos Systems USA Telos Systems 1241 Superior Avenue E Cleveland OH 44114 USA 1 216 241 7225 phone 1 216 241 4103 fax 1 216 622 0247 24 7 Technical Support
138. s Operating Temperatures 4 10 degree C to 40 degree C lt 90 humidity no condensation Dimensions and Weight 3 5 inches x 17 inches x 15 inches 10 pounds Studio Audio Connections Via Livewire IP Ethernet Each selectable group and fixed line has a send and receive input output Each studio has a Program on Hold input Each Acoustic Echo Canceller has two inputs signal and reference and one output LW equipped studios may take the audio directly from the network Interface Nodes are avail able for pro analog and AES3 uses standard Livewire Nodes Telco Connections Audio standard RTP Codecs g 711u Law and A Law and g 722 Control standard SIP trunking Resources Internet Speed tests www speakeasy net www speedtest net Internet VOIP Jitter Test myspeed visualware com indexvoip php free NOTES RESOURCES ADDITIONAL INFORMATION 65 Packet sniffer www wireshark org download html free VoIP Soft phone SIP PC clients X Lite Soft phone SIP PC client free www counterpath com x lite html Ekiga free open source SoftPhone ekiga org Linphone open source VOIP phone www linphone org Open Source PBX Distributions www digium com pbxinaflash net www trixbox org www freepbx org www asterisk org Commercial PBX Products www microsoft com voip www lg nortel com www trixbox org www siemens com hipath www mitel com www digium com Gateway products and suppliers
139. s Use wwwadmin and the password that you set earlier then click OK You should see the screen below Apphcabons Places Symeno a Gass cass acom Dud de Ge Turer 7 eism Ojos De t t Yew Watery Deokmarts Dos Help G CO OL lennon R gt Me O MOR Valeo Ger States Cluster mentiras EJainco F Calas Logn lr a Mama DIOS dy marvel fie bro lioe file bwase VeManabiTO6 inbox 20562 un 56H Manel Men SRE J Use FreePBX to do some administration chores Click on FreePBX Administration and you ll see a new menu that supports almost all of the features you ll regularly use in Asterisk You ll do most of your setup from this menu Here s what we re going to do next Create an extension that we ll later use with the VX Set up a trunk to accept incoming calls and to be used to make outgoing calls Set up an inbound route to tell incoming calls how to get to our extension on the VX 84 Section 5 4 Set up an outbound route so that you can make calls out Then we ll use the VX web interface to use the line that we created and then make a test call Finally we ll create a trunk from the PSTN and route calls from that trunk to our extension all by using the web GUI We ll go a step at a time from this point Create an extension or two To get comfortable with Asterisk create an extension or two to use for testing You can use the extension to test your connection with the
140. s leave the Phone hybrid IP field empty leave the Phone line and Phone hybrid for NX12 fields at their default value of 0 Use 2nd show split mode for NX12 should be left unchecked Phone hybrid ip Phone line 0 Phone hybrid for Nx 12 0 Use 2 show split mode for Nx12 al Phone Module ID E Hybrid Nr 0 None 1 Conferencing allowed No Fixed line 1 to 24 0 No 0 NOTES RESOURCES ADDITIONAL INFORMATION 59 The Phone Module ID field lets you select from the multiple phone modules that you may have installed in your Element Normally you will have just one so select 1 from the drop down box For Hybrid Nr select the hybrid that corresponds to the Primary Source audio you selected above The VX supports an unlimited number of hybrids per studio To find out what Hybrid number corresponds to the audio source you selected go to VX Engine Studio configuration page and check the sequence number of the Selectable LW channel that you use as the audio source here The first Selectable LW channel counts as Hybrid 1 Conferencing Allowed lets you permit button mash conferencing Unless you specially wish to prohibit this type of conferencing select Yes from the drop down box Fixed Line lets you assign a line permanently to a specific fader channel emulating a dedicated hybrid Normally this is used with a 4 Phone Fader Module If you want to use a Hybrid in this mode specify one of your VX system s
141. s needed treating each line as a trunk If this is what s needed you don t have to do anything else Otherwise you can change the server settings and line specific configuration from the Server Configuration pages It shows the server address and name and the list of configured lines lines using defaults are not shown The SIP Server field allows you to change the server address while keeping the other settings intact You can also assign a more descriptive Name to the server which will be then shown in place of the server address As it is used for display purposes only it can be anything you want like POTS Gateway The External IP setting is discussed in the next section Please note that to change the server settings you have to unload any shows referencing the server it is not possible to change them while the server is in active use The Extension and server together make up the SIP address of a line It must be unique as in the context of SIP the line is the address and associated configuration To change the line from trunk to station mark the Register checkbox This will make the VX register it with the server meaning log in whenever a show referencing this line is active informing that the address is reachable The Expires field lets you to change the interval in INSTALLATION AND CONFIGURATION 19 seconds in which the VX will refresh its registration When left empty the server and VX will negotiat
142. s you can use the lock function Press an already on air line button to lock it The line status icon will change to display the locked symbol Pressing the button again unlocks the line and the lock icon goes away Locked calls remain on air until unlocked and then another call is taken to air or the call is dropped or held The Drop and Hold buttons have no effect on a locked line Tip If configured in your system using fixed lines is another option for handling guest or VIP calls that need to say on air while you switch between other callers With fixed lines you d have a dedicated console fader for each of these calls so you can control volume and switch them independently in and out of the conversation etc Next Button and Function The Talent next priority is 1 longest waiting ready hold 2 longest waiting hold 3 longest ringing in The producer can manually override these and assign priority as desired The producer mode next priority is 1 longest ringing in In producer mode Next does not take any held lines Block All S Pressing this button will cause all inactive and ringing lines to be dropped and blocked from accepting any calls Calls on air on the handset on hold and the fixed lines will not be af 38 Section 3 fected The usual application for this function is to let you prevent early callers from getting in on contests until after you ve made the announcement and released the
143. s not wanted A suitable switch would be the Cisco 2960G or another from that family The switch that is part of the Axia Element PowerStation is also suitable Of course the VX and Axia consoles were designed to work smoothly together For the initial out of the box configuration the foregoing is not a concern You can safely use a simple Ethernet switch or an existing LAN to get your configuration PC to the VX LAN port INSTALLATION AND CONFIGURATION 11 because by default there is no Livewire traffic flowing Only after you assign Livewire outputs in configuration as described below will this be a concern The lower WAN Wide Area Network port is isolated from the Livewire port and thus provides a firewall ensuring that no traffic can pass from the Wide Area network often connected to the Internet to access VoIP to the studio Livewire network Out of the box the defaults for the network connections are The WAN port is disabled To use it you must enable it on the Main configuration page as described below and move the services you want to use to it also on the Main configuration page All services HTTP web access SIP VoIP VSet audio and control and Livewire audio I O are provided via the LAN port No Livewire inputs or outputs are assigned or enabled You do this on the Studios configuration page VSet Installation VSets are normally connected to the Livewire LAN Engine port via the Ethernet switch l
144. set or on air into hold If more than one line is in this category they will all be highlighted Press the line that you want to hold Or select the line you want to hold to highlight it and then press Hold THE VX PRODUCER APPLICATION 45 Hold Ready puts a call that is ringing on air on the handset or on hold to the hold ready screened state Selectable Lines Lock Unlock Button and Function Lock Normally taking a call to air causes any others on air to be dropped If you need to conference two or more calls you can use the lock function Press an already on air line button to lock it The line status icon will change to display the locked symbol Pressing the button again unlocks the line and the lock icon goes away Locked calls remain on air until unlocked and then another call is taken to air or the call is explicitly dropped or held The Drop and Hold buttons have no effect on a locked line Tip If configured in your system fixed lines will probably be the better option for VIP calls that need to say on air while you come and go with others You will have a dedicated console fader for each of these calls so you can control volume switch them independently in and out of the conversation etc Fader Assign Buttons Fader1 Fader2 Fader3 Fader 4 In older on air phone systems we were limited to one or two hybrids audio paths so we only used one or two console faders With the VX system this limitation is remo
145. settings where the username is user and the password is empty you don t need to enter anything Configure your VX Studio and Shows In the same screen where you entered VX IP num ber you can enter a VX Studio name In the Host Studio Name field simply enter the studio name you used in your VX s configuration In the example above the VX Studio name is Studio 1 If you wish to change a VX Show using an Element Show Profile simply specify the name of the VX Show you wish to load in the Show Name field If you leave it empty no VX Show will be changed and whichever show is currently loaded will continue to be used Leave the Show Password field empty There are no passwords for Shows in the Telos VX so none are needed from the Element Configure other options Element s Call Controller module uses the left row of telephone line selection buttons to control Hybrid 1 the right row to control Hybrid 2 If you wish to use the right row of Call Controller buttons to control Hybrid 1 and the left row to control Hybrid 2 click the Reversed Hybrid box VX can be used in either 12 or 24 line mode You can select either Auto 12 Lines or 24 Lines with the Mode Selection radio buttons In 12 line mode 12 lines are displayed on both columns of the Call Controller module the left key bank answers lines on Hybrid 1 the right keys answer lines on Hybrid2 In 24 line mode both left and right key banks will answer lines using Hybrid 1 To ans
146. sfully unaware of all this will undoubtedly continue with their conditioned habit referring to a certain caller as being on a particular line Thus is the word line destined to join dial in a peculiar departure from original meaning In this manual we refer to lines as your users would Usually that means a particular tele phone number that is probably associated with a button on the VSet phone and or console controller To keep the ambiguity down we ll try to remember to call the physical links con nections or interfaces And we ll sometimes refer to calls when the sense points more to the conversation than the connection INSTALLATION AND CONFIGURATION 9 Livewire for audio 1 0 The VX uses Livewire for all audio I O For studios that are already Livewire based this ap proach saves money and simplifies installation With its simple support of bi directional audio flow Livewire is ready for hassle free mix minus When needed traditional audio connections are provided via VX Interfaces with Axia Livewire connectivity These come in both analog and AES3 versions Because they are networked they can be located where convenient either in individual studios or in a central rack room Each standard Interface connects eight stereo audio inputs and eight outputs The many audio converters and connectors that would be needed to interface the multiple hybrids and program on hold inputs are reduced to a singl
147. sier better and at lower cost But it also paves the way for a richer more natural connection with your listeners as the IP platform becomes the basis for tomorrow s creative applications It will be interesting to see what comes next Warm regards Steve Church Founder April 2011 2 Section 1 What s the Big idea Is it not a bit strange that many computer laden all digital broadcast studios con nect to the PSTN Public Switched Telephone Network using technology invented in the day of phones with hand cranks and bulbous ringers Where else in pro audio do we mix two audio directions on a connection forcing us to imperfectly pry the two apart in our interface gear Where else do we use blasts of 100 AC Volts for signaling This becomes even stranger when you consider that the core of the telephone network is also digital with sophisticated internal signaling systems and independent audio paths Surely we can do better than this ancient bell banging and analog audio mash up stuff in our contemporary telephone interfaces That was the idea of ISDN and it was a good one as far as it went The telephone network had begun to transition to digital in the 1960s and by the 80s the conver sion of the switching and internal transmission was nearly 100 complete in many countries The idea of domestic data communication was just getting underway Remember bulletin boards and 1200 Baud modems The inventors of ISDN rea
148. stems authorized dealers are not authorized to assume for Telos Systems any addi tional obligations or liabilities in connection with the dealers sale of the Products
149. store Firmware 24 Section 2 With the individual Show x configuration pages you tell the system how to assign Telco lines to VSets and other controllers You give each line a text name that appears on controllers You decide if the line is to be Selectable a number of lines switched into a few faders or Fixed one to one correspondence from lines to faders You specify if a line is to be affected by the Block All function Finally you can override the default SIP gateway on a per line basis The SIP Extension is like an extension number in a PBX system with the difference that it is generally not required to be a number It maps incoming calls to the line positions on the controllers Together with Server it makes the SIP address identifying the line The server is also used to form SIP addresses from numbers for outgoing calls The checkbox in Server column allows you to override the default server set in SIP configuration You can assign the same extension to multiple buttons Calls will ring in on the first available button providing a kind of hunt group The server address is ignored in this case Near the top of the page you see what studios are using this show VX in the News and Production rooms The flexibility of the VX extends to the newsroom and production studios It s easy to create a studio in the VX that can be dedicated to a workstation or special need Most newsrooms these days are simply PC workstat
150. t 105 dB referenced to 0 dBFS Digital Input to Analog Output 103 dB referenced to O dBFS 106 dB A weighted Digital Input to Digital Output 138 dB Total Harmonic Distortion Noise Analog Input to Analog Output lt 0 008 1 kHz 18 dBu input 18 dBu output 4 Digital Input to Digital Output lt 0 0003 1 kHz 20 dBFS 4 Digital Input to Analog Output lt 0 005 1 kHz 6 dBFS input 18 dBu output Crosstalk Isolation and Stereo Separation and CMRR Analog Line channel to channel isolation 90 dB isolation minimum 20 Hz to 20 kHz Analog Line Stereo separation 85 dB isolation minimum 20Hz to 20 kHz Analog Line Input CMRR gt 60 dB 20 Hz to 20 kHz Controllers VSet12 telephone VSet6 telephone VSet1 telephone VX Producer Windows application Axia Element studio console Axia IQ studio console Neosoft Neowinners Broadcast Bionics PhoneBox VX VX Engine IP Ethernet connections One 100BaseT gigabit Ethernet via RJ 45 LAN connection One 100BaseT gigabit Ethernet via RJ 45 WAN connection Processing Functions All processing is performed at 32 bit floating point resolution Send AGC limiter Send filter 64 Section 5 Gated Receive AGC Receive filter Receive dynamic EQ Ducker Sample rate converter Line Echo Canceller hybrid Acoustic Echo Canceller wideband Power Supply AC Input Auto sensing supply 9OVAC to 240VAC 50 Hz to 60 Hz IEC receptacle internal fuse Power consumption 100 Watt
151. t thing no one is perfect and it s possible that we have missed a line or two If you are experiencing problems let us know and we ll figure it out That said there are a couple of things that VX isn t likely to support in near future One of them using multicast for both SIP signalling and RTP This is an optional feature in the SIP standard not widely used and disabled in VX because it can cause conflicts with Livewire which is mul ticast too The second is telephone URI handling which might sound surprising for a phone system Don t worry it s fine Simply put it means that VX identifies each line with a single SIP address extension server entered in the configuration and expects that any mangling of numbers like adding or removing prefixes or changing between internal and external num bers is done by a gateway or PBX For VX to do that would require it to know a lot more about the phone network and believe us you don t want to configure that Axia Element Console as VX Controller The Element console makes a perfect partner to the VX With apologies to our colleagues in the Axia division who would prefer to put that the other way round the VX makes a nice partner to the Element They would also point out that other Axia consoles such as the IQ are capable and compatible VX partners Control and audio are tightly integrated for a talent leasing all in one place control of telephone operations Both sel
152. terisk Trunk or trunks May be SIP IAX2 ZAPTEL or Dahdi Hardware such as expansion cards gt gt 99 Vx Engine or Telephone sets Vx Engine or trunks to other PBX s Example Call 311 555 2368 called from a cell phone in Barstow California 760 256 8463 Cell carrier routes the call to the SIP provider over the Public Switched Telephone Network as the number translates to the SIP provider 311 555 2368 with Caller ID of 760 256 8463 is received The SIP provider sends the call to the Asterisk SIP trunk because 555 2368 is registered to the trunk belonging to the subscriber Caller ID and the 10 digit DID number are sent to the Asterisk SIP trunk belonging to the Subscriber 311 555 2368 is routed to extension 2368 on the Asterisk switch All info sent is passed onto the Vx The Asterisk switch receives the digits and attempts to match the DID number to an Incoming route If matched the call is sent further to either and extension trunk or special treatment in the system such as voice mail or a busy signal if there are no free trunks or buttons at the far end Incoming Call flow to VX via Asterisk Open Source PBX DID numbers from SIP Provider 311 555 2368 311 555 9467 311 555 1067 DID Numbers from ISDN PRI 311 555 7200 7299 311 555 1300 1399 Numbers from POTS Gateway 311 950 1022 311 976 1234 311 853 1212 311 958 1114 gt
153. the Telco line standard DTMF Dual Tone Multifrequency tones are played on air but they are scrambled so that listeners are not able to easily detect the number being dialled The correct DTMF tones are played to the VSet handset and loudspeaker Ringtones Choosing the Ringtones menu lets you choose different ringtones for each incom ing line position on the VSet Ringtones with higher numbers take precedence For example If one line has ringtone 1 assigned to it and another line has ringtone 5 and both lines are ringing the Vset will play ringtone 5 SIP and DTMF The VX uses standard SIP procedure for dialing A SIP call setup message containing the number is sent to the IP network The gateway to analog POTS lines generates the DTMF Dual Tone MultiFrequency audio signal that the PSTN uses to direct the call A gateway to ISDN lines translates the SIP message to the equivalent ISDN call setup message The VX does not generate DTMF audio Likewise the gateway translates call progress signals from analog tones or ISDN messages to SIP The default is a simple click There is sometimes the need to send DTMF to the PSTN after a call is connected such as for automated attendant systems In this case the VX sends a special SIP message that tells the gateway to generate the corresponding audio DTMF tones according to the standard specified in the IETF Internet Engineering task force RFC2833 INSTALLATION AND CONFIGURATION 29
154. to both the business offices and studios Sharing lines among studios Check and check A pure digital connection to the Telco keeps audio clean and maintains isolated send and recieve signal paths In studios with an AolP audio infrastructure an IP based telephone system is icing on the cake Dozens of inputs and outputs are connected with a single RJ 45 And you profit in other ways A single on air phone system server can supply all the studios in your facility with rich telephone capability A common wiring and Ethernet switch infrastructure serves both your studio audio and telecom needs WELCOME 3 On air VSet controllers have rich capability owing to their connection over IP Screening software running on PCs connect over the same network and can include integrated softphones thus smoothing operations and saving the money that would otherwise have to be spent on hardware phones Mixing console control surfaces can incorporate phone system controllers that need no additional connection their signaling just rides on the network connection already there Rich status information can be displayed either on the phone control module or the console s main screen Recording and playback of DJ telephone conversations are simplified PC based editors send and receive audio directly over the network using their native Ethernet connections Receiving Telco service via IP may be much less expensive than the traditiona
155. to the NX series and your legacy products Advanced audio processing and the fact that you never have to overcome Telco loop losses or extra two to four wire conversions means that the voice quality is as good as it can be Calls from mobile phone calls will be less than perfect at times but VX extracts the best possible from them NOTES RESOURCES ADDITIONALINFORMATION 95 How are callers on the VoIP trunks going to sound What about echo Won t cell phones sound even worse than usual The VX has enough processing horsepower to deal with even extreme echo situations and four wire Telco delivery means that the only external echo path is from the caller s line when it s analog and the caller s handset The VX uses Telos latest hybrid technology 5th genera tion enhanced with the latest state of the art acoustic echo cancellation Even when using open speakers and changing levels during a call the new algorithm makes feedback nearly impossible Steve Church once told me that IP cell phones can sound better than usual 3 kHz circuit switched phone technology something about G 722 dot something Is this true Right Current Cisco VoIP phones for example support the g 722 codec The VX supports this as well However Steve was probably referring to AMR Wide Band also known as G 722 2 sometimes called HD Audio It s 7khz and doesn t sound at all like phone audio in fact it sounds better than regu
156. tocol This means it is compatible with a wide variety of VoIP services gateways and PBXs Gate ways are used to interface PSTN lines to SIP This is how analog POTS and ISDN lines connect to the VX Gateways are off the shelf for POTS T1 E1 and both BRI and PRI ISDN These can be rack mount units that support large numbers of connections or low cost desktop boxes that interface a few POTS lines 8 Section 2 Grandstream GXW4004 4 port FXO gateway The gateway could also be a full up IP PBX such as from Cisco Digium Asterisk and many others In this case compatible VoIP phones can be used for general office locations SIP lets you move calls between the office and studio systems with no audio degradation A PBX such as Asterisk could add additional capability such as automated attendant functionality With appropriate cards Asterisk makes an excellent gateway to T1 and ISDN PRI Telco lines Using VolP to Connect to the Telco Network Recognizing the growth in market share of VoIP PBXs Telcos are beginning to offer SIP Trunk ing service which delivers phone network connectivity directly over a controlled IP link With this service you wouldn t need a gateway While it remains a niche in early 2011 SIP trunking is growing in support from both PBX vendors and carriers Over time this will almost certainly reduce the use of the older POTS and T1 trunking Eventually it may replace it completely See the Resources s
157. ts etc Web browsing email etc This is the power of IP realized To accomplish this level of functionality with older technolo gies would have been impractical Console Controllers Integrating phone line selectors into the studio mixing console is an operator pleasing feature that is easy to accomplish in a networked studio As this is written in March 2011 the following consoles have VX compatible call control modules available Axia Element Console Axia IQ Console 3rd Party Producer Applications The VX uses an open protocol for control which permits non Telos software applica tions to be used in place of or to augment the Telos VX Producer application Broadcast Bionics and NeoSoft offer such applications The VX system components are linked via a standard Ethernet switch In Livewire enabled studios this will already be present and no additional switch would be needed Facilities that already have a VolP PBX would also probably have a suitable switch in place The Acoustic Echo Canceller A new acoustic echo canceller algorithm solves the longstanding problem of feedback and echo when a loudspeaker to microphone acoustic path is required in the studio such as when DJs prefer to record calls without using headphones or when guests need to hear calls without headphones The AEC in the Telos VX is a remarkable new development lts performance is shockingly impressive permitting very high loudspeaker volume wit
158. twork as your Asterisk box Log into your VX engine with the web browser and select SIP Enter the IP address of the Asterisk in the Server field at the bottom of the page and click Add The configuration page for the server will open where we can change the server settings and add new lines Earlier we created extension 2222 using FreePBX Now we need to add a corresponding line configuration for VX to register it with the server Expires allows you to change how often VX will refresh it s registration Leave it empty and the VX will register as often as Asterisk says it must Auth User is typically your extension number and thus can be left empty as well Auth Password should be the same as your secret from the PBX in a Flash Extensions setup page N Configuration Main Asterisk SIP General Settings Studios Shows SIP Server 192 168 0 248 Call Audio Processing Name Asterisk ironies External IP Logging Information ich Studios ply Calls Devices Lines Stream Statistics S Extension Register Expires Auth User Auth Password Shows Studios Script Manager Script Information 2222 M Lf LY eee New Show1 Delete Misc 5600 New Show1 Delete lOManager Info 5601 gt System Status Logs Apply Backup Restore Firmware New Show 1 Delete Add New Though the VX supports SIP trunks as well as SIP extensions lines extensions are p
159. urations save your Show and Source profiles and load one of your newly edited Show Profiles using your Element console s Show Profiles command You can use the Element Control Center Phone Channels page to confirm your VX connection as shown below Push some buttons on the Element marvel at how cool it is and impress your staff Use it to phone up your Significant Other to congratulate her him on her his perspicacity and fine taste in mate selection 60 Section5 EN C fi 192 168 0 140 mod_phone giement Element Control Center System Status Setup Customize Log 1 3s Log History Log Setup Element Control Center Phone channel configuration ID Module Server Address Login Password Screenshot i E Module Manager Active phone connections Modules CAN bus information ID Type State Host Brightness control 1 VX Comnected 192 168 0 9 What s next The first stage of the application of any new technology is to replicate the function of what came before but once the new platform is in place creative people invent new and unexpect ed ways to use it IP is a potent enabler that has already showed us plenty of surprising things it s inevitable that more is on the way The Telos VX represents an exciting change in direction for studio telephone operations The open IP nature of the system along with its rich user interfaces and a powerful platform offers ongoing oppo
160. urs after a port though this seems to be getting better Don t run crazy win a house or car contests on these providers though they re fine for request lines and even talk show call in numbers If you plan to do heavy contesting with big prizes it s not nice and potentially dangerous to bring your carriers switching office down Introduction to SIP s insides for the Curious The following is a peek inside of SIP There is no reason you need to know any of this to use the VX It s an excerpt from Steve and Skip s book Audio over IP SIP is fast rising to be the big daddy buzz acronym among telecom technology acolytes SIP is how calls are set up over IP connections so it is actually pretty important Together with help ers like Proxy Servers and User Agents SIP permits all the familiar telephone like operations dialing a number causing a phone to ring hearing ringback tones or a busy signal It also enables next generation capabilities such as finding people and directing calls to them wher ever their location Instant Messaging and relaying so called presence near the phone or not do not disturb etc information SIP began rather humbly as a simple message protocol for setting up connections But the term has grown to be an umbrella for the family of protocols and tools that have been developed by the IETF to enable VoIP telephony and related services By the mid 1990s audio and video were becoming routine on th
161. ve canceller must be part of the system when loudspeaker monitoring is expected T received signal from phone echo Reference in 4 yr echo free mic Mic in ALC Out ES to phone local speech Fortunately technology has come to the rescue Modern Acoustic Echo Canceling is the answer The audio at the studio microphone consists of the host s voice combined with the unwanted telephone audio that is delivered to the room via the loudspeaker An AEC removes the loudspeaker audio leaving only the host It does so by synthesizing the transfer function of the acoustic path The reference loudspeaker signal is passed through this function and subtracted from the microphone signal thus canceling the echoed telephone part of what the microphone picks up AECs have been used in high end audio and video conferencing systems for many years High end broadcast hybrids and on air systems such as the Telos Delta 2x12 and Nx12 have included a limited form of AEC But only recently has AEC technology advanced to the stage where it is truly effective Thankfully it comes just when the added delay of mobile and VoIP connections make it near essential The good news has been a result of both break throughs in the design of AEC algorithms and the ever increasing power and lower cost of the processor chips that are needed to implement them 50 Section 5 These latest generation cancellers are a miracle You can have ear splitting volume
162. ved That means that you can assign a fader to each call if you want to do so allowing independent control of each call s volume Normally when a call is taken it goes to fader 1 but you can assign it to any of six console faders The number of possible faders depends upon your specific installation and configura tion When a call is on air or before it is taken the round fader assign buttons to the right of the LCD can be used to move the call to the desired fader The number in the yellow rectangle in the line info field shows you which fader is or will be used Fader assign is configurable to be enabled or disabled to keep the LCD uncluttered for people who will not use the feature 46 Section4 Next Button and Function Take Next The Talent next priority is 1 longest waiting ready hold 2 longest waiting hold 3 longest ringing in The producer priority is 1 longest ringing in The producer can manually override these and assign priority as desired Block All Pressing this button will cause all inactive and ringing lines to be dropped and blocked from accepting any calls Calls on air on the handset on hold and the fixed lines will not be af fected The usual application for this function is to let you prevent early callers from getting in on contests until after you ve made the announcement Pressing Block All again will release the lines and allow incoming calls The specific lines blocked are dete
163. w field where you can enter a LW channel number to which this Fader LW channel will listen Normally it will listen to the same LW number as it is sending to With native Livewire consoles such as the Axia Element this is what you need But if you want to connect the VX to another console via an interface Node the Node can generate only a From source not a backfeed For this case choose Advanced receive and enter the LW channel number of one of Node s sources 22 Section 2 received signal from phone echo A Reference in ak Mic in ut echo free mic L y gt e AEC Gut a to phone local speech Each studio has an acoustic echo canceller available The two inputs and the output are assigned to Livewire channels on this page The output of the AEC is labeled Backfeed because the output of the AEC is what you usually would feed back to phones For configuration purposes you should think of the AEC as a separate block outside of other VX functions There is no internal connection from to the AEC and other VX signal paths The Acoustic Echo Canceller helps with the problems that occur when you need to have loud speaker monitoring of calls in the same room as the microphone feeding the phone Without a canceller the received caller audio would be returned to the caller as an annoying echo In the old days this acoustic coupling would more probably cause feedback howl than echo but today s mobile phone and VoIP delays
164. wer lines on any other hybrid operators will use the module s SET key to choose which hybrid to use Selecting Auto uses the configuration of the VX Show to dynamically set the number of lines used 58 Section 5 Click Save when you re done Note that after editing and saving a Show Profile you must reload it from your Element console in order for the saved changes to take effect Create the needed Source Profiles Navigate to Element s Source Profiles page Click New Source Profile and using the Primary Source drop down box select the audio channel that you configured in VX Engine under Studio configuration In the list you should see a Livewire channel number and short description of the Hybrid give it an Element source name such as VX Hybrid1 and select Phone for Source Type Primary source 2410 lt S1 Sel1 VX24 gt Ely Hyb1 y Show sourcename Allow application to override y Show both 1446 SRC 8 OlegAnalog 2401 VIP1 Vx24 2410 1 Sel1 VX24 Channel 2 2411 1 Sel2 VX24 Channel 3 2422 S2 Selectable 3 VX24 Channel 4 2430 S3 Selectable 1 VX24 Channel 5 2440 S4 Selectable 1 VX24 J Channel 6 2441 S4 Selectable 2 VX24 Channel 7 2450 S5 Selectable 1 VX24 Channel 8 METE CR Montor rots Studio Monitor input Y V mixer input Source type Fy Operator CR producer F y CR guest Studio guest gt External microphone Phone Computer player In the source profile option
165. whether on air or not If no call is ringing in the system assumes you want to dial out and opens a window to let you do that The Line Info Field Fader number Before a call is taken to air this shows which fader it will be assigned to when it put on air After a call is on air this shows which fader is assigned to the call The number changes when you move a call using the fader assign buttons Next indicator Shows which call will be taken when you press the Next button Line status icon Line name This is the name that was given to the line during show configuration Caller info Text you enter into this area will appear on any other VX Producer applications and VSets that are assigned to the same studio It automatically replaces the line name text Telephone number Shows incoming caller ID or outgoing dialed number The small arrow near the number points left for incoming and right for outgoing numbers It automatically replaces the line name text Time Shows time that a call has been waiting on hold or if on air how long it has been on air Drop Hold and Hold Ready Buttons J 2 Drop Hold Hold Ready Drop disconnects a call that is ringing active on the handset or on air If more than one line is in this category they will all be highlighted Press the line that you want to drop Or select the line you want to drop to highlight it and then press Drop Hold puts a call that is ringing active on the hand
166. xtensions There s plenty of bandwidth on a LAN so this works fine while staying with a standards based approach We like Asterisk as a VX adjunct It can add voice mail automated attendant blocking callers from caller ID off premise SIP extensions and more to a VX installation Asterisk is free Linux based PBX software that runs on a PC The VX and Asterisk PBX are an attractive combo we expect will become popular within the broadcast industry 94 Section 5 How can get reliable VoIP trunk lines What is involved Can you recommend a vendor Yes we can assist We have been working closely with the VX beta sites and other early adopt ers of VoIP so we have plenty of experience to share There are several types of VoIP dialtone providers You ll want to consider how the service will be delivered to you via the Internet like Vonage or via a dedicated IP circuit from the provider that includes a Service Level Agreement and guaranteed Quality of Service as offered by a number of vendors including most of the traditional Telcos For discussion of this and other matters check the Telos web site on a regular basis as we continue post material on this and related topics know that SIP is supported by the new IP codecs Will the VX be able to connect to my Zephyr IP in the field Other codecs As we hinted above Yes The VX supports g 722 7khz wideband audio and g 711 3 4khz phone quality What about SIP S
167. ypassing Asterisk s processing when compatible codecs are present at both ends This is generally a good idea but it can be troublesome because it forces the VX to change the IP address and other parameters for the line source While Reinvites are supported in VX since April 2011 we have discovered an issue with Asterisk which can cause audio problems in some cases If Asterisk is used as a PSTN gateway or connected to a telco it should be fine but if it is accessible from the public internet ie anyone with a softphone can call it it is better to disable Reinvites If one is not sure better to be on the safe side Until re invites are supported the Asterisk canreinvite parameter should be set to no in later versions of Asterisk those are two options directmedia and directrtpsetup In case anybody is wondering Asterisk doesn t re negotiate the codecs For example VX supports g 711 and g 722 Let s say someone is calling from a softphone supporting g 711 and Speex Asterisk supports all four and will advertise it this way transcoding if necessary However when it makes a direct connection it doesn t change the codecs to those actually supported by each party Thus VX will end up thinking that the other end supports g 722 and the softphone will think that VX supports Speex resulting in silence in both ways When Asterisk and VX are on the same subnet Asterisk s NAT Network Address Translation support s

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