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1. Feedback INPUT program without phone lines of the consoles in the different Systel IP studios input audio level activation and AGC reference level Queue OUTPUT Output queues audio from each Systel IP system to each of the phone faders in each studio Intercom INPUT OUTPUT Input and output to the different intercoms configured in the studios The audio levels indicated with are managed in parallel from the different user screens within the web control applications The screen representation of each line includes a tri color audio presence indicator It also has a VUmeter icon that when clicked in sends audio from that line to a precision i VUmeter located at the right side of the top of the screen 30 AEQ SYSTEL I P 8 Gain amp AGC z r Nominal Level 4 dBu 2 Lines VOIP da Systel IP 172 26 38 100 IP12 Nominal Level 4 gt dBu 2 2 3 a 3 2 INPUT Mila tee STEL Adjust AGC to Nominal level 4 dBu i i i OUTPUT MENO a Lines INPUT OUTPUT Feedback INPUT Intercom INPUT OUTPUT MAC cano aer O Gano GSM O cn o SHEN O cain 2 2 o PATERNO cain 3 ijo MMT O enn lt jb acc ls O Gano AGC Enabled AGC 4 5 BIER O cano gt ASIA C Gain o gt MATTE cano wen O Gano gt em OUTPUT Line vor 4 0 Gan o aen O Gain a Sg O cain MOTE O can o db acca O Gano LESSER O con gt AEB O Gain lo ji acc ls Gano le LESCH O
2. A CPU with good performance price ratio The best or most expensive is simply not required The determining factor is mainly the number of simultaneous calls to achieve A VGA monitor keyboard and mouse for the PBX PC although they will probably be only necessary during the installation process Configuration and management can be carried out by means of a web browser remotely Steps to follow Download the ISO image from the AsteriskNOW server http www asterisk org downloads asterisknow Record iton a CD or DVD Insert the CD in the PC where we want to install the PBX Start up the PC and follow the instructions appearing on screen Choose the installation with Asterisk and FreePBX The AsteriskNOW server also provides a detailed installation guide in the following URL httos wiki asterisk org wiki display AS T Installing AsteriskKNOW digium AsteriskNOW To install with Asterisk 11 and FreePBX type To install with Asterisk 11 only type Once the installation process is finished please extract the CD from the drive and reboot the PC Once it has rebooted assign it a static IP address Connect the PBX PC to the LAN network and if possible provide it with Internet access If everything has gone well you will be able to access the PBX web panel by typing its IP address in the URL box of a web browser running on any other PC within the same network User name is admin and password is also admin 71 A
3. RTP Local Ports Range of ports that the system can use for audio traffic using RTP protocol A total of 24 ports must be configured for Systel IP 12 and 8 for Systel IP 4 IMPORTANT NOTE In case that NAT is used the network administrator in the Systel side must configure a PORT FORWARDING from the router to the Systel IP private address indicating the same port range that the Systel RTP ports range Call Transfer DTMF Tones This option must be configured when the SIP server implements call transfer between extensions typical in SIP PBX The DIMF sequence to be sent to the SIP server as a prefix to indicate the call transfer from one line of the system to other extension is defined here This sequence is 2 in Asterisk PBX Buffer Size This is the max size in milliseconds of the audio receive buffer This value directly influences the audio delay If the network presents a large jitter value the buffer size will need to be larger in order to avoid packet loss and hence audio cutouts The better the network connection is and so the lower the jitter the smaller the buffer size can be and as a result the delay in the reception of the voice will be shorter Packet Size Determines the quantity of audio in milliseconds that is transmitter in each of the IP packets The larger this size is the longer audio cutouts will be in case a packet is lost but the lower the overhead in consumed bandwidth is In the end it is a trade off
4. directions No extension X 8 S SA No extension y gt E NO_EXTENSION No extension y gt E NO_EXTENSION No extension y gt E NO_EXTENSION No extension y gt E NO_EXTENSION No extension y gt E NO_EXTENSION No extension y gt E NO_EXTENSION No extension y gt NO_EXTENSION AEQ SYSTEL IP IMPORTANT NOTICE this tool shouldn t be used while the web server is running as interference between the user s and the web server s operations may happen Its use is not recommended while operating with system Do you want to continue All the telephone lines in the system are shown in the window either 12 or 4 Each line is represented in a row including 8 fields gt 2 CONNECTED 6002 9722 Hangup The first field that has no header will show the pencil symbol when a line has been configured from this same window This symbol will disappear when the line is left without configuration again Te The second field Line corresponds to the equipments IP line Tee number from 1 to 4 for SYSTEL IP 4 and from 1 to 12 for SYSTEL IP 12 The label will have a blue background if the line has nothing assigned neither an extension nor an IP handset when the window is accessed On the other hand it will have red background when something is assigned to the line at that moment Line VOIP 2 SIP Extension The third field SIP Extension ESSE noextension w
5. Output Click on Back to leave the Change Level Handset screen o es 1 e Change Program amp Studio Permite salir de la warning x aplicacion y reiniciarla en otro escenario de A ad ie trabajo Se solicita confirmaci n Si No e Close sesi n Permite salir de la aplicaci n Tambi n se solicita confirmaci n e Back Permite volver a la pantalla de trabajo 5 4 3 General Chat 08 01 03 Control Need more information about next event 08 01 32 Coordinator s in the program folder 08 01 44 Control Thanks 07 49 34 Presenter Im ready for the next caller 07 51 23 Presenter Please finalise the call with John i 07 58 12 Director Program ends in 10 mins By clicking on the window located just below it messages 02 01 03 Control Need more information about next H event can be typed In 08 01 32 Coordinator ts in the program folder 08 01 44 Control Thanks If the position is not equipped with a keyboard an on 07 49 34 Presenter m ready for the next caller 07 51 23 Presenter Please finalise the call with John screen virtual keyboard will appear so the user can type 07 58 12 Director Program ends in 10 mins with it Everything that is written here will become 08 01 03 Control Need more information about next ewent available to all program users 08 01 32 Coordinator It s in the program folder 08 01 44 Control Thanks There are a
6. Pin 12 GND Pin 5 AES 2 IN Pin 13 AES 2 IN Pin 6 GND Pin 14 GND Pin 7 AES 2 OUT Pin 15 AES 2 OUT Pin 8 GND Identification of the DB15 connector DIGITAL I O 3 4 Pin 1 AES3 IN Pin 9 AES IN Pin 2 GND Pin 10 GND Pin 3 AES3 OUT Pin 11 AES3 OUT Pin 4 GND Pin 12 GND Pin 5 AES 4 IN Pin 13 AES 4 IN Pin 6 GND Pin 14 GND Pin 7 AES 4 OUT Pin 15 AES 4 OUT Pin 8 GND Highlights Each of the four digital audio input and output includes two different audio channels according to AES 3 or S PDIF standards AES 1 IN input provides synchronization to SYSTEL IP 12 with the source connected to it no matter if it provides an AES3 or AES11 signal Each output can provide sync to other pieces of equipment that are able to extract it from the AES 3 audio signal 2 2 2 4 1 Digital input programming jumpers IMPORTANT NOTE Access and configuration to the programming jumpers requires previous experience with installation and configuration of electronic cards or computers Don t open the unit if you lack this experience at the risk of produce permanent damages or suffer electric shocks 14 AEQ SYSTEL IP SA Digital inputs and outputs AES IN and AES OUT are programmed from factory as AES EBU If connection to an S PDIF system is required the unit must be open to change the corresponding jumpers AES or S PDIF digital mode selection jumpers Opening t
7. PsK Y Network Key CLAVEDELAWIFI 8 63 ASCII or 64 HEX Save Settings Don t Save Settings WIRELESS The Enabled status in WI FI PROTECTED SETUP must be removed because it is relatively easy to gain unauthorized access A name must be given to the WiFi network in the SSID field Select Enable WPA WPA2 Wireless Security enhanced in the Security Mode field Enter a proper WiFi password in the Network Key field To finish just click on Save Settings to store the changes made Now click on LAN Setup where a screen like this will be displayed 88 AEQ SYSTEL IP AEQ Easy Setup Internet Connection Wireless Connection LAN Setup Time and Date Parental Control Rules Logout Helpful Hints E S These are the settings of Use this section to configure the internal network settings of your router and also to configure the LAN interface for the the built in DHCP Server to assian IP addresses to the computers on your network The IP device Address that is configured here is the IP Address that you use to access the Web based t interface If you change the IP Address h d to adjust your PC E EES SE management interface If you change the ress here you may need to adjust your PC s your network that should network settings to access the network again always have fixed IP addresses add a DHCP Reservation for each Please note that this section is optional and
8. corresponds to the SIP extension ee configured for the line according to the explanation to be provided 6001 in chapter 4 2 3 in this manual Alternatively a SIP Handset can be sas selected for the line instead of a SIP extension IP HANDSET 1 IP HANDSET 2 IP HANDSET 3 IP HANDSET 4 m The fourth field consists on a button that allows the user to apply the configuration selected in the former field extension or handset to the line When this icon is colored a a new configuration is pending After applying the configuration the icon will turn transparent e The fifth field is the lock button giving access to the mm extension authentication data configuration screen Basically o the user and password for the selected extension can be rr modified in this screen just in case they were not correctly configured Auth Password secret6000 The sixth field Line Status represents the status of the line Line Status during the test process 32 AEQ SYSTEL IP The main possible states are e Extension connected to a phone number with a Line Status coding algorithm just as the first example of the attached image CONNECTED 0639189887 9722 e OK ready to test as in the 2 3 and Al mm example The extension associated to the line is o emm highlighted in green This green label appears when the user selects an extension different to P the configured one but the change has not been ag exr
9. A 1 Benefits provided to Systel IP by Asterisk Systel IP based telephone systems can be enhanced by adding a voice over IP PBX like Asterisk New functions that an Asterisk PBX can provide include e Call routing between different Systel IP units One simply needs to associate an extension in the PBX to two extensions each in one Systel IP unit so that the PBX can transfer the calls from one extension to another e Integration of the broadcast telephone system with the corporative phones this way it will be possible to transfer calls between Systel IP units and corporative extensions A VoIP Asterisk compatible phone is required at each workplace e Pre recorded message playback such as welcome messages for calls coming from outside the station e Interactive menu presentation so the callers themselves can route their calls to the final destination within the station dial 1 to talk to a certain program etc e Voice messaging where as many voice mails as necessary can be defined Both voice mails associated to the corporative extensions recording messages when the user is not available at its workplace for example and voice mails not associated no any extension such as automatic recorders to store recordings for example can be defined PSTN i p i S Syatel iP server Systel IP server A and configeratie and Cafi anon 3 Sa voip xx b pro er VS gt Y 7 sudo 7 f e Es E AY Mao player
10. When the call is established the icon includes a small red upwards pointing arrow indicating that the call is outgoing or a downwards pointing green arrow when the call is incoming Making a call If a call is not present or being received the icon will be blue when clicking on it a Call Menu window will be accessed in order to enter a phone number or URI A common area at the top allows dialing numbers or URI that are not in the phone books By composing the number or URI with the Phone Number window s keyboard or by clicking on the keyboard icon at the left so a numeric keyboard appears that allows the user to compose the phone numbers After that confirmation is provided by clicking on the green phone button There are three different possible windows that allow to call phone numbers URIs or contacts that are stored in the phone books which can be changed by clicking on the lower tabs Call Menu Line 06 Close Call Menu Line 04 Close Call Menu Line 01 Close Phone Number Phone Number Phone Number si 6003 wild BES SE vi ew Si vi Name Number Time Name Id Number Name VO LOISs 6001 00 00 JONATHAN PAYO S w 1 6006 Y LONDON JERRY 91131462134 00 00 MUNICH ANDREAS w 2 6005 Li MARIBEL 916335577 00 10 Michael Boy 3 PETER Ka MUNICH ANDR 87654321987 00 15 MARIBEL E w 4 6008 Y Michael 123456789 X 00 35 Mr Smith 5 JONATHAN PAYO A E Phone Book Prog Ph Redial Phone Book Prog Ph
11. cannot be activated and these buttons will not be shown In this example IP configuration data for the discovered Systel IP 4 must prevail in order to keep connectivity while we will accept the values in the database for the Rtp TOS packet prioritization as it is more adequate When all is synchronized all the values in red will change to black e Systel IP Systel IP 172 26 38 120 IP 4 SYSTEL IP4 SMALL STUDIO Network VOIP Change Description Setup 8B 0D systel change Description BOD Systel_ a IP 172 26 38 120 172 26 38 120 CodecList g722 pcma pcmu g729 g726 g722 pcma pcmu g729 g726 Mask 255 255 0 0 255 255 0 0 Buffer Size 20 20 Gateway 172 26 1 1 172 26 1 1 Packet Time 20 20 Dns 0 0 0 0 0 0 0 0 E Rtp TOS 46 46 WAN Active False False Proxy Host WAN IP 0 0 0 0 0 0 0 0 Proxy Port 5060 5060 WAN Mask 255 255 255 0 255 255 255 0 Registrar Host WAN Gateway 0 0 0 0 0 0 0 0 Registrar Port 5060 5060 WAN Dns 0 0 0 0 0 0 0 0 Registry Expiration 3600 3600 Internal Auto Manual Auto Auto SIP Local Port 5060 5060 Internal IP 10 38 120 0 10 38 120 0 First RTP local Port 5004 5004 Internal Mask 255 255 255 0 255 255 255 0 Last RTP local Port 5030 5030 External Sync AES 11 LOCAL LOCAL Nat Mode 1 1 I Public Address Keepalive 0 0 29 AEQ SYSTEL IP 4 2 1 2 Updating the Systel IP unit firmware We can connect to the different registered Systel IP units from the main Systel IP JA Units screen in ord
12. showing all the information dimmed as it cannot be modified In order to be able to modify and individually configure a Systel IP unit we must uncheck that box this will enable the modification of every configuration parameter which we will be able to modify to the desired values After the desired parameters have been modified you can confirm those changes by means of the button or cancel them by pressing the O button in that case the Common Parameters SIP checkbox is activated again and all the parameters will return to the default values SIP Extensions The top right area of the initial window you can access by pressing the Close Customize Units button allows for the configuration of the system SIP These extensions will be assigned and released in the different Systel IP system as the different programs start and end At any given moment a system will have the configured extensions for the programs currently running in the different studios the system provides service to An extension can be available for a system during the morning and for another one in the afternoon We can create as many SIP extensions as required in this section but have in mind that during operation up to 12 lines can be assigned simultaneously to one or more programs in each SYSTEL IP 12 unit or up to 4 lines in SYSTEL IP 4 Also have in mind that a SIP extension may have one or more assigned lines or simultaneous calls up to 12 or 4
13. 1 tO 4 oocccccccconcnccccnoccnnncconanonnononannnnnononnconnononanennnoss 13 2 2 2 4 Digital inputs and outpute n eoaannnnnnennnneonennnensrnrnnnnnrnnensrnreressnnrnernnnne 14 2 2 2 4 1 Digital input programming jJUMPEEUS occcccoccncccnccnnccnncnncnnanononnncnnonancnnnnnos 14 2 2 2 5 Analog inputs and OUTPUTS occccccccnnccnncccnnnnconnccnnnncononononnnnnonnancnnnonnnanennnnss 16 2 2 LO OWL SUDO E 17 2 2 2 7 Notes about SYSTEL IP 12 audio wiring ccccoooocccnncccnnononnccnnnncnnnnnanennns 17 3 9Y gt TEM INSTALLATION comunica dci tilda 18 Dele COMO Cl NC esti lisa 18 3 1 1 Configuration PC and control Web server 18 3 1 1 1 Installation of the setup and control web server applications 18 912 SONO SIMA Sara 19 3 1 2 1 Installation and initial configuration of the control client in Windows 19 3 1 2 2 Installation and initial configuration of the control client in OS 19 3 1 2 3 Installation of the handset or SYSTEL IP HS micro telephone 20 3 1 2 4 Installation of the SYSTEL IP ST supports cocccoooooonccnnnccnnnnnoccnnnnnnnnnnananons 21 3 1 2 5 Installing IP phones as an alternative to the analog SYSTEL IP HS W lafe EE 22 e Kei E 23 4 1 Preparing a computer for setup and control of the system Starting the application 23 4 2 SYSTEL IP SETUP Description of the screens in the Setup application 24 4 21 OY SUC IP UNIS
14. Also a 2 queue can be created using Analog 2 System used in a single studio Analog console up to three queues 1 handset Studio Name Name Mode mput Name Output Name Pos STUDIO SMALL 1 SYSTEL IP4 SMALL STUDIO ANALOG 1 QUEUE ANALOG 1 3 ANALOG 2 QUEUE ANALOG 2 2 ANALOG 3 QUEUE ANALOG3 1 ANALOG 3 FEEDBACK ANALOG 3 HANDSET 1 INTERCOM HANDSET 1 HANDSET 1 If the studio console is analog and we use only one handset up to three queues can be created Note that this is not practical as the system capacity is up to 4 simultaneous calls 41 AEQ SYSTEL IP SA 4 2 3 VOIP VOIP The VOIP Voice over IP menu allows the user to configure the parameters associated to the IP telephony system These parameters will be directly applied on the Systel IP units By default the application will apply the same VolP parameters to all the systems fj Systel IP Setup es a Eo Systel IP Units T Studio Wiring Q Security Level Ei Common SIP Parameters VOIP SIP Extensions E E a 6001 a 6000 a SIP Server Host SIP Server Port 5060 Register Yes 0 No Registry expiration 3600 Seconds Advanced settings SIP Handsets m SIP Local Port 5060 E RTP Ports 5004 y direa Dm aea Systels IfMandset UserName Host Port Auth Id auth Password Line SYSTEL IP 1 Buffer Size 20 Packet Size 20 QoS lt No data to display gt NAT Trave
15. Applications Connectivity v Reports e Settings Logout admin Add Trunk Add SIP Trunk te Channel 90 dahdi AEQ sip E PSTN sip General Settings Trunk Name Outbound CallerlD CID Options Allow Any CID v Maximum Channels Asterisk Trunk Dial Options Tt Ll Override Continue if Busy C Check to always try next trunk Disable Trunk _ Disable Dialed Number Manipulation Rules prepend prefix match pattern om Add More Dial Pattern Fields Clear all Fields Dial Rules Wizards pick one v Outbound Dial Prefix Outgoing Settings Trunk Name PEER Details In order to integrate the connection to the provider within the PBX a SIP Trunk must be added This option can be found in the Connectivity menu under the Trunks section The most basic parameters to configure are 74 AEQ SYSTEL IP Trunk Name Netelip Outbound CallerlD lt number that will appear in the outgoing calls gt PEER Details type friend username lt username gt context from trunk host sip netelip com canreinvite no secret lt password gt nat yes fromdomain sip netelip com insecure very fromuser Register String lt username gt lt password gt sip netelip com lt username gt It is possible to check whether the PBX has connectivity with the provider s Server and whether it has been correctly registered by having a look at the system status option FreeePBX System Status within the Reports menu The status of the defined T
16. GPO 1 4 Pin 8 GPO4 Highlights please note that there is a common ground GND for inputs 1 to 4 and another one for outputs 1 to 4 There is also a 5V reference voltage at pin 12 to ease the wiring 12 AEQ SYSTEL IP Identification of the DB15 connector GPIO 5 8 Pin 1 GPI5 Pin 9 GND_ GPI 5 8 Pin 2 GPI6 Pin 10 GND_ GPI 5 8 Pin 3 GPI7 Pin 11 GND_GPI 5 8 Pin 4 GPI8 Pin 12 5V GPIO 5 8 Pin 5 GPO5 Pin 13 GND_GPO 5 8 Pin 6 GPO6 Pin 14 GND_GPO 5 8 Pin 7 GPO7 Pin 15 GND_ GPO 5 8 Pin 8 GPO8 Highlights please note that there is a common ground GND for inputs 5 to 8 and another one for outputs 5 to 8 There is also a 5V reference voltage at pin 12 to ease the wiring Identification of the DB15 connector GPIO 9 12 Pin 1 GPI9 Pin 9 GND_GPI 9 12 Pin 2 GPI10 Pin 10 GND_GPI 9 12 Pin 3 GPI11 Pin 11 GND_GPI 9 12 Pin 4 GPI12 Pin 12 5V GPIO 9 12 Pin 5 GPO9 Pin 13 GND_ GPO 9 12 Pin 6 GPO10 Pin 14 GND_ GPO 9 12 Pin 7 GPO11 Pin 15 GND_GPO 9 12 Pin 8 GPO12 Highlights please note that there is a common ground GND for inputs 9 to 12 and another one for outputs 9 to 12 There is also a 5V reference voltage at pin 12 to ease the wiring 2 2 2 3 Handset connectors 1 to 4 c SYSTEL IP 12 provides 4 8 pin RJ45 connectors to connect remote powered headsets The only model that can be connected to them is the SYSTEL IP HS handset If a wired or wireless
17. ROGE ADMINISTRATOR TONI TONI ADMINISTRATOR Y amp ADMIN Version 1 0 1 20 20 01 2015 14 03 22 In order to assign the different phone extensions each program will be able to use during its broadcasting we will use the drop down menus located in the Extension column in Incoming Calls area where all extensions present in the system are listed We will define the number of audio lines simultaneous calls required in the program for each selected extension The maximum simultaneous number of lines for a same call is 12 if our equipment is Systel IP 12 or 4 in case it is Systel IP 4 50 AEQ SYSTEL IP SA LM 4 2 7 PhoneBook ree This menu option allows the user to manage the phone contacts and define the calls schedule E BH Systel IP Setup Phonebook a Systel IP Units Programs Phone Books az Ee General Phone Numbers El ze aa EO 0 H Merge Phonebook Show Programmed Phone Numbers View All y Add Delete Sr Pr p Studio Wiring _ PROGRAM_1 BARCELONA 123456789 PROGRAM_2 la ua 3th PROGRAM EDU 34916864492 vom AFTERNOON A MUSIC IN THE NIGHT GUS 1915334455 e JOHN 6001 ea JONATHAN PAYO 6003 e p LONDON JERRY 91131462134 Users MARIBEL 916335577 o ames Nmber Blacks MUNICH ANDREAS 87654321987 Michael 123456789 Michael 123456789 Bes PARIS RTF PAUL 33101202303 Programs PETER 6002 PETER 6002 JONAT
18. SIP Protocol Session Initiation Protocol SIP is a protocol developed by the IETF MMUSIC Working Group with the aim to become the industry standard for initiation modification and ending of interactive user sessions where multimedia elements such as video video voice and instant messaging take place SIP is simply used to initiate and end voice and video calls Once the communication is established the information exchange is performed only with RTP One of the SIP goals was to provide a set of functions for procession calls and some other capabilities currently present in the public switched telephone network This way it implemented typical functions in a conventional phone such as dial a number make a phone ring when it is called listen to the tone or busy signal etc However implementation and naming are different in SIP SIP requires proxy and registration elements to provide a practical service Although two SIP terminals can communicate to each other without the need of additional SIP infrastructure by means of lt name gt lt ip_ address gt type URIs this is the reason why SIP is defined as a point to point protocol this approach is not practical for a public service due to the inherent problems to IP addressing where getting fixed public IPs is almost impossible and anyway quite costly SIP makes use of elements called proxy servers that help with the routing of request to the actual user location authenticate and prov
19. Security Levels fal a Systel IP Units ADMINISTRATOR sii la Dial amp Transfer y CONTROLLER Chat H Studio Wiring PRODUCER Y Handset gt TALENT OnAir ld Line Information Note Y Line Input Gain VOIP S Line Output Gain Custom Phone Name a Phonebook Save Phone Online Lines amp Studios Control Y 2 Y Menu SystelIP Setup SystelIP Units 3 Y Menu SystelIP Setup VOIP Users Y Menu SystelIP Setup Phonebook Y Menu SystelIP Setup Users Y Menu SystelIP Setup Security Level Programs v Menu SystelIP Setup Studio Wiring Menu SystelIP Setup Programs ML Phonebook d S S A ADMIN El ADMINISTRATOR Version 1 0 1 20 20 01 2015 14 03 22 The lower part of the screen shows the editing tools By hovering the mouse cursor over each button a label briefly indicating its function will appear F Insert record Allows you to insert a new user group Delete record Allows you to delete a user group 2 Edit record Allows you to modify a user group Allows you to save modifications to a user group x z Cancel edit Allows you to discard modifications made to a user group Allows you to refresh modified data Confirm Confirmation is required in order to delete an existing user ep Decterecora group 48 AEQ SYSTEL IP 2 Users 4 2 5 Users This screen gives access to creation and management of users An user is an individual person that takes p
20. an IP protocol Generated IP or VolP calls come through the Internet connection instead of the telephone network The main advantage of this kind of services is that it avoids the costs charged by telephony companies especially for international calls The same way it saves fees and maintenance of PSTN lines as it uses an existing data network VolP to VolP calls are usually free between any providers as opposed to VolP to PSTN calls that are generally not free of charge for the user An IP telephony operator provides the IP lines and is responsible for the interconnection of the incoming and outgoing IP calls to the IP or PSTN network Converting digital calls from ISDN RDSI to IP There is a large number of digital ISDN RDSI PBX in the market that don t allow making IP calls A VoIP ISDN gateway is required t route the incoming calls to the digital PBX as well as for taking the outgoing calls and sending them through Internet using the same gateway VoIP Gateway A VoIP Gateway is a network device that converts voice calls between an IP network and public switched telephone network or its digital PBX in real time Routing calls from a non IP PBX to SYSTEL VolP Gateways perform the conversion of calls from a conventional line or ISDN PBX to IP This way an existing installation can be used together with SYSTEL either connecting both of them to a service provider by means of a SIP Trunking or to a new Asterisk PBX Virtual
21. and output at the console to connect a generic intercom terminal with analog or digital audio input amp output or using IP phones as intercom terminals renouncing to an external IP line for each intercom created on an IP phone The drop down menu will show the available input output pairs We can select the desired one If the Intercom option is selected both inputs and outputs will be enabled The drop down menus will show the available inputs outputs for this resource filtering those that are already being used It makes no difference to select from the list of inputs or outputs as both of them are configured simultaneously from any of these lists Modify Add Resource INTERCOM e Input A Output IP HANDSET 4 IP HANDSET 4 Name ANALOG 2 Name Pos DIGITAL 1L DIGITALAL 1 DIGITAL 1 DIGITAL 1L PC IPHANDSET1 JIPHANDSET 1 PHONE 1 INTERCOM IPHANDSET2 IPHANDSET 2 PHONE 2 INTERCOM JIPHANDSET3 JIPHANDSET 3 HANDSET 1 INTERCOM HANDSET1 HANDSET1 HANDSET 2 INTERCOM HANDSETZ HANDSETZ 37 AEQ SYSTEL I P 4 2 2 1 SYSTEL IP 12 wiring configuration examples Let s assume that we have a 12 line unit called SYSTEL IP1 STUDIO_1 DIGITAL WIRED Studio Name STUDIO_1 DIGITAL WIRED SYSTEL IP1 STUDIO_2 DIGITAL SYSTEL IP1 STUDIO_3 ANALOG 6 QUEUES SYSTEL IP1 DIGITAL 1L DIGITAL 1R DIGITAL 1L HANDSET 1 HANDSET 2 FEEDBACK INTERCOM INTERCOM DIGITAL 1L HANDSET 1 H
22. answered in a certain time the call will automatically change its status to WAIT When the telephone is hanged up any line in communication with it will also be hanged up Dialing When in idle mode the user can dial from the IP phone itself and will call to using the first available line which will change to the HANDSET status Making a call from a specific line is also allowed by dialing its number followed by the separator and the number to call to If dialing is performed by means of the software it will force that the assigned IP phone rings first in order to warn the operator and when he answers the dialing will be performed from the line selected in the screen If the IP phone doesn t respond in a given time or it is not unhooked the call will still be made but once the interlocutor unhooks he will be parked in WAIT status Sending tones When an IP handset is assigned to a line and it is connected DIMF tones can be sent to the receiver by using the telephone keyboard 67 AEQ SYSTEL IP 6 TECHNICAL CHARACTERISTICS SYSTEL IP 4 Engine for 4 IP lines Inputs and outputs XLR type connectors SYSTEL IP HS handset RJ45 connector 2 analog balanced inputs 2 analog balanced outputs 1 selectable analog digital AES EBU AES3 or SPDIF input 1 selectable analog digital AES EBU AES3 or SPDIF output 1 IP port WAN for 4 VoIP lines 1 IP port LAN for control 1 DB15 connector for 4 optoc
23. arrasadas 25 4 2 1 1 Synchronize configurations cccooonnccnncccconncnncocnoncnnnononnnonnnononncannnonnnnennnoss 28 4 2 1 2 Updating the Systel IP Unit firmware 0000nnnnnn0000nnnnnnnnnnnnnnnnnnnnnnnnnnnnnnne 30 4 2 1 3 Adjust Gain WINKOW cccccocoocncccnnnccccnonnncnnnnonononannnnnnnnnnnnnnnnnnnnnnnnnnonannnnnns 30 4 2 1 4 VoIP Test WINKOW ccccccccccconnccnnnccccnonnconnnnnnnonononcnnnnnnnnnnannnnnnnonnnnnnanenons 31 ES ICO WIL E DEE 35 4 2 2 1 SYSTEL IP 12 wiring configuration examples occccccccccoocccconcccnonnnanncnnos 38 4 2 2 2 SYSTEL IP 4 wiring configuration examples oocccccccconncnncccconnnncnnnancnnnoos 40 A TE 42 4 2 4 Security LOVE leccion tino 48 A oo q PA 49 e O PTOI A o A 50 AL e MONG ee 51 4 2 8 Saving and restoring the database cccccooonncccnnncccconoonnnnnncnnnnnnancnnnnnnnnnnnannnnnnnnnns 54 2 AEQ SYSTEL IP 5 CONTROL TERMINAL BASED ON WEB BROWSER 0 cceceeeeeseeeeseeeeeeeeneeeeeeeeeseeeeennnes 55 5 1 WEB CONTROL CLIENT FUNCTIONAL DESCRIPTION cceesesseeescessseeeeseensneeeees 55 EA MOM te lee 55 5 3 Program operation WINDOW EE 56 DA Control screen right common area general controle 58 5 4 1 Call queue control WINKOW ccccococccccnnnccccoonncnnnnnonononanncnnnononononcnnnnnnonononanennnnnnnns 58 5 4 2 Reject Calls and Menu buttons aannnnnneonannnnnnnnennnnnnnnnnsnnnnnnnneossnnnnrnreessennnneeennne 58 EC e e e eo S
24. as the ADMIN user who has no password by default until one is defined Systel IP Setup main screen is accessed 8 BH Systel IP Setup Pai Systel IP Units 4 Studio Wiring VOIP Security Level O Systel IP Backup database Restore database ADMIN E ADMINISTRATOR Version 1 0 1 20 20 01 2015 14 05 42 You can find a menu at the left with all the configuration options divided in several groups Systel IP Units Y m0 VOIP a Security Level D Users Programs Phonebook d Backup database Restore database 1 Systel IP Units Basic information about the Systel IP units registered 2 Studio Wiring Assignment of resources to the configured studios 3 VoIP SIP parameters and SIP extensions for the Systel units registered 4 Security Level The user groups are created here and their level of use privileges is defined for the configuration and user applications 5 Users Management of user profiles to access Systel IP software 6 Programs List of programs with their lines and users 7 Phonebook Management of contacts for a Systel IP system 8 Main Screen By clicking on it main screen is returned from a configuration option You can find two keys at the lower part of this screen They are used to generate backup copies of the database and to restore them in case of need 24 AEQ SYSTEL IP Also a status bar
25. call is received the corresponding line status will change to INCOMING CALL and the user can choose whether to accept or reject it The call window allowing for the selection of the available intercom device and user will appear in the same way Test Line Accept Eep INCOMING CA Intercom The operations that can be carried out with IP handsets are connect disconnect and accept incoming call When the user selects connect the system will send the call to the IP handset according to the configuration present in the database The system hangs the call to disconnect 34 AEQ SYSTEL IP u H 4 2 2 Studio Wiring Studio Wiring Window that allows assigning resources to each studio configured in the Systel IP unit Shows a list with the currently existing studios and the unit each one is assigned to allowing for the creation of new studios up to a maximum of 4 per Systel IP 12 or two for Systel IP 4 the modification of existing ones or the deletion of one or more studios previously configured Every studio needs to have the following resources configured QUEUE These are outputs from the Systel IP system to console inputs or faders They can be analog or digital We can have up to 6 queues per studio with Systel IP 12 or up to 2 with Systel IP 4 FEEDBACK These are inputs to the Systel IP system coming from an auxiliary output without phones from the console These inputs can be analog or digital and we
26. changes click on If you want to delete one of the existing units use the Delete button In order to automatically add a Systel IP unit use the Auto Discover option by using this option the Systel IP Setup software we performs a search of all possible Systel IP systems within the A network The status of this process will be displayed in a new Auto Discover window that indicates the search time by means of a progress bar that will be filled in green while the process is being completed mg Ia New equipment detected IMPORTANT NOTE The IP addresses and subnet masks of the computer must be within the range of IP addresses of the units to be discovered The unit s default IP address is 172 26 36 250 in the case of Systel IP 4 and 172 26 35 250 in the case of Systel IP 12 2 AEQ SYSTEL IP fa BH Systel IP Setup Systel IP List Son o y Device Model c2090570277d SYSTEL IP1 Auto Discover Sdaef9bb066c SYSTEL IP2 New equipment detected SYSTEL IP4 SMALL STUDIO Wane Port TypeName Systel IP Units VOIP Test j i Firmware Sync configuration 172 26 38 120 4422 Systel IP 4 3916c5792b5d 172 26 35 40 4422 Systel IP 12 4317fdf5c15a Computer IP addresses 255 255 0 0 255 255 0 0 255 255 255 0 255 255 255 0 Existing Equipment lt No data to display gt Once the end is reached it will indicate that the search has been completed and
27. depending on the system model where the extension has been created Each SIP extension corresponds with an individual line or line group identifier number in a SYSTEL unit Depending on which SIP server the Systel extensions are registered in a name and password may be necessary 45 AEQ SYSTEL IP 7 BH Syste IP Setup VOIP Systel IP Units Common SIP Parameters SIP Extensions a al xtensions ZO 4 H Add Delete Studio Wiring pu o Codes oa names bed G722 Use Auth ID Yes 0 No G711A Auth Password G711U 8 sa G726 32 Number Auth Id Auth Password Description Security Level 6001 a R SIP Server Host 6000 te 2 SIP Server Port 5060 6008 EA AAA Zem Register Yes No Registry expiration 3600 Seconds Advanced settings IP Hand a SIP Local Port 5060 RTP Ports 5004 AL SE E rl egen vost port auth 1 ima SYSTEL P 1 O Buffer Size 20 Miliseconds EC SS Packet Size 20 Milliseconds 26 Customize Units Qos No data to display gt NAT Traver sal off auto static amp ADMIN El ADMINISTRATOR Version 1 0 1 20 20 01 2015 14 03 22 SIP Handsets The SIP handsets in the system are configured in the lower right area of the window These SIP handsets are standard SIP pones or any other audio device implementing SIP to be used as intercom devices so that the system operators can communicate with the callers usin
28. during the program broadcasting Each program will have its own programmed numbers list BH Systel IP Setup Phonebook f Systel IP Units Programs Phone Books E Programmed Phone Numbers E H Merge Phonebook Show Programmed Phone Numbers Modify a Se Programs Studio Wiring PROGRAM_1 E 1915334455 PROGRAM_2 E VIENA LUCAS 555 00 20 La 3th PROGRAM EZ LONDON JERRY 91131462134 00 25 MUSIC IN THE NIGHT PABLO IGLESIAS 91666666 14 10 E ROMA SOFIA ANDREOTTI 312345676543 14 15 S Ej LOISs 6001 14 55 Ei PETER 6002 14 56 e ANTOINE PERRIER 334455667788 15 08 i a E EDU 34916864492 15 55 Users E BARCELONA 3493333444555 18 30 mames JL Nmber Blacks Michael 123456789 JOHN 6001 Programs PETER 6002 JONATHAN PAYO 6003 LIM LOIS 6001 TONI 915444 SE GUS 1915334455 ROGE 916861300 q EDU 34916864492 MARIBEL 916335577 BARCELONA 3493333444555 PARIS RTF PAUL 33101202303 gt LONDON JERRY 91131462134 MUNICH ANDREAS 87654321987 PRS VIENA LUCAS 555 ROMA SOFIA ANDREOTTI 312345676543 Y amp ADMIN Version 1 0 1 20 20 01 2015 14 03 22 This window shows us which contacts are in the program phonebook as well as the name number and precise time the call will be launched at 92 AEQ SYSTEL IP Hour 7 1 14 56 15 55 We can change name number and call time of a programmed contact Mame Wii Hour 00 20 53 AEQ SYSTEL IP SA The software i
29. o External Sync AES 11 Users WAN for VOIP only INTERNAL NETWORK Active Auto Programs IP 0 Manual an Mak Se 2 as IP Subnet L Gateway 0 Mask 255 255 255 0 Phonebook DNS Computer IP addresses 255 255 0 0 172 31 3 222 255 255 0 0 O 192 168 0 11 255 255 255 0 192 168 7 97 255 255 255 0 amp ADMIN Fd Version 1 0 1 20 20 01 2015 14 05 42 The fields that must be filled in are Name Model the values of the LAN and WAN fields as well as the INTERNAL NETWORK selection and the possibility to use external AES external sync In order to modify an existing Systel IP unit use the Edit button this button shows up the same window as the Add button but all fields are already filled in with the actual information corresponding to the Systel IP unit you want to modify AEQ SYSTEL IP Name YSTEL IF 1 Model 2 IP 12 IP 4 LAN DIGITAL AUDIO IP i172 2 38 100 Input 1 No Output 1 No Input 2 No Output 2 No Mak 255 255 0 0 Input 3 No Output 3 No SC GK Input 4 No Output 4 No DNS 0 0 0 0 External Sync AES 11 WAN for VOIP only INTERNAL NETWORK IP 192 163 D 21 Manual Mak 20 250 doo 0 IP Subnet oc 0 060 0 9 Mask 255 255 255 0 DNS 0 0 0 0 All the parameters can be modified except Device ID Model DIGITAL AUDIO Input and Output types and the INTERNAL NETWORK mask In order to save changes click on e To cancel any
30. positions 1 amp 2 being output by the SYSTEL IP2 system through two analog wires Analog Outputs 5 and 6 to the input connectors corresponding to channels 1 amp 2 of the console that uses these inputs as mono A console auxiliary output is sent to Systel s Analog input 5 which will incorporate the program audio without telephones FEEDBACK through Analog input 5L Analog input and output 8 of the Systel are wired to provide Intercom to the producer and studio operator wiring the input to a console input and the output to a talkback output HANDSET 4 is wired to provide Intercom to the studio producer 4 2 2 2 SYSTEL IP 4 wiring configuration examples Equipment shared between two studios Have in mind the limitations stated in chapter 2 1 2 4 of this manual Audio input and output 1 AUDIO 1 is programmed from factory as AES EBU If they need to be connected to an analog system the unit must be open and the corresponding jumpers must be changed Handset 2 port will become disabled We are going to program audio 1 inputs and outputs as Digital what allows us to share the unit between two studios STUDIO SMALL1 Studio Name Name Mode tmut Name Output Name Pos STUDIO SMALL 1 SYSTEL IP4 SMALL STUDIO DIGITAL 1L QUEUE DIGITAL 1L 1 STUDIO SMALL 2 SYSTEL IP4 SMALL STUDIO DIGITAL 1R FEEDBACK DIGITAL 1R o ae e at Sg Waar e i It has one queue QUEUE 1 that is output by SYSTEL IP 4 through Output 1 internall
31. see paragraph 2 2 2 3 in this manual Use a continuous CAT5E FTP cable or better connecting wire by wire straight including the screen This way the range will exceed 300 meters If prolongations are required make sure that shielded connectors are used and that all the screens are soldered in connectors and pigtails Have in mind that the pinout is proprietary and is not compatible with hubs or Ethernet Switches nor with PoE power supply Pin1 4 48V DC Pin 5 SPK BLUE BLUEWHITE Pin 2 48V DC Pin 6 MIC Pin 3 Pin 6 Pin 3 MIC Pin 7 CONTROL WHITE amp GREEN GREEN Pin 4 SPK Pin 8 OV DC Pin 2 Pin 7 Shield OV DC ORANGE WHITE amp BROWN Pin 1 Pin 8 WHITE amp ORANGE BROWN RJ45 Handset Connector pinout This connector includes the electret microphone biasing It is compatible with most professional headsets operator micro headphones both wired and wireless that have RJ9 connection interface In order to connect one of those headsets disconnect the micro telephone 4 pin RJ9 connector from the SYSTEL IP HS body and connect a RJ9 connection cable to the headset or its base Then connect the SYSTEL IP HS micro telephone cable that was disconnected to the new headset or its base check the manual of the device 20 AEQ SYSTEL IP SA 3 1 2 4 Installation of the SYSTEL IP ST supports It consists on a base for the SYSTEL IP HS micro telephone and a 10 Tablet iPad It
32. server active in the network then the phone will automatically obtain its own IP address If a fixed IP address wants to be assigned then the following procedure must be followed Press the configuration menu key of the phone The Setup menu will appear on the screen Select option 9 for network configuration Network Select option 1 WANConnectionType by pressing the edit key Using the option key select the Static IP option Accept by means of the ok key Select option 8 Non DHCP IP Address using the edit key Type the IP that you want to assign and then accept by means of the ok key Repeat the same steps to configure the rest of parameters Non DHCP Subnet Mask Non DHCP Default Gateway and finally Non DHCP DNS1 Save network configuration by pressing the save key e Reboot the telephone by choosing option 12 Reboot Step 2 Accessing the telephone s web server Use your web browser to access the telephone s web server by typing its IP in the browser s URL BAR The telephone main screen will then appear with some basic configuration options user mode J SPAS02G Configuration Ui x M 172 26 33 86 Step 3 Activating Administrator options In order to be able to configure advanced parameters such as those related to the SIP interface the Administrator mode must be activated first In order to do that click on the Admin Login link shown at the top right corner of the screen Admin Login basic advan
33. time task IP configuration using DHCP By default the gateway accepts DHCP automatic IP configuration If your network has a DHCP server connect the SmartNode and turn it on The CD provided together with the gateway contains a utility called SNDiscovery exe Copy it to your PC s hard disk and run it Note under Windows 7 you must right click on the file and choose the Execute as administrator option Available Srnarth odes IP Address MAC Address 192 168 0297 OC AG BA06 54 048 ETA A5 2 2009 01 14 H323 SIP F 5 FXO 192 168 0 37 00 A0 BA D6 54 B0 SM4940 1E24v R5 T 2011 01 17 H323 RBS SIP a IT Settings Double click an entry to get to the web interface Right click on an entry to see additional connection options A window showing all the Patton SmartNodes available in the network will appear Just click on the IP address of the unit you want to configure to open a web browser where the configuration page will appear Default login is administrator without password If you wish you can change the DHCP assigned IP to a static one by clicking on Network gt IP DNS gt Interfaces gt eth0 typing the new IP address and subnet mask and then applying the changes with the Apply button at the bottom of the screen ImportfExport Configuration Link Supervision Status Network IP ONS DHCP NATIMAPT WserDefned aon aren AFE IP C JEE JBE SEN IP Address Dt 192 168 0 175 mar 255 255 255 0 Q
34. together with other third party applications external to the system Furthermore an internet browser can also be installed in this control and web server PC so it can behave as a control terminal like any other However only one instance of the web server must be installed and executed This instance will control each and every Systel IP system in the installation On the other hand it IS possible to install and use more than one setup application in different PCs In this case we recommend that the database folder where the control web server is installed is shared in the network and then modify the SystellpSetup ini file in the PCs where the setup application is installed to change the PATH to the database This file is located in C Program Files AEQ SystellP Setup 3 1 1 1 Installation of the setup and control web server applications An executable file is provided with each system that allows the installation of the software in Windows Vista Windows 7 amp Windows 8 operating systems Just execute the installer and follow the steps that appear on the screen 7 jB Setup Systel IP Welcome to the Systel IP Setup Wizard This will install Systel IP version 1 0 1 15 on your computer It is recommended that you close all other applications before continuing Click Next to continue or Cancel to exit Setup During that process some Ajax components are also E seup fmsotunisut besa sencha Ext 15
35. whether it is required or not that the system registers each telephone extensions into the SIP server in order to be able to send and receive calls to from it We will find the need to register extensions or not depending on how we want the SIP server and the equipment to interact how does the SIP server know which IP and port is our system listening to waiting for calls The system can either register its SIP IP and port or alternatively the server will have our IP and port linked to the extension number name by configuration Do the systems within the network have dynamic or fixed IP addresses If it is fixed it is more practical and reliable to configure than to use registering If they are dynamic it is better to require registering Registry expiration Time in seconds that the registering will remain valid in the SIP server before expiring The system must periodically refresh the registration before this time expires A longer time allows for a reduction in used bandwidth while a lower time provides the communication between system and server with better reliability in case the server is rebooted for some reason and it loses registration information the system will remain unregistered until it refreshes this registration The typical value for the registration expiration time is 1 hour Advanced settings By clicking on this button some advanced configuration parameters are accessed SIP Local Port The most commonly used is 5060
36. you do not need to change any of the such device settings here to get your network up and running Save Settings Don t Save Settings ROUTER SETTINGS Use this section to configure the internal network settings of your router The IP Address that is configured here is the IP Address that you use to access the Web based management interface If you change the IP Address here you may need to adjust your PC s network settings to access the network again Router IP Address 172 26 1 201 Subnet Mask 255 255 0 0 Local Domain Name AEQ ID Enable DNS Relay Y DHCP SERVER SETTINGS Use this section to configure the built in DHCP Server to assign IP addresses to the computers on your network Enable DHCP Server Y DHCP IP Address Range 100 to 199 address within the LAN subnet DHCP Lease Time 1440 minutes DHCP CLIENT LIST Host Name IP Address MAC Address Expired Time AVOID ARP ATTACK Avoid Arp Attack 24 DHCP RESERVATION The only thing to do here is to select the router s IP and subnet mask in case we want do change them 89 AEQ SYSTEL IP SA Internet telephony or IP telephony is a telephone service based on VoIP IP telephony is an immediate application of this technology allowing us to make ordinary telephone calls over IP or other packet networks using PC gateways and standard or IP telephones APPENDIX E BASIC IP TELEPHONY CONCEPTS IP vs VoIP telephony The basic steps
37. 9 ANALOG 3 IN Pin 2 GND Pin 10 GND Pin 3 ANALOG 3 OUT Pin 11 ANALOG 3 OUT Pin 4 GND Pin 12 GND Pin 5 ANALOG 4 IN Pin 13 ANALOG 4 IN Pin 6 GND Pin 14 GND Pin 7 ANALOG 4 OUT Pin 8 GND Pin 15 ANALOG 4 OUT Identification of the DB15 connector ANALOG I O 5 6 Pin 1 ANALOGS IN Pin 9 ANALOG 5 IN Pin 2 GND Pin 10 GND Pin 3 ANALOG 5 OUT Pin 11 ANALOG 5 OUT Pin 4 GND Pin 12 GND Pin 5 ANALOG 6 IN Pin 13 ANALOG 6 IN Pin 6 GND Pin 14 GND Pin 7 ANALOG 6 OUT Pin 8 GND Pin 15 ANALOG 6 OUT Identification of the DB15 connector ANALOG I O 7 8 Pin 1 ANALOG 7 IN Pin 9 ANALOG 7 IN Pin 2 GND Pin 10 GND Pin 3 ANALOG 7 OUT Pin 11 ANALOG 7 OUT Pin 4 GND Pin 12 GND Pin 5 ANALOG 8 IN Pin 13 ANALOG 8 IN Pin 6 GND Pin 14 GND Pin 7 ANALOG 8 OUT Pin 15 ANALOG 8 OUT Pin 8 GND 16 AEQ SYSTEL IP 2 2 2 6 Power supply F The power supply connector is located at the left side of the units back panel The power supply is of the universal auto range type 100 240 V AC 50 60 Hz 50 VA 2 2 2 7 Notes about SYSTEL IP 12 audio wiring Note in paragraphs 2 2 2 4 and 2 2 2 5 that each connector provides two balanced audio inputs and two balanced audio outputs mono in case they are analog signals stereo for digital This allows an easy audio installati
38. AEQ AEQ SYSTEL IP 4 AEQ SYSTEL IP 12 USER S MANUAL ED 01 15 V 1 1 02 03 2015 Firmware Versions CPU 1 01 FPGA 1 00 VoIP 1 00 or higher Software Versions AEQ Systel IP Setup 1 0 1 20 or higher AEQ Systel IP Web Server 1 0 1 15 or higher CONTENTS Algas lapa raise cece TP o O PA 4 11 APpicatl ns OF this Prod te EE 4 1 2 Description of the basic system SYSTEL IP 4 SYSTEL II 4 1 3 Eine e Ee Eeler e misa acacia neo 5 2 PHYSICAL DESCRIPTION OF THE UNUM asicicisiicnmanirarrric dd 7 E Od Ek NP AOS CODINA y a 7 21d DOSCrIDHOM OF INE ONL DANG ainia 7 2 1 2 Back panel description and CONNECTIONS oooooocccnccnnncccnncnnncconononancconononancconenonanennnns 7 Zi lees ls Elmermel PONS LAN amp WAN orina 8 2 1 2 2 General purpose inputs and outputs GPIO ooooooccccccccccnocncconnnonononnnnnnns 8 2 1 2 3 Handset connectors 1 and 3 9 2 1 2 4 Audio input and output Larson taa 9 2 1 2 4 1 Input 1 jumpers Contiguraton 10 2 1 2 5 Audio inputs and outputs 282 10 Place POWO SUDD APA An E O aanre 10 2 2 SYST Tee E de en NEE 11 2 2 1 Description of the front panel E 11 2 2 2 Back panel description and CONNECTIONS cccooocccnnccccncnnnconnnncnnnnnnnnnconnnnnnnnnncnonanenns 11 2 2 2 1 Ethernet ports LAN amp WAN oocooccccccoccncocnnccncononccoconcnnconanoncnnnncnnnnonnnnnnos 12 2 2 2 2 General purpose inputs and outputs GPIO oooooocccnnccccconnnccnnnccnnnonano 12 2 2 2 3 Handset connectors
39. AND I 7 5 3 Program operation window Programs are controlled in a work screen with several variants depending on the number of lines and console faders used 7 Ss EN SYSTEL IP 12 mmm eee O BUSY 00 00 T P r pa nee i y A a a LEVEL L ememr Tea t C sonz em o LEVEL LEVEL rn 00 00 d una Gel onar S ee FS SS Ce d aie ae a ee a EA a onan 2 7 sy LEVEL Q oe IDLE 00 00 LEVEL IDLE 00 00 ADMIN Programa 1 STUDIO OR HANDSET 1 96 AEQ SYSTEL IP AEQ Up at the right the queues for the calls to be sent through each fader are located The former screen manages 12 lines and six faders You can see one or two individual windows for each line at the left At the right in the middle the Reject Calls and Menu buttons are available At the bottom right when idle the Chat window for studio operators can be found and above it dynamically the Call Phonebook level adjustment and menu option windows will be temporary presented At the bottom a status bar is presented Main screen variations are Individual windows for configurations between 1 and 12 lines The screen is resized for the available number of lines SYSTEL IP l o E SYSTEL IP 7 ON AIR gt IDLE 90 00 eer 3 AF ell Mobile 1 mobile E v A Analog 1 A A Analog 2 AEQ SYSTEL IP na ra a SYSTEL IP Configurati
40. ANDSET 2 Output Name re DIGITAL 1L DIGITAL 1R HANDSET 1 HANDSET 2 It contains two queues QUEUE positions 1 amp 2 being output by the SYSTEL IP1 system through the Digital Output 1 wire that is sent doubled to the console input channels 1 and 2 The console will be setup so input 1 contains left channel and input 2 contains right channel A console auxiliary output is sent to Systel s Digital input 1 which will incorporate the program audio without telephones FEEDBACK through Digital Input 1L HANDSET 1 amp 2 are wired to provide Intercom to the producer and studio operator STUDIO_2 DIGITAL Mame _ Mode A Input Name Output Name 4 Pos STUDIO_1 DIGITAL WIRED SYSTEL IP1 DIGITAL 2L QUEUE DIGITAL 2L 1 STUDIO_3 ANALOG 6 QUEUES SYSTEL IP1 DIGITAL 2L FEEDBACK DIGITAL 2L HANDSET 3 INTERCOM HANDSET 3 HANDSET 3 HANDSET 4 INTERCOM HANDSET 4 HANDSET 4 lt has two queues QUEUE positions 1 amp 2 being output by the SYSTEL IP1 system through the Digital Output 2 wire that is sent doubled to the console input channels 1 and 2 The console will be setup so input 1 contains left channel and input 2 contains right channel A console auxiliary output is sent to Systel s Digital input 2 which will incorporate the program audio without telephones FEEDBACK through Digital Input 2L HANDSET 3 and 4 are wired to provide intercom to the producer and studio operator STUDIO_3 ANALOG 6 QUEUES STUDIO_1 III WI
41. EE 15 H DB15 connector pinout Pin 1 GPI1 Pin 9 GND_GPI 1 4 Pin 2 GPI2 Pin 10 GND_GPI 1 4 Pin 3 GPI3 Pin 11 GND_GPI 1 4 Pin 4 GPI4 Pin 12 5V GPIO Pin 5 GPO1 Pin 13 GND_GPO 1 4 Pin 6 GPO2 Pin 14 GND_GPO 1 4 Pin 7 GPO3 Pin 15 GND_GPO 1 4 Pin 8 GPO4 Highlights please note that there is a common ground GND for inputs 1 to 4 and another one for outputs 1 to 4 There is also a 5V reference voltage at pin 12 to ease the wiring AEQ SYSTEL IP 2 1 2 3 Handset connectors 1 and 2 c SYSTEL IP 4 provides two 8 pin RJ45 connectors to connect remote powered analog handsets Handset 1 port is located above handset 2 The only model that can be connected to them is the SYSTEL IP HS handset If a wired or wireless operator handset is to be connected it must be plugged in the 4 pin RJ9 handset connector provided by the SYSTEL IP HS see section 3 1 2 4 in this manual Physically each pin carries the signals described below Pin 4 Pin 5 BLUE BLUE amp WHITE Pin 3 Pin 6 WHITE amp GREEN GREEN Pin 2 Pin ORANGE WHITESBROWN Fin 1 WHITESORANGE Pin BROWN RJ45 connector pinout Pin 1 48 V DC Pin 5 SPK Pin 2 48 V DC Pin 6 MIC Pin 3 MIC Pin 7 CONTROL Pin 4 SPK Pin 8 OV DC Shield OV DC Highlights The use of shielded cable is mandatory and the shield must be properly connected to provide adequate power supply and interference suppression
42. EE 59 5 5 Control screen left area Individual line Control wimdow 59 5 6 Detailed description of the line control windows buttons indicators and fields 60 5 6 1 Operator phone Procedure to make and receive Calls ccccccooonccnnccconm 60 920 25 LING AMD IRC AO E 61 5 6 3 Send receive level Indicators ccccccccoconncccnnnccnonononncnnnnnonononannnnnnonononannnnnnnonnss 61 A A E A E E 61 5 6 5 Active indicators per line line status 62 5 66 Other status le e ata 63 5 6 7 Main buttons at the right of each Ime 63 5 6 7 1 WAIT ON AIR QUEUE button operation neno0n0neeeennnnnneeeeeeeeneen 63 5 6 7 2 Lock button OD eegekece egegedg ergeet linen inician cartas 64 5 6 8 Partner name editable field operation n nnnnonnnnnnnnnnnnnnsnnnnnnnrnnsrreronrrnsrreennnenne 67 5 6 9 Partner observations editable field operation ooocccccccnccccnnncconnnocococononoconononos 67 5 7 Using an IP phone instead of an analog handset as the SYSTEL IP handse t 67 GEESCHT 68 AE VPARP Dir laa 69 APPENDIX A INSTALLATION AND SETUP OF AN ASTERISK PBX FOR SYSTEL AND CONNECTION TO A SIP TRUNKING oooccccccconnccccccnnocanccocnonnnncnnnnonanncnnononanonennnnancrrrnnnnanerrrnnananens 70 A 1 Benefits provided to Systel IP by Asterisk n00000nnnnnnnn0nannnnnnonnnnnnnnnnnonnnnnnnnnsennnrnreeenene 70 A 2 Installing the Asterisk PBX ooooccococcococoncon
43. EQ SYSTEL IP SA Ke Admin Applications e Connectivity v Reports y Settings e Logout admin FreePBX System Status FreePBX Notices System Statistics No new notifications Processor how all show al Load Average 0 29 CPU 0 FreePBX Statistics Memory Total active calls 0 Apr Memory 10 Internal calls 0 Swap 0 External calls 0 Disks Total active channels 0 H 5 FreePBX Connections ra 0 IP Phones Online 5 Boot 5 1P Trunks Online ee Networks E oe 5 54 KBis ethO transmit 0 95 KB s Uptime eth receive 0 00 KB s eth1 transmit 0 00 KB s System Uptime 22 hours 14 minutes Asterisk Uptime 22 hours 13 minutes Server Status Last Reload 38 minutes S Asterisk MySQL Web Server SSH Server HY VW FreePBX n Schmooze we let freedom ring Copyright 2012 Schmooze Com Inc It is recommended that the Module Admin from the Admin is selected and after that upgrade all the appearing modules with the latest available version Upgrade All It will also be possible to install new modules to provide the PBX with new functionalities We recommend that the following modules are installed Announcements in order to playback pre recorded messages welcome etc Asterisk Info to have access to more info about the PBX status Asterisk SIP Settings allows advanced SIP configuration and also change the language of the voices IVR to be able to add interactive phone menus Ty Admin e Applications Connectivity e Repor
44. EUE in a deep pink color meaning that if the user clicks on it the line will return to its position in the queue 63 AEQ SYSTEL I P 5 6 7 2 Lock button operation The lock button allows locking a call on air when the lock is activated represented by a closed lock with a green tick This avoids that the call is automatically put in WAIT when another call goes ON AIR so both of them can interact simultaneously in the program The lock button is deactivated by default represented by an open lock with a red cross Have in mind that when the call is removed from air by putting it in WAIT mode or alternatively by transferring it to a HANDSET the lock will become open remaining this way until the user closes it again 5 6 7 3 QUEUE button operation console fader or channel This button allows the user to select which channel or fader will be used in the console to control each call It is called QUEUE because in normal operation the application allows queuing calls to be put on air by means of each fader In order to be able to put a call on air it is mandatory to assign it a queue This means that we have to choose which input channel of the console will be used for it So if for a given channel the queue button simply shows a Q we must click on it to choose a queue If the QUEUE button is drawn in black and the user clicks on it the call is removed from the queue If on the other hand it i
45. HAN PAYO 6003 ROGE 916861300 LM LOIS 6001 ROMA SOFIA ANDREOTTI 312345676543 TONI 915444 TONI 915444 GUS 1915334455 VIENA LUCAS 555 ROGE 916861300 EDU 34916864492 MARIBEL 916335577 BARCELONA 3493333444555 PARIS RTF PAUL 33101202303 LONDON JERRY 91131462134 Po ol Incudedin Programs o MUNICH ANDREAS 87654321987 AFTERNOON VIENA LUCAS 555 MUSIC IN THE NIGHT 3th PROGRAM Version 10 120 20 01 2015 140322 In order to manage the phone contacts of a Systel IP system we rely on a general Phone book General Phone Numbers a temporary call phonebook to make during the program Programmed Phone Numbers and a particular phonebook per program called Programs Phone Books Besides there are some tools available to ease the management of phonebooks and exchange information among them The right side of the main screen shows the General Phone Numbers area a general agenda containing all the telephones for all programs in the station Numbers in the black list are displayed in grey they can be hidden by means of the View tab located in the top right side a General Phone Numbers View All AA This same area displays some icons to add modify or delete a telephone number from the general list Modifications will take effect on the rest of agendas using this contact too Before deletion really takes effect a window that requires confirmation and informs about which parti
46. IN 18 75VDC UHT ga O E SPDIF 2 3 upper ONOFF DD CA NW KH AR d Z eg a TA Oe Me Output 1 cays igh afte kas 4 jumpers in e Deich AES mode ey wu 1910 J0ne t E q o Z 1 OT Wiioig Input 3 5 jumpers in AES mode 5 jumpers in AES mode 5 jumpers in AES mode 5 jumpers changed from AES 1 2 lower to S PDIF 2 3 upper 15 AEQ SYSTEL IP Programming the digital inputs and outputs as S PDIF In order to connect SYSTEL to pieces of equipment with S PDIF format inputs and or outputs the programming of the jumpers in pos 2 3 adapts the levels and unbalances the signals by joining o IN1 IN2 IN3 8 IN4 to their corresponding GND so the signal is taken from each IN and IN or its GND o OUT1 OUT2 OUT3 amp OUT4 to their corresponding GND so the signal is provided between each OUT and OUT or its GND 2 2 2 5 Analog inputs and outputs E The connectors used are DB15 female with the following pinout a 1 15 g Identification of the DB15 connector ANALOG I O 1 2 Pin 1 ANALOG1 IN Pin 9 ANALOG 1 IN Pin 2 GND Pin 10 GND Pin 3 ANALOG 1 OUT Pin 11 ANALOG 1 OUT Pin 4 GND Pin 12 GND Pin 5 ANALOG 2 IN Pin 13 ANALOG 2 IN Pin 6 GND Pin 14 GND Pin 7 ANALOG 2 OUT Pin 8 GND Pin 15 ANALOG 2 OUT Identification of the DB15 connector ANALOG 1 0 3 4 Pin 1 ANALOGS IN Pin
47. LAN Setup Save Settings Dont Save Settings network to operate on Enabling Hidden Mode is WI FI PROTECTED SETUP ALSO CALLED WCN 2 0 IN WINDOWS VISTA another way to secure Parental Control Rules your network With this Enabl option enabled no wireless nadie E dients will be able to see Current PIN 55360951 your wireless network Generate New PIN Reset PIN to Default wesc ges E a an Wi Fi Protected Status Enable Configured wireless devices to connect a to your router you will Reset to Unconfigured need to manually enter the Add Wireless Device with WPS a Internet Connection Time and Date Logout WPS PIN UnLock If you have enabled Wireless Security make sure you write down the WIRELESS NETWORK SETTINGS Key or Passphrase that you have configured You will need to enter this information on any wireless Enable Wireless device that you connect to Y your wireless network Wireless Network Name SSID NOMBRE_WIFI Also called the SSID Enable Auto Channel Selection e Wireless Mode Wireless Router Y Wireless Channel 6 Y Transmission Rate Best automatic Y Mbit s WMM Enable Y Wireless QoS Enable Hidden Wireless LJ Also called the SSID Broadcast WIRELESS SECURITY MODE Security Mode Enable WPA WPA2 Wireless Security enhanced Y WPA WPA2 WPA WPA2 requires stations to use high grade encryption and authentication Cipher Type AUTO TKIP AES Y PSK EAP
48. Login link shown at the top right corner of the screen Admin Login basic advanced 80 AEQ SYSTEL IP SA Step 4 Configure Proxy SIP registering parameters and user s extension In order to configure the Proxy SIP registering parameters Ext 7 must be accessed in Administrator mode Proxy and Registration sections will show these fields to be configured 11 1 11 Small Business Pro cisco SPA502G Configuration Utility Subscriber Information Proxy enter the Systel IP system IP address followed by the 5070 port according to the ip port format Register decides whether register or not the phone in the Systel IP If the phone is going to have a fixed IP address then activation of its registration is not recommended Register Expires Enter the value 60 so that the telephone refreshes its registration once every minute Make Call Without Reg Select Yes if you decided not to register the telephone in the SIP Proxy Ans Call Without Reg Select Yes if you decided not to register the telephone in the SIP Proxy User Login basic advanced Call History Personal Directory Attendant Console Status General Line Enable NAT Settings NAT Mapping Enable NAT Keep Alive Enable SIP Settings SIP Port SIP Debug Option Call Feature Settings Message Waiting Default Ring Mailbox ID Proxy and Registration Proxy 172 26 35 10 Registe
49. OG 7 ANALOG 8 INTERCOM ANALOG 8 ANALOG 8 If the QUEUE option is selected only output selection will remain enabled The drop down menu shows us only the available outputs We can choose the one we want and define by means of the Position field the number of queue or Fader within the studio Resource QUEUE from SystellP Console Fader pacos O O E Output Mame ANALOG 6 Position 6 Once QUEUE resources queues or outputs from the SYSTEL system to the studio are configured we need to configure FEEDBACK in order to establish the input circuit to the SYSTEL unit from the studio being configured 36 AEQ SYSTEL IP The drop down menu will show in this case only the available inputs We can choose the one we want Resource FEEDBACK to 5ystellP Console AUX OUT e Input VOS ANALOG 1 Name ANALOG 2 ANALOG 3 ee ANALOG 4 A ANALOG 5 ANALOG 6 ANALOG 1 DIGITAL 3L ANALOG 1 1 ANALOG 2 A ANALOG 2 2 ANALOG 3 An ANALOG 3 3 ANALOG 4 QUEUE ANALOG 4 4 ANALOG 5 QUEUE ANALOG 5 z ANALOG 6 QUEUE ANALOG 6 6 Last we need to configure INTERCOMf in order to define the bidirectional intercom circuits between the different operators and studio producers and the SYSTEL system In this section we need to decide whether an intercom circuit is created on one of the 4 remotely powered HANDSET terminals for the SYSTEL IP HS or alternatively we want to use a pair of analog or digital input
50. RED Ce CA E MS ETE a me EEN STUDIO_2 DIGITAL SYSTEL IP1 ANALOG 2 QUEUE ANALOG 2 2 ANALOG 3 QUEUE ANALOG3 3 ANALOG 4 QUEUE ANALOG 4 4 ANALOG 5 QUEUE ANALOG 5 5 ANALOG 6 QUEUE ANALOG 6 6 ANALOG 1 FEEDBACK ANALOG 1 ANALOG 7 INTERCOM ANALOG 7 ANALOG 7 ANALOG 8 INTERCOM ANALOG 8 ANALOG 8 It has six queues QUEUE positions 1 to 6 being output by the SYSTEL IP1 system through six analog wires Analog Outputs 1 to 6 to the input channels 1 to 6 of the console that has these inputs configured in mono A console mono auxiliary output is sent to Systel s Analog input 1 which will incorporate the program audio without telephones FEEDBACK through Analog input 1 Analog inputs and outputs 7 and 8 of the SYSTEL unit are wired to provide Intercom to the producer and studio operator through generic intercom systems that in the case of the operator can be substituted by an analog input of the console deviated to CUE and an auxiliary output of the console connected to a talkback key 38 AEQ SYSTEL IP Let s assume that we have another 12 line system called SYSTEL IP2 STUDIO 21 Name Mode Input Name Output Name Ire STUDIO 22 SYSTEL IP2 ANALOG 2 QUEUE ANALOG 2 2 STUDIO 24 SYSTEL IP2 ANALOG 1 FEEDBACK ANALOG 1 STUDIO 23 SYSTEL IP2 ANALOG 7 INTERCOM ANALOG 7 ANALOG 7 HANDSET 1 INTERCOM HANDSET 1 HANDSET 1 It contains two queues QUEUE positions 1 amp 2 being output by the SYSTEL IP2 system through two analog wires Anal
51. Redial Phone Book Prog Ph Redial Program Phone Book Programmed Phone Calls Redial List 60 AEQ SYSTEL IP Program Phone Book This is the phone book for the active program It is managed in the telephone menu of the configuration and scheduling application Its fields are name and number Call by clicking on the yellow telephone after selecting one of the contacts Programmed Phone Calls This is the planned call scheduling for the currently active program It is managed in the telephone menu of the configuration and scheduling application lts fields are scheduled call time name lock icon avoiding its deletion after the 00 10 Michael SS en V program ends completed call icon and program agenda call icon In order to mark a scheduled call as protected select by clicking on it and then click on the locked folder icon This will activate the lock icon Call by clicking on the yellow telephone after selecting one of the contacts Redial list This is the list of last calls made red upwards pointing arrow or received green downwards pointing arrow It performs two functions recall by clicking on the yellow telephone or feed the phonebook by means of the Save button store the numbers to from we have called being called as new contacts by means of the save button of the Redial list iw L SS Adding new contacts to the phonebook it is possible to V Call Menu Line04 Close Phone Num
52. The handsets can be installed up to 300m far from the SYSTEL by using CAT5E or superior cabling Two LEDs can be found besides each connector The yellow one indicates that there is power supply and the green one indicates that a SYSTEL IP HS handset is connected at the other end of the wire Instead of using handsets 1 to 4 IP phones can be connected to the switch in the LAN port In this case each IP phone will take one of the twelve available IP lines away 2 1 2 4 Audio input and output 1 o XLR 3 female and male connectors Balanced connection The connector at the left corresponas to the input while the one at the right is the output XLR 3 Female and male panel connectors pinout AEQ SYSTEL IP Pin 1 gt Ground Pin 2 gt Input or output Pin 3 3 Input or output Two operating modes can be selected by means of internal programming jumpers e Digital Both audio input 1 and output 1 carry two digital audio channels each one in AES EBU or SPDIF format handset 2 connector will remain available e Analog Audio input 1 and output 1 carry analog input handset 2 connector is not available in this mode When input 1 is operating in digital mode it synchronizes the SYSTEL IP 4 system with the signal it is connected to no matter if it carries an AES3 or AES11 signal On the other hand when output 1 is operating in digital mode it allows other pieces of equipment to synchronize to the SYSTEL by extracting the au
53. VoIP PBX In VoIP a virtual PBX Hosted PBX or IP Centrex offers PBX functions such as internal calls call transfer voice mails or conference as if it was a public internet service and or from the Public Service Telephone Network PSTN Using a SIP trunking on an internet access calls are routed to an IP telephony company that provides the PBX service from their own facilities 91 AEQ SYSTEL IP Virtual VoIP PBX and IP PBX Virtual PBX functions are the same as physical IP PBX The main difference is that virtual PBX are hosted in the web so the user does NOT need to purchase install and maintain a physical system The user will simply hire the virtual PBX service from an external service provider usually an IP telephony operator GK Asterisk Asterisk is a revolutionary open source IP PBX that the Digium company provided for free to the community and has become an industry reference From the very first versions it experienced an outstanding increase in installed software packages and user count The key to success is simple high flexibility free software Many voice over IP operators use and recommend it for professional use Asterisk Configuration of SYSTEL and VolP operators in Asterisk If you are going to install SYSTEL we recommend that the Asterisk environment is used There are many technical and operators with a deep Asterisk understanding around the world that can provide the system a high added value
54. VoIP phone However a VoIP phone allows the user to make full use of VoIP technology for example transmitting audio with high quality algorithms 90 AEQ SYSTEL IP AEQ Some IP audiocodecs as well as the mentioned Softphones can also be used as IP telephony terminals and hence become partners for Systel IP In order to do that they must incorporate SIP protocol that will allow them to signal and route the calls AEQ Phoenix Studio Phoenix Mobile Phoenix Mercury Phoenix Venus and Phoenix Stratos audiocodecs feature SIP and hence can be used as SYSTEL terminals IP line or SIP Trunk The IP line service IP voice line or SIP Trunk is a connection between an IP PBX or Asterisk PBX and the applications of a VolP telephony operator or Internet Telephone Service Provider ITSP allowing for the exchange of voice over IP VolP traffic Companies that want to make full use of their IP PBX not only use IP to communicate internally but also outside the company need an IP line or SIP Trunk The IP telephony operator is responsible for the interconnection of the IP incoming or outgoing calls to the Public Switched Telephone Network PSTN Additionally the operator can offer additional services such as the preservation of traditional geographic national international numbering system IP calls Voice over IP also called VolP is the technology that allows for transmission of voice through Internet in the form of data packets using
55. a while Settings gt General gt Auto lock gt Never Settings gt General gt Passcode lock gt Off Settings gt General gt Lock Unlock gt Off 3 Disable Control Center so it doesn t show up when the edges of the screen are touched Settings gt Control Center gt Access from applications gt Off 4 Activate restrictions so the tablet cannot be used with other applications what could interfere with proper SYSTEL system control and status display Settings gt General gt Restrictions gt Activate restrictions 19 AEQ SYSTEL IP SA In order to use an external keyboard this is recommended so the on screen virtual keyboard doesn t interfere with the system status display when trying to type texts in the keyboard must be linked to the iPad In order to do this just activate Bluetooth and touch on the keyboard device found Settings gt Bluetooth gt On When the keyboard is detected the iPad will automatically stop using the on screen virtual keyboard when text is to be typed in In order to maximize the application on screen Generate a link to the start screen in Safari so the Safari screen appears maximized Safari gt Export gt Add to start screen 3 1 2 3 Installation of the handset or SYSTEL IP HS micro telephone SYSTEL IP HS is a preamplified remote powered handset which must be connected to a special connector in SYSTEL IP 4 see paragraph 2 1 2 3 in this manual or SYSTEL IP 12
56. alkback output HANDSET 2 is wired to provide Intercom to the studio producer STUDIO 23 lts resources are configured the same way as Studio 22 Studio Name Name Mode Imput Name Output Name Ire STUDIO 21 SYSTEL IP2 DIGITAL 3L QUEUE DIGITAL 3L 1 STUDIO 22 SYSTEL IP2 DIGITAL 3R QUEUE DIGITAL 3R 2 STUDIO 24 SYSTEL IP2 DIGITAL 4R QUEUE DIGITAL 4R DIGITAL 3L FEEDBACK DIGITAL 3L ANALOG 4 INTERCOM ANALOG 4 ANALOG 4 HANDSET 3 INTERCOM HANDSET 3 HANDSET 3 lt has four queues QUEUE positions 1 to 4 being output by the SYSTEL IP2 system through Digital Outputs 3 and 4 sent doubled to audio inputs 1 amp 2 and 3 amp 4 The console is configured so inputs 1 amp 3 carry left channel and 2 amp 4 carry right channel A console mono auxiliary output is sent to Systel s Digital input 3 which will incorporate the program audio without telephones FEEDBACK through Digital input 3L Analog inputs and output 4 of the SYSTEL unit are wired to provide Intercom to the studio operator wiring the input to a console channel and the output to a talkback output 39 AEQ SYSTEL IP HANDSET 3 is wired to provide Intercom to the studio producer STUDIO 24 Studio Name Name Mode _Input Name Output Name Pos STUDIO 21 SYSTEL IP2 ANALOG 5 QUEUE ANALOG 5 1 STUDIO 22 SYSTEL IP2 ANALOG 6 QUEUE ANALOG 6 2 STUDIO 23 SYSTEL IP2 ANALOG 5 FEEDBACK ANALOG 5 HANDSET 4 INTERCOM HANDSET 4 HANDSET 4 It contains two queues QUEUE
57. all lines 4 or 12 depending on the model can be simultaneously live participating in a program with no loss of quality GPI O programmable functions RING WAIT ON AIR CUE PLAY 4 GPI 4 GPO and power supply on each DB15 female connector All functions are replicated over TCP IP in the control network Audio specifications Analog inputs input impedance 20Kohm Electronically balanced professional line level Nominal input level 4 dBu Max input level 24 dBu Analog outputs output impedance lt 100 ohm Electronically balanced professional line level Nominal output level 4 dBu Max output level 24 dBu Digital inputs outputs AES EBU interfaces configurable as AES 3 or SPDIF Inputs include SRC AES 1 input can be used for external AES 11 synchronization Encoding Algorithms Phone audio in G 711 G 726 G 729 50Hz 3KHz High Definition audio with G 722 algorithm 50Hz 7KHz Echo cancellation Independent digital gain control for all inputs and outputs with an adjustment range of 12 dB and muting Automatic gain control for telephone returns Configuration software and control web server 32 and 64 bit Windows operating systems Windows XP Windows Vista Windows 7 and Windows 8 Functionality Assigns audio handset IP phone and chat circuits to the different studios univocally Renames circuits Defines and manages phone books allowing the user to s
58. annot be put on air In the rest of states when clicking on the button the call will pass to the status indicated in the alternative button When a communication is established and both ends HANDSET1 unhook while an operator is talking with the partner by means of his handset the button name changes to WAIT as when clicking on the button the call will be put on hold In this case if we click on the button when in the wr sa WAIT mode the call will be put on hold and the ON AIR button function will change to ON AIR in order to be able to put the call on air when clicking on it This cannot happen if the Q button is blue with the Q legend in this case ON AIR will be displayed in gray because no fader has been selected to send this call on air more information can be found in paragraph 5 6 7 3 Once the legend in the Q button has changed to Q1 Q2 Q6 or VIP1 VP2 VIP6 line is assigned to fader 1 2 6 the ON AIR key will become active and Bei the call can be transferred to the program directly as a quick method to put a call on air without using the call queue button as explained in another part of this manual We can check this by observing how the status bar changes to red color meaning ON AIR ONAR 04 26 AIR 04 26 If the call is waiting in a queue and we click on the operator s telephone button in order to give some indication to the interlocutor in the queue the alternative button will display QU
59. art in a group and has a set of restrictions in his her rights to access some parts of the configuration and real time operation applications defined by the groups it belongs to The ADMINISTRATOR user with ADMIN login cannot be edited or deleted All other uses can be modified and or deleted Confirmation is required in order to delete an existing user Users can be created without filling in the password field Consequently no password will be required in the application for this user p BH Systel IP Setup ro Users L Weg Do mame A Systel IP Units ADMINISTRATOR a EDU EDU EE ADMINISTRATOR Y E GUS GUS PRODUCER Studio Wiring H LOIS LOIS i sss a aa aa aa TALENT H PETER PETER1 CONTROLLER ld H PRODUCER PRODUCER PRODUCER ROGE S o ADMINISTRATOR VOIP ADMINISTRATOR ia PRODUCER Security Level 2 Programs ay Phonebook O E p p a ZA ADMIN l ADMINISTRATOR Version 1 0 1 20 20 01 2015 14 03 22 The lower part of the screen shows the editing tools By hovering the mouse cursor over each button a label briefly indicating its function will appear Allows you to insert a new user Delete record Allows you to delete a user Edit record Allows you to modify a user Allows you to save modifications to a user j Cancel edit l EE Allows you to discard modifications made to a user a Refresh data Allows you to refresh modified data Confirm Confirmat
60. ation doesn t support the indicated audio codecs Unavailable the called telephone has been ringing but nobody answered the call Cancelled the call has been cancelled before being established Server Error undetermined internal error Connection Down the call has been locally ended due to lack of incoming RIP traffic No response in Registration registration error The server doesn t respond Registration Forbidden registration error The server indicates that authentication has failed authentication ID and or password Error X undetermined error An error code is indicated so it can be provided to Tech Support service 62 AEQ SYSTEL IP 5 6 6 Other status indicators We can find these other status indicators at the right of the line status indicator NEXT TO AIR when this indicator appears and the queue advance button is pressed this call will be the next one to be put on air HD Quality call is established using G 722 high quality algorithm 5 6 7 Main buttons at the right of each line A set of 4 buttons can be seen at the right of each line They manage most of the functionality of the SYSTEL line once the communication has been established 5 6 7 1 WAIT ON AIR QUEUE button operation The same button will give way to the WAIT ON AIR and QUEUE states in a cyclic way When the line is IDLE the ON AIR button is represented in grey meaning that a not existing communication c
61. be put on air in its corresponding fader When a call is put on air the former one is removed from the program and becomes on hold unless it is protected by a lock symbol In this case it won t be removed from the program and the new call will interact on air with it 5 4 2 Reject Calls and Menu buttons Reject Calls This button locks i incoming calls in the system in order to Reject Calls choose a new program or avoid listeners to enter too early in a cal 70000 program When clicking on it both the Reject Calls itself and the unhook buttons of the affected lines depicted Reject Calls here change to orange color When clicking on it again the system remains unlocked Menu Gives access to the complementary options of the operation window e Chat Font Size Adjusts the font size for both the general chat Chat Font Size 12 y and the field that contains observation about the partner at each Change Program amp Studio e Auto Answer Puts the system in auto off hook mode dset Level 58 AEQ SYSTEL IP e Handset Level Adjust the microphone and ear set level for the operator Presents adjustment faders as well as input and output Vumeters and numbers representing the actual gain When working with a mouse the level can be adjusted by simply dragging the cursor over each volume potentiometer When working on tactile screens it may be more accurate to touch the and buttons associated to each gain control
62. ber Just click on a phone number from the list not a previous contact click on this button and a new window opens that allows the user to fill in the new contact name which can be immediately Name Fourones and more stored or sent to the black list STEEN lt f Save SR Cancel 3 1111222333444 Ka A Michael K 5 6 2 Line number indicator Also close to this key the SYSTEL line number is presented from 1 to 12 in Systel IP 12 and from 1 to 4 in Systel IP 4 5 6 3 Send receive level indicators Another feature that is included in this key is the presence of two tri color LEDs that indicate the line input and output levels They can be adjusted by means of the LEVEL key 5 6 4 Hang up button Hangs up the line call not matter what is its status unless it is on air and a E wn protected by the lock This button is only active when it is red When in grey there is nothing to hang up 61 AEQ SYSTEL IP 5 6 5 Active indicators per line line status This is a multi message alternative indicator that informs about the line status not only in what respects to the exterior communication but also regarding to the listen or internal communication It also provides information about the particular errors of the communication established in that particular line Next we will explain the different possible call states Note that some states include a counter that indicates the time elap
63. between audio cutout length and bandwidth overhead 43 AEQ SYSTEL IP QoS This is a service type field DSCP in the IPv4 header That is this is a value that is included in this IP header whose function is to distinguish between voice packets and the rest when QoS Quality of Service policies are to be applied The recommended value for real time audio transmission is 46 NAT Traversal Establishes the mode the system works whenever a SIP server that is in Internet needs to be accessed from a local network In that case the SIP server must know the public IP address of the router the system is behind of and the port the SIP calls must be sent to so they reach the Systel system The NAT mode determines how to indicate the SIP server the equipment s public IP and port The admissible values are e off this is the value to be set when NAT is not used that is when both the system and the server can communicate with each other directly without a router in the middle that translates public addresses to private addresses e auto this option should be activated when NAT can occur and the system is registering extensions in the SIP server In this mode the Systel will analyze the server s replies to the register messages in order to get its public IP and ports If the SIP server doesn t have the keep alive sending option enabled to the registered equipment then the Keep Alive sending option must be activated in the Systel e static this
64. caller a welcome message before he she is being transferred to the corresponding extension This option is very easy to configure If the Announcements module has been previously installed then an option with the same name will appear in the Applications menu He Admin e Applications e Connectivity e Reports Settings e Logout admin Qe id Announcement Add Announcement Ada Announcemen BuzonDeGrabaciones Welcome Description Recording None Y Repeat Disable y Allow Skip L Return to IVR A Don t Answer Channel L Destination after playback choose one Y Submit Changes The following parameters need to be configured in the screen that is shown Description Welcome Recording lt message name gt Destination after playback Extensions This way if we return to the previously configured inbound route DesdeNETELIP and the welcome message is set as the destination Set Destination Announcements Welcome we will have reached our goal lt is worthy to note that the message should be recorded before it can be associated to the welcome In order to do that go to the System Recordings option under the Admin menu Just the same way we could have routed the incoming calls to an interactive menu V VA and let the caller choose the destination of the call among the presented options by him herself 76 AEQ SYSTEL IP A 3 3 2 Outgoing calls routing configuration In order to be able t
65. can have only one per studio INTERCOMs These are inputs outputs to the system to be able to talk off air with the calls up to 4 per studio We can use the inputs outputs available in pairs for the Intercom That is if we for example select the Digital 1L input that s only because the partner lines Digital 1R input Digital 1L output and Digital 1R outputs are free as they have not been used for QUEUE or FEEDBACK Whenever at least one of these signals is used for QUEUE or FEEDBACK the other three won t be available for INTERCOM E BR Systel IP Setup Studio Wiring E D Add Modify Delete Modify x Add Name Mode z Name STUDIO_4 DIGITAL 1L QUEUE DIGITAL 1L Ssa P 1 O DIGITAL 1R QUEUE DIGITAL 1R la DIGITAL 1L FEEDBACK DIGITAL 1L sal Studio Name kamnit ee ee a STUDIO_1 DIGITAL WIRED e HANDSET 2 INTERCOM HANDSET 2 HANDSET 2 E STUDIO_2 DIGITAL ER IP HAND 2 INTERCOM IP HANDSET 2 IP HANDSET 2 STUDIO 21 SYSTEL IP 2 a STUDIO 22 SYSTEL IP 2 STUDIO 23 SYSTEL IP 2 a STUDIO 24 SYSTEL IP 2 STUDIO SMALL 1 SYSTEL IP 4 SMALL STUDIO STUDIO SMALL 2 SYSTEL IP 4 SMALL STUDIO Programs aL Phonebook 18 ADMIN la ADMINISTRATOR Version 1 0 1 20 20 01 2015 14 05 42 The Studio Name and Systel Name fields are located in the left area of the screen Creation deletion and modification renaming is allowed for studios corresponding to a Systel unit You can find three units in this exam
66. can be found at the bottom of the screen where information about the software version as well as the nickname and user level current logged in G ADMIN Ea ADMINISTRATOR Version 1 0 1 20 20 01 2015 14 05 42 4 2 1 Systel IP Units Systel IP Units Systel IP is ready to operate as a multi equipment system sharing the same configuration and phonebook database This window shows basic information about the registered Systel IP units such as Device ID device identifier It will be automatically filled in the first time the application connects to it and is not modifiable by the user Name name of the equipment This name can be given as desired and changed at any moment Model model of the Systel IP equipment Once created this field is not modifiable By hovering the mouse cursor over the different registered units Systel IP List screen area we will see how the right side of the window displays information about the selected unit Apart from the previously described basic information we will also be presented the network configuration LAN WAN and Internal Network for each piece of equipment LAN WAN Local network where we can have both control and SIP signaling configured In case we want to make the control and SIP signaling networks different so that voice traffic doesn t interfere with the local network we must enable the WAN port this way we will have control in the LAN and SIP signaling for the v
67. cano ji asch O Gano AMB O canlo acc gt O Gano AMB can o d acc gt O Gano LIEHS O cano d acc O Ganj MBA O can o db acc gt O Gano gt MA O Gain o db acca O Gano Indicators period 1000 milliseconds Vumeters period 24 milliseconds 172 26 3 2 54184 O Also in that top side you can find a box to fix the AGCs Nominal Level EI dBu dbf nominal levels and a button for general adjustment that will set up all the AGCs to the selected level Adjust AGC to Nominal level 4 dBu There Is also an adjustment for the VUmeters A 1000 TE integrating period and the persistence of the audio l presence indicators to adapt the application to the vumeters period 24 preferences of the signal control thus allowing the user to observe incidences without paying a lot of attention milliseconds E 4 2 1 4 VoIP Test window By touching the corresponding button the VolP Test window is accessed where it ta is possible to check the VolP configuration directly on a unit the one that is selected VOIP Test This way the system TP installer or maintenance aset emm ri ee technician can make sure that the configuration is Line SIP Extension Line Status Last Call Action correct calls are made and Noetension ___ D gt No_ectension accepted properly and A A audio IS available in both _ v 2
68. ced 78 AEQ SYSTEL IP SA Step 4 Configure Proxy SIP registering parameters and user s extension In order to configure the Proxy SIP registering parameters Ext 7 must be accessed in Administrator mode Proxy and Registration sections will show these fields to be configured stis ifi Small Business Pro cisco SPA502G Configuration Utility NAT Settings SIP Settings Subscriber Information Proxy enter the Asterisk or Systel IP address here Register decides whether register or not the phone in the SIP proxy If we want to receive calls from it just select Yes Register Expires leave the 3600 default value so the telephone registers once an hour Make Call Without Reg Select Yes if you decided not to register the telephone in the SIP Proxy Ans Call Without Reg Select Yes if you decided not to register the telephone in the SIP Proxy UserLogin basic advanced Call History Personal Directory Attendant Console Status General Line Enable NAT Mapping Enable NAT Keep Alive Enable SIP Port SIP Debug Option Call Feature Settings Message Waiting Default Ring Mailbox ID Proxy and Registration Proxy 172 26 35 10 Register no Y Make Call Without Reg Register Expires 3600 Ans Call Without Reg Display Name Cisco Phone UserID Password Use Auth ID Auth ID U
69. cular phonebooks for each program contain this contact is displayed AFTERNOON MUSIC IN THE NIGHT 3th PROGRAM The phone number exist in W I 3th PROGRAM AFTERNOON MUSIC IN THE NIGHT Delete Record By placing the mouse cursor on a contact a list will appear on the lower right area displaying all programs containing it 51 AEQ SYSTEL I P In the Phonebook management main screen s left area we can find a list including all the programs of the station By clicking on each of them a list with all the contacts that his program includes is displayed at the bottom left area We can always feed this list by passing contacts from the general agenda using the green arrows Also we can add contacts to a program phonebook massively by copying them from another program by using rr the option to Program PROGRAM 1 PROCRAM 3 A ath PROGRAM Ea MUSIC IM THE MIGHT Merge Phonebook Lo We can find the Show Phone Numbers Programmed button at the top area of the Programs Phone Books window next to the Merge Phonebook button It displays a list with the contacts a program will call as it progresses during its broadcasting We can add modify or delete contacts from this list in the same way we could do with the general phonebook Programs Phone Books ER gt Merge Phonebook Show Phone Numbers programmed This phonebook allows us to organize the calls schedule to be made
70. dio sync from the audio in AES3 format 2 1 2 4 1 Input 1 jumpers configuration IMPORTANT NOTE Access and configuration to the programming jumpers requires previous experience with installation and configuration of electronic cards or computers Don t open the unit if you lack this experience at the risk of produce permanent damages or suffer electric shocks Analog or digital mode selection jumpers Audio input and output 1 AUDIO 1 is programmed from factory as AES EBU If they need to be connected to an analog system the unit must be open and the corresponding jumpers must be changed Handset 2 port will become disabled Ask Technical Support service if you need to perform this operation AES or S PDIF digital mode selection jumpers Digital inputs and outputs AES IN and AES OUT are programmed from factory as AES EBU If connection to an S PDIF system is required the unit must be open to change the corresponding jumpers Ask Technical Support service if you need to perform this operation 2 1 2 5 Audio inputs and outputs 2 amp 3 E CF XLR 3 female and male connectors Balanced connection The connector at the left corresponds to the output while the one at the right is the output XLR 3 Female and male panel connectors pinout Pin 1 gt Ground Pin 2 gt Input or output Pin 3 gt Input or output 2 1 2 6 Power supply The power supply connector is located at the left side of the units back pa
71. e is configured from factory with DHCP activated This means that if there is a DHCP server active in the network then the phone will automatically obtain its own IP address If a fixed IP address wants to be assigned then the following procedure must be followed Press the configuration menu key of the phone The Setup menu will appear on the screen Select option 9 for network configuration Network Select option 1 WANConnectionType by pressing the edit key Using the option key select the Static IP option Accept by means of the ok key Select option 8 Non DHCP IP Address using the edit key Type the IP that you want to assign and then accept by means of the ok key Repeat the same steps to configure the rest of parameters Non DHCP Subnet Mask Non DHCP Default Gateway and finally Non DHCP DNS1 Save network configuration by pressing the save key e Reboot the telephone by choosing option 12 Reboot Step 2 Accessing the telephone s web server Use your web browser to access the telephone s web server by typing its IP in the browser s URL BAR The telephone main screen will then appear with some basic configuration options user mode J SPAS02G Configuration Ui x X O 172 26 33 86 Step 3 Activating Administrator options In order to be able to configure advanced parameters such as those related to the SIP interface the Administrator mode must be activated first In order to do that click on the Admin
72. ed further help Please contact the Technical Support service of your AEQ dealer or directly with out central Technical Support 86 AEQ SYSTEL IP SA Included here the configuration example for the D Link CLOUD ROUTER N300 wireless access point to provide access to the Systel IP control tables APPENDIX D SETTING UP A WiFi ACCESS POINT FOR SYSTEL IP Configuration is carried out with any web browser by connecting to the router s IP address In order to do that just connect a network cable to any of the 4 LAN ports of the router don t use the yellow one corresponding to the WAN network The routers default IP is 192 168 0 1 but the actual IP in this example was changed to 172 26 1 201 A screen similar to this one will be displayed gt C ff D 172 26 1 201 index asp ol 332 Aplicaciones F sonalizar v nculos Tutorial de Java Es Using Swing Compo Baby WebServer Otros marcadores a Esta p gina est escrita en Quieres traducirla Traducir No Configuraci n v Product Page DIR 605L Hardware Version Bx Firmware Version 2 00 Login to the router User Name Password Login WIRELESS Copyright 2009 2013 D Link Corporation All rights reserved A user and password is prompted in this dialog The default user is admin and the current password is aeqaeq there is no password by default Once successfully logged in a screen like this w
73. eld blank This field will be however necessary if the Host field is not filled in Host SIP phone s IP address or host name If the SIP phone registers in Systel IP as if it was a proxy then this field may be left blank as the User Name will suffice At this moment Systel IP doesn t implement NAT traversal aids for the IP handset so they wont work properly when routers implementing NAT have to be crossed this is common when accessing through Internet 46 AEQ SYSTEL IP Port Telephone s SIP port only UDP is supported through which the phone receives the IP messages The standard SIP port 5060 will be adequate in most cases depending on the SIP phone Auth Password Password required to the SIP phone when it tries to register in Systel IP This field is currently unused as authentication of users registering in SIP is not implemented Auth Id User name associated with the password requested to the SIP phone when it tries to register in Systel IP This field is currently unused as authentication of users registering in SIP is not implemented Line Number Line number from the 4 or 12 available ones depending on the Systel IP type where the IP handset will be configured This parameter is used to fix a certain IP handset to a given SIP line of the Systel IP system If the value 0 is configured then a line will be automatically selected among those free in the moment when the IP handset is used The val
74. ensron applied yet This way it is indicated that the appearing status corresponds to the currently CONNECTED iphandseti g722 configured extension and not to that selected in the corresponding field e Not associated to any extension or IP handset e Line connected to an IP handset with a coding algorithm as in the sixth example of the attached image e Incoming call indicating for extensions the incoming number and for IP handsets the number dialed by the user e Calling a phone number The states corresponding to a number being dialed and when the remote phone is already ringing waiting for reply are distinguished e Line error the extension cannot correctly register in the PBX etc A field that might contain complementary information is displayed in the lower part of the screen when clicking on the blue icon The complete states list for the test calls made from this screen is equivalent to the call states list from the Real Time Operating software It can be looked up in chapter 5 6 2 Last Call The seventh field Last Call provides details about why a line test failed A field that might contain complementary information is displayed in the lower part of the screen when clicking on the blue icon el Error 503 Action The eighth field Action displays for each line a button to execute the suggested action to be performed on that line to continue with the test Operating procedure Any SIP telep
75. ent within the network The Close Customize Units button allows you to close this new window i a a 1 Cep RS ar Clase Customize Units 44 AEQ SYSTEL IP 8 BH Syste IP Setup Lo 19 een ES VOIP E Systel IP Units Common SIP Parameters EE y Audio Codecs PrE Studio Wiring ae As Audio Codecs Order LIZ a ua G711A ise G726 32 a y G726 32 SIP Server Host Seasity Level SIP Server Port 5060 SIP Server Host e Registry expiration 3600 Seconds 3 SIP Server Port 5060 A Outbound Proxy Host Registe Y SN e Users dai r d Outbound Proxy Port 5060 Registry expiration 3600 Seconds SIP Local Port 5060 Advanced settings RTP Ports 5004 EJ oe SIP Local Port 5060 Call Transfer DTMF Tones 2 E e LV RTP Ports 5004 Buffer Size 20 Milliseconds Close Customize Units le Call Transfer DTMF Tones 2 Packet Size 20 Milliseconds en Buffer Size 20 Milliseconds Qos 46 Packet Size 20 Milliseconds NAT Traversal off auto static O D Qos e NAT Traversal off auto static SYSTEL IP1 D SYSTEL IP2 SYSTEL IP4 SMALL STUDIO amp ADMIN El ADMINISTRATOR Version 1 0 1 20 20 01 2015 14 03 22 This new window shows the configuration for the selected Systel common Parameters SIP iv IP unit This configuration will be as specified in Common SIP Parameters as long as the Common Parameters SIP checkbox is checked
76. er to update their firmware to possibly newer versions containing a improvements etc By selecting the desired piece of equipment and clicking on the _ ara Firmware button a new window will be accessed e Firmware upgrade Systel IP 172 26 38 100 IP 12 SYSTEL IP1 CPU 98 01 07 25 14 cPU Upgrade FPGA 1 0 05 09 14 FPGA Upgrade VOIP 1 1 0 10 23 13 vOIP Upgrade VOIP 2 1 0 10 23 13 VOIP 3 1 0 10 23 13 This window offers information about the current Firmware versions of the Systel IP system besides the option to upgrade CPU FPGA and VolP modules to possible future versions including some kind of improvement in the operation of our Systel IP system We will have access to the File Explorer by clicking any of those three buttons in order to choose the corresponding firmware file to update Files are located by default inside C Program Files AEQ SystellP Setup Firmware And new ones can be received from AEQ post sales service or one of the authorized dealers 4 2 1 3 Adjust Gain window By clicking on the corresponding button we can access the window that allows NY supervision and configuration of the audio levels of the input and output lines of lt each Systel IP system It displays and allows adjustment of Adjust gain Lines INPUT OUTPUT Input and output of each of the 12 IP phone lines or 4 in Systel IP 4 input gain input AGC reference level and output gain
77. eues as depicted in the previous figure It contains two queues QUEUE positions 1 2 being output by the SYSTEL IP 4 system through the Output 1 wire programmed as digital It is being sent doubled to channels 1 and 2 of the console configured so that input 1 receives left channel and input 2 receives right channel A console auxiliary output is sent to Systel s input 1 configured as digital which will incorporate the program audio without telephones FEEDBACK through Digital input 1L HANDSET 1 and 2 are wired to provide Intercom to the studio producer and studio operator System used in a single studio Analog console one or two queues 2 handsets Studio Name Name m e JI imput Name Output Name Pos STUDIO SMALL 1 SYSTEL IP4 SMALL STUDIO ANALOG 3 QUEUE ANALOG 3 1 ANALOG 3 FEEDBACK ANALOG 3 HANDSET 1 INTERCOM HANDSET 1 HANDSET 1 HANDSET 2 INTERCOM HANDSET 2 HANDSET 2 If the studio is analog then we can have 2 handsets and a queue as depicted in the image above It contains one queue QUEUE position 1 being output by the SYSTEL IP 4 system through the Output 3 wire that is sent to channel 1 of the console configured so that input 1 receives mono audio A console auxiliary output is sent to Systel s input 3 configured as digital which will incorporate the program audio without telephones FEEDBACK through Input 3 HANDSET 1 and 2 are wired to provide Intercom to the studio producer and studio operator
78. g them Only those SIP phones registered in the Systel IP database will be allowed to make and receive calls Systel IP equipment defines a specific SIP port for IP handsets different to the SIP extensions SIP port By default this port is 5070 This way any IP phone to be used as IP handset needs to use the following proxy server lt Systel IP LAN port IP address gt 5070 This means that each IP handset will be tied to a single Systel IP unit The registration of the IP phones in the Systel IP as if it was a proxy is completely optional If it is going to register then it will be enough to configure its user name in the database If on the other hand it won t register then its IP address and SIP port need to be configured in the database as well Whenever the SIP phones can have a fixed IP address it is recommended that they don t register in the Systel IP system but then their IP addresses will have to be configured in the database All the IP handsets to be used will be created one by one in this section The Studio Wiring section will allow for the association of the IP handsets to the studios These are the parameters that can be configured for each IP handset User name User name configured in the SIP phone This field will be commonly required However if the particular SIP phone admits calls simply directed to its IP address and SIP ports without the name being specified then it will be possible to leave this fi
79. hare edit and copy them Manage a call scheduler and a black list Configure the initial audio levels for each line and each study Configures the format of the customer s screen defining the number of lines per program and operation with one or several up to 6 call queues SIP configuration for communication with an IP PBX FXO gateway and external Internet or internal LAN or WAN service providers Multi unit operation shares resources between the different systems in a network Web control client There is at least one compatible web browser for each one of these operating systems Windows and IOS for iPad Functionality Call establishment by number dialing with SIP identifiers or from call book entries Optical tally and acoustic RING signal Caller identification Accept incoming calls either manual or automatically Register new contacts in the call book Talk by means of the headset or microphone headphone with the people at the remote line end Put calls on hold while the caller can listen to the program AEQ SYSTEL IP Put calls on air either individually or several calls simultaneously so they can contribute to the program Assign the VIP function to one of the partners putting it ON AIR using a dedicated fader Display and adjust of the input levels manually or using an AGC the return levels of each of the studio phone lines or the headset itself Display the status of al
80. has been initially homologated for Apple iPad 2 iPad 3 and later Please ask us about an updated list of homologated devices If the Tablet iPad is 10 sized but is not in this list please check whether the connectors and controls position are compatible with the support especially whether the connectors for power supply and speaker are free The tablet can be tilted to avoid reflections in the screen In order to assembly it all we need to have is a 10 iPad a Phillips screwdriver a 3mm Allen key and a SYSTEL IP HS The packaging includes the base the tablet support 2 black screws for the hinge and 5 small clear screws for the SYSTEL IP HS Start by fixing the tablet support to the base by means of the two screws for the hinge Next turn the base upside down supporting it on the tablet support in order to fix the SYSTEL IP HS handset to the base by means of the 5 small screws If the silicone feet provided with SYSTEL HS are there they must be removed before installing the screws They can be sticked again below the SYSTEL IP ST In order to fix the tablet just loose the Allen screw located at the rear of the support slide the tablet below the claws until it is centered make sure that the controls and connectors are clear and then re tight the Allen screw while tightly holding the tablet with the claws until it is properly tightened In order to adjust the tablet tilt just use the hinge In order to g
81. he unit It is VERY IMPORTANT to turn the unit off and disconnect the power supply cable before opening it Remove the 10 screws depicted below 4 at the upper cover the 3 upper ones at the left side and the 3 upper ones at the right side Pull upwards from the cover and remove it Pd 4 _ i e IO Recognizing the programming jumpers area The jumpers are located at the front left quadrant below the ribbon cables that can be easily moved in order to access the programming Each jumper blocks has 3 pins The AES position configured by default in the system corresponds to pins 1 2 with the jumper in the lower position closer to the front panel GE 3 ewe Output 3 alee de ES pay E y a ane in IEN 3 3 mode 639 D hy re wy S S LA E v A wm 5 of S Ei cl A SEE zy Em pb SCH Output A a Bd gt ee dr 4 jumpers in 3 igs o s AES mode Ce z p AN man ep r a WC GE 9919 9619 di o S D Na BNST Jas l gt DAA z St S EE TTT gt Za e z i El ei ell U Th Gi A Ae oad a E 4 D d Sin i r Sb ei S AO a a A A Tih la O oe i i g HE Ia e Ji atl A Wee GAO t fy Late ba lt Ss di D N N W J n NS RE N Voy NQ a e 13 3 E lt r D d Sr mo 4 Es d kk ei ale gt i 4 jumpers changed EC9BW 48D15 S ara moe ERA ou ira Irie o a SI leo De UA ae ug ES from 3 er V SE w 77an Ee AES 1 2 lower to
82. hony extension or any IP handset from those registered in the data base can be selected for each of the lines by the technician When an extension or handset is selected for a line the user will see how the line status changes in the screen while the necessary operations are performed registering the extension in the SIP server etc In case that an authentication error is produced the technician is able to modify the extension data Auth Id and or Password and check whether the issue gets solved or not When other kinds of errors occur no response from server etc it will be necessary to return to the VolP configuration check the parameters and synchronize the system with the database again if any of them is changed 33 AEQ SYSTEL IP AED Line Status REMOTE_RINGING 6002 6002 INCOMING_CALL sip 6001 172 26 1 80 SIE Panor DB os If the line status turns to OK an outgoing call can be made by clicking on the Call button appearing on the line itself A window that allows the user to select the intercom handset device to be used and dialing a destination number will show up The intercoms list will contain those devices configured for the studios and created in the equipment we are testing with On the other hand the system will only list the studio IP handsets assigned to one of the system lines in this moment Test Line Call Intercom HANDSET i STUDIO 1 Destination 06391839888 Whenever an incoming
83. ide service to authorized users only ease the establishment of call routing policies and provide the user with added capabilities SIP also provides registration functions that allow the user to inform the Proxy server about its current location For more detailed information about SIP protocol checking this link is recommended http tools ietf org html ric326 1 92 AEQ SYSTEL IP
84. ill show up Pagina del producto DIR 605L Versi n de hardware Bx Versi n del firmware 2 00 A continuaci n se muestran los par metros de red actuales y el estado de la conexi n Si desea volver a configurar sus par metros inal mbricos haga clic en el bot n Configurar Tambi n puede acceder a los par metros avanzados haciendo clic en Configuraci n manual Par metros de Internet Conexi n a Internet IP din mica DHCP Estado Desconectado Par metros inal mbricos Nombre de red SSID DLINK_ID Estado Configurar Seguridad Autom tico WPA o WPA2 Personal Clave de red CLAVEDELAWIFI Informaci n del dispositivo Nombre de usuario admin Contrase a a2eqaeq Cuenta mydlink Estado Sin conectar No ha activado el servicio mydlink em gem Cancelar Configuraci n manual WIRELESS 87 AEQ SYSTEL I P SA The basic router information is displayed in this screen Click on Manual configuration to configure the router After clicking on Wireless Connection something like this should appear orcos e ova once stares Easy Setup Helpful Hints a i Enable Auto Channel Scan Use this section to configure the wireless settings for your D Link Router Please note that so that the router can Wireless Connection changes made on this section may also need to be duplicated on your Wireless Client select the best possible channel for your wireless
85. ink color in the VIP a button both in the queue selection window and in the queue button of the line window Close Mr Jonathan Payo is a respected leader of the Gypsu community will answer listener s questions about issues of social integration JONATHAN PAYO Note that the Q2 button in the line button has changed to pink color and its text to VIP2 while the QUEUE 2 has changed its label to pink too By clicking the NEXT TO AIR button the call is put on air by means of an exclusive fader that remains locked until the call is terminated JONATHAN PAYO Mr Jonathan Payo is a leader of Gypsy community He ll answer listener s Peter is a priest who works with members of the hispanic community In this example Jonathan has been on air for 4 18 minutes controlled by the QUEUE 5 fader while PETER RODRIGUEZ has been waiting for his turn for 4 23 minutes in QUEUE 6 fader 65 AEQ SYSTEL IP 5 6 7 4 Level button operation LEVEL When clicking on this button a window will pop up showing some faders and vumeters to adjust the input level for each line in the system as well as the general return level for that line to the line partner When working with the mouse each fader can be dragged to put it in the desired position When working with tactile screens it may be more accurate to touch the and keys associated to each gain adjustment The unit is adjusted from facto
86. ion is required in order to delete an existing user rece recone Cancel 49 AEQ SYSTEL IP SA This screen allows for the creation and management of the programs that are going to use certain shared Systel resources From this Windows we can add modify and delete programs When selecting an already created program we can see which users are authorized to operate it Some users can be deleted and different users can be added from the general user list available in the lower left area of the window BE 4 2 6 Programs rere a BH Systel IP Setup SS Programs T Des z Incoming Calls Systel IPUnits Programs O ES Es as Lines Group D Status Add Delete 6001 4 WR E 1 y A 0 dE Studio Wiring slk o dt PROGRAM_1 E slk 8 alt Loi PROGRAM 2 D F 0 gt 41 gt 20 PROGRAM e Ee ee HES ee vom AFTERNOON eee e Se 8 MUSIC IN THE NIGHT zE a or E fa 0 41 gt Security Level slk o WEN vs CA v fa 0 4 1 gt Users Outgoing Calls Lines configured may be used to make outgoing calls while on IDLE state E p LOIS LOIS ADMINISTRATOR ADMIN ADMINISTRATOR AL PETER PETER1 PRODUCER PRODUCER PRODUCER Phonebook CUS GUS GUS o JoUS PRODUCER EDU EDU LOIS LOIS TALENT PETER PETER1 CONTROLLER PRODUCER PRODUCER PRODUCER ROGE
87. l change to the WAIT status if it has any line assigned Dialing When in idle mode an user can dial from the phone itself and will call using the first available line changing to the on_handset status Making a call from a specific line is also allowed by dialing the line number followed by the separator and the number to call to Dialing by software will first force the assigned IP handset to call and when it unhooks it will dial from the line being used If it doesn t respond in a given time or it is not unhooked the call will be made but passing to WAIT status Sending tones When an IP handset is assigned to a line and it is connected tones can be sent to the receiver by using the telephone keyboard 22 AEQ SYSTEL IP 4 SETUP SOFTWARE In this chapter the main features of the Setup software and control web server will be described lt operates on 32 and 64 bit Windows operating systems Windows XP Windows Vista Windows 7 and Windows 8 Purpose Configuration of the system and management of the contacts and call phonebooks This software allows you to Assign audio circuits handsets IP phones and chats to the different studios exclusively Rename circuits Define and manage phonebooks allowing the user to share edit and copy them Manage acall scheduler and black list Setup the initial audio levels for each of the lines and each of the studios Set up the format of the client sc
88. l the phone lines and where they are being routed to Distinguish between producer operator and presenter roles Label and comment calls and chat among the different controllers assigned to the program AEQ SYSTEL IP SA 2 PHYSICAL DESCRIPTION OF THE UNIT In order to understand the installation and cabling process associated to this system first of all the installer needs to become familiar with the connectors and the rest of elements present at the front and back panels of the unit 2 1 SYSTEL IP 4 description 2 1 1 Description of the front panel There are six LEDs indicating the status of the system and the communication These are the indicating LEDs from left to right HS1 and HS2 LEDs indicate the status of the handset connections e Off no connection e Green connected WAN LED indicates the status of the IP Phone WAN network connection e Off not connected to a phone network e Flashing green physical link established LAN LED indicates the status of the local control network connection e Off no connection to the local network e Flashing green physical link established HW ERROR LED indicates the initialization or error status of the unit e It is red lit during the boot of the system If it doesn t extinguish within some seconds the unit needs servicing POWER ON LED indicates the status of the system power supply e Off no power supply e Green the unit is powered correctly Besides
89. lso a scroll bar and an Accept button 5 5 Control screen left area Individual line control window Peter is a priest who works with members of the hispanic community Each line is managed from this window At the left the operator s line unhook and hang up buttons as well as the incoming or outgoing call line number and send receive levels indicators are presented Below that in the center the main line status indicator together with a timer indicating how long that line has remained in that status etc can be found Up in the middle the partner name editable field is displayed and below it the operator s partner observations field is available The four active line management buttons are located at the right lock WAIT ON AIR QUEUE LEVEL and Queue 99 AEQ SYSTEL IP 5 6 Detailed description of the line control windows buttons indicators and fields 5 6 1 Operator phone Procedure to make and receive calls When an incoming call is received these two images appear alternately meanwhile an acoustical warning is produced in the PC or tablet Unhooked telephone ready to talk to the partner Wien a call is received just click on the blue intermittently star surrounded telephone icon and it will be passed to the operator s handset or intercom When the call is ON WAIT or ON AIR the key will remain green inviting the user to click on it to recover the call for the operator
90. m speech language listening and deleting messages etc In order to do this please ask an expert or our technical support department 13 AEQ SYSTEL IP Regarding extensions related to Systel IP units no voicemail activation is required Extensions are added within the Applications menu under the Extensions option The most basic parameters that must be configured for each extension are User Extension lt extension number gt Display Name lt name gt secret lt password gt dtmfmode SIP INFO application dtmf relay only for Systel IP extensions Voicemail Status enabled only if voice mail is required In order to configure the SIP phones please check their user manual Configuration of extensions for Systel IP was described in section 4 2 3 As a rule of thumb it will only be necessary to configure the proxy and registrar with the Asterisk PBX IP Address username with the extension number and password with the corresponding value A 3 3 Telephony provider SIP Trunk configuration If external calls are provided by means of a SIP telephony provider then the most adequate thing to do is to integrate the connection to that provider inside the PBX This way it will be easy to introduce welcome messages interactive menus and route the received calls to either corporative extensions or Systel IP extensions As san example we will describe how to configure a connection to Netelip provider ru K P Admin v
91. mass e 8 E GPIO HD VoiP TELEPHONE OW AJR SYSTEM 1 2 WAN AEQ SYSTEL IP 2 2 2 1 Ethernet ports LAN amp wan A SYSTEL IP 12 includes two Ethernet ports By using the LAN port the unit can be connected via IP to a local network for its configuration and control from the application installed on a configuration PC and the web control server It can also be controlled from different web clients installed in computers or tablets By means of the WAN port the unit can receive calls from a SIP proxy Asterisk or similar from Gateways or from IP phone providers by means of SIP trunkings Up to 4 IP phones can also be connected to the WAN port in order to communicate off air with the interlocutors instead of using analog handsets Physically the connectors are RJ45 10 100 BT with the following pinout Pin 4 Pin 5 BLUE BLUE amp WHITE Pin 3 Pin 6 WHITE amp GREEN GREEN Pin 2 Pin ORANGE WHITESBROWN Fin 1 WHITESORANGE Pin 8 BROWN RJ45 connector pinout 2 2 2 2 General purpose inputs and outputs GPIO 3 1 100000000 0000000 15 3 The connectors used are DB15 female with the following pinout Identification of the DB15 connector GPIO 1 4 Pin 1 GPI1 Pin 9 GND_GPI 1 4 Pin 2 GPI2 Pin 10 GND_GPI 1 4 Pin 3 GPI3 Pin 11 GND_GPI 1 4 Pin 4 GPI4 Pin 12 5V GPIO 1 4 Pin 5 GPO1 Pin 13 GND_GPO 1 4 Pin 6 GPO2 Pin 14 GND_GPO 1 4 Pin 7 GPO3 Pin 15 GND_
92. mode should be selected when NAT can occur and the system doesn t register its extensions in the SIP server In this case the use must configure the SIP server s public IP address and its local SIP port IMPORTANT NOTE In this case the network administrator in the Systel side must configure a PORT FORWARDING from the router to the Systel IP private address indicating the same ports that the equipment local SIP port All these parameters are common for all the Systel IP equipment registered in the Systel IP Setup software They are provided pre configured from factory for the most common work situations If a specific configuration is required for any of the Systel IP units we must click on the Customize Units button A new window will appear in the right area of the screen This window offers the user the possibility to customize the parameters for the units selected in the lower right area of the screen in addition to the previously described parameters a new one appears Outbound Proxy Host Optional field containing the name or IP address of the SIP signaling Proxy server which allows the system to reach Internet in case the network administrator so decides In general Proxy servers act as mediators between clients and servers Most typical Proxies are the ones used in HTTP to browse Internet and SIP for IP telephony By means of the Proxy servers network administrators have a better control on how Internet is accessed from the equipm
93. ncludes a tool designed to manage backup copies of the data base or copy systems during the deployment of a SYSTEL IP system in several production centers This tool is accessed from the last menu option 4 2 8 Saving and restoring the database Y a BH Syste IP Setup Se Systel IP Security Level a Programs D Lig Phonebook A AEQ S Backup database Restore database 8 ADMIN i ADMINISTRATOR FA Version 1 0 1 20 20 01 2015 14 05 42 By clicking on Backup database button a new window will open allowing you to choose the backup file name and the folder to be stored A uz p C Program Files AEQ SystellP Setup Backup E EX Nombre Backup Date 2014 08 4 Time 20 28 40 Tipo Backup Database By clicking on Restore database button you can load a backup file previously stored Confirmation is required 94 AEQ SYSTEL IP 5 CONTROL TERMINAL BASED ON WEB BROWSER SYSTEL control includes a Web server that allows real time control and agenda creation remotely from different work places with hierarchical organization organized to achieve a high productivity and operation error prevention with the help of a standard web browser we can guarantee compatibility with at least one browser for each of these operating systems Microsoft Windows and Apple iOS 5 1 Web Control Client functional description Make calls by dialing numbers SIP iden
94. ndo All Changes Submit All Changes 2009 Cisco Systems Inc All Rights Reserved SPA502G IP Phone In order to configure the telephone s user extension data check next section Subscriber Information The parameters to be configured are Display Name name of the person that will use the telephone User ID name or number of the extension to assign to this telephone Password password assigned to this extension in case that Asterisk PBX registering is activated It is not necessary in case of registering is in a Systel IP Use Auth ID the usual value to fill in here is No Auth ID leave blank if the previous field was filled with No Last to make the telephone store the changed settings and reboot using them click on the Submit All Changes button 79 AEQ SYSTEL IP APPENDIX B2 SETTING UP AN IP PHONE TO BE USED AS A HANDSET FOR OFF AIR CONVERSATION WITH THE SYSTEL IP CORRESPONDENTS For example Cisco SPA303G SPA 502G or similar Next we will configure step by step a Cisco SPA502G IP in order to use it as a handset allowing for conversation using the micro headphone and dialing with the keyboard What we basically need to configure are the SIP interface parameters at the phone It consists on the configuration of the telephone s SIP interface parameters and also its addition in the configuration software according to the procedure explained in section 4 2 3 Step 1 Assignation of an IP address The phon
95. nel The power supply is of the universal auto range type 100 240 V AC 50 60 Hz 25 VA 10 AEQ SYSTEL IP 2 2 SYSTEL IP 12 description 2 2 1 Description of the front panel There are eight LEDs indicating the status of the system and the communication These are the indicating LEDs from left to right HS1 to HS4 LEDs indicate the status of the handset connections e Off no connection e Green connected WAN LED indicates the status of the IP Phone WAN network connection e Off not connected to a phone network e Flashing green physical link established LAN LED indicates the status of the local control network connection e Off no connection to the local network e Flashing green physical link established HW ERROR LED indicates the initialization or error status of the unit e It is red lit during the boot of the system If it doesn t extinguish within some seconds the unit needs servicing POWER ON LED indicates the status of the system power supply e Off no power supply e Green the unit is powered correctly Besides the power switch can be found in some versions at the right side of the unit s front panel 2 2 2 Back panel description and connections TILLI 000000 Seeeee a 1 0009 sn 0 0 0 eeaagace SU 4 TILLI eeecee eeeeee eeaece eeaeae eeaace eeacea p SYSTEL IP PY AEQ q 2 48V DC POWERED E e AUDIO HANDSETS EN E e a
96. no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface ethO router no shutdown port fxo 0 0 encapsulation cc fxo bind interface IF_FXO 00 switch no shutdown port xo 0 1 encapsulation cc f xo bind interface IF_FXO 01 switch no shutdown Its worth noting that the FXO ports configuration will depend on the country the gateway is going to be used in By default if nothing is changed the FXO ports will have the configuration recommended by ETSI used in most countries in Europe There are configurations available for other countries au ch gb mx nl nz us za The way to indicate the desired configuration for a FXO port in the configuration file is to add the use profile fxo command like in the following example corresponding to United States of America port fxo 0 0 use profile fxo us encapsulation cc fxo bind interface IF FXO 00 switch no shutdown Step 3 Loading a configuration file Once the gateway is connected to the network open a web browser and type its IP in the URL bar Click on the Import Export options in the menu that appears Import the configuration select the previously created file and click on Import Next click on Reload and click on in again in the confirmation dialog to load the new configuration and reboot the gateway Once the gateway reboots it will be configured and ready to be used together with your AEQ SYSTEL IP system Do you have additional questions or ne
97. o call outside by means of a SIP telephony provider a routing rule must be provided to the PBX for outgoing calls re W yE Admin v Applications Connectivity e Reports e Settings Logout admin KE Add Route Add Route J ToPSTN Route Name Route CID C Override Extension Route Password Route Type Emergency Ll intra Company Music On Hold default v Route Position Last after ToPSTN v Add More Dial Pattern Fields Dial patterns wizards pick one v Trunk Sequence for Matched Routes 0 y 1 Y 2 v We will select the Outbound Routes option from the Connectivity menu The outgoing routes addition screen will appear These are the fields we must fill in Description TONETELIP Dial Patterns that will use this Route XXXXXXXXX Trunk Sequence for Matched Routes 0 Netelip In this case the external SIP provider will be used whenever the dialed number has nine digits 17 AEQ SYSTEL IP APPENDIX B1 CONFIGURING A GENERIC IP PHONE AS AN EXTERNAL EXTENSION COMPLEMENTARY TO THE SYSTEL IP INSTALLATION For example Cisco SPA502G or similar Next we will configure step by step a Cisco SPA502G IP phone so it gets integrated either into the Asterisk PBX or with a Systel IP system What we basically need to configure are the SIP interface parameters at the phone Step 1 Assignation of an IP address The phone is configured from factory with DHCP activated This means that if there is a DHCP
98. occoconconononononcnnononcnnconanononnnnrnnonnnrnnonannnconancnnnnnnens 70 A 3 Asterisk PBX configuration nnsnennnneennnennnnsennenrerensrnrrrsnrnnrnnrrnnnrnrnnrrrsnrrernnrrnsnnrrrnnrrenne 73 A 3 1 General Configuration Settings menu 73 A 3 2 Extensions configuration occcoconcnccconcnnconononcnnonnnnnonnnnoonanonnnnoncnnononrnnonnannncnnarinenaneos 73 A 3 3 Telephony provider SIP Trunk Gontfguraton 74 A 3 3 1 Incoming calls routing configuration oocccccccnnnccnnccnnnccnncnonancnnnononanennnos 75 A 3 3 2 Outgoing calls routing configuration cccccccccnnnonnnccnncnnnonanncnnnnncnononannnnns 77 APPENDIX B1 CONFIGURING A GENERIC IP PHONE AS AN EXTERNAL EXTENSION COMPLEMENTARY TO THE SYSTEL IP INSTALLATION sccceeeeeeeeeeeseesseeeeeeeeeeeees 78 APPENDIX B2 SETTING UP AN IP PHONE TO BE USED AS A HANDSET FOR OFF AIR CONVERSATION WITH THE SYSTEL IP CORRESPONDENTS cccccsssesseeeeeeessseeeseneees 80 APPENDIX C INSTALLING AND CONFIGURING A GATEWAY ccccccsessseeeeeeseeeeeeeneeees 82 APPENDIX D SETTING UP A WIFI ACCESS POINT FOR SYSTEL IP cccccccessseeeseeeeees 87 APPENDIX E BASIC IP TELEPHONY CONCEPT S cccceeeeeeeseeeeeeeeeeeeeeeeessneeeeeeeeeeenneeeeees 90 3 AEQ SYSTEL IP 1 INTRODUCTION 1 1 Applications of this product SYSTEL IP is a call in system with multiconference capability that drastically reduces the costs for this
99. og Outputs 1 and 2 to the input connectors corresponding to channels 1 amp 2 of the console that uses these inputs as mono A console auxiliary output is sent to Systel s Analog input 1 which will incorporate the program audio without telephones FEEDBACK through Analog input 1L Analog input and output 7 of the Systel are wired to provide Intercom to the producer and studio operator wiring the input to a console input and the output to a talkback output HANDSET 1 is wired to provide Intercom to the studio producer STUDIO 22 Studio Name Mame Mode nput Name Output Name 2 Pos STUDIO 21 SYSTEL IP2 DIGITAL 1L QUEUE DIGITAL 1L 1 STUDIO 23 SYSTEL IP2 DIGITAL 2L QUEUE DIGITAL 2L 3 STUDIO 24 SYSTEL P2 DIGITAL 2R QUEUE DIGITAL 2R lt DIGITAL 1L FEEDBACK DIGITAL 1L ANALOG 3 INTERCOM ANALOG 3 ANALOG 3 HANDSET 2 INTERCOM HANDSET 2 HANDSET 2 It has four queues QUEUE positions 1 to 4 being output by the SYSTEL IP2 system through Digital Outputs 1 and 2 sent doubled to audio inputs 1 amp 2 and 3 amp 4 The console is configured so inputs 1 amp 3 carry left channel and 2 amp 4 carry right channel A console mono auxiliary output is sent to Systel s Digital input 1 which will incorporate the program audio without telephones FEEDBACK through Digital input 1L Analog inputs and output 3 of the SYSTEL unit are wired to provide Intercom to the studio operator wiring the input to a console channel and the output to a t
100. oice traffic in the WAN ADSL in some cases INTERNAL NETWORK Determines the range of internal IP addresses defined in the equipment 4 in the case of SYSTEL IP 12 or 2 for SYSTEL IP 4 They are automatically determined but in case of interference with other IP addresses within the network it can be manually established When done automatically all four or two IP addresses will be computed from the LAN IP by means of an algorithm guaranteeing that IP addresses are not repeated for different Systel IP units If the manual mode is selected the IP from which the rest of addresses are calculated must be specified in the IP Subnet field WARNING It is recommended that INTERNAL NETWORK is set as Auto We should use the Manual mode for assigning IP s only if the automatic IP generation creates any IP addresses colliding with any other equipment within the network Also none of them must be equal to the WAN address 25 AEQ SYSTEL IP SA P L Systel IP Setup e gt Systel IP Units foal Systel IP List Ta El ze SYSTEL IP 1 PN PS RP A Ki E O ds TE 8 Auto Discover Add Delete Edit VOIP Test Adjust gain Firmware Sync configuration w j ier device Teme Oe ET os c20905702774 SYSTELIP1 IP12 a Sdaef9bb066c SYSTELIP2 1P12 ee wi O Lo SYSTEL IP4 SMALL STUDIO 1P4 s aeai Sg Model IP 12 IP 4 be LAN DIGITAL AUDIO E IP 172 26 38 100 in
101. on when the system is shared by several up to 4 studios In most cases a simple 4 pair wiring loom will carry the analog or digital audios between the system and each studio AEQ can provide these looms in bobbins or finished with connectors If you already have enough radial wiring between the central control and the studios we can also make the installation easier by providing the termination between the system and the radial cabling by means of the wiring Accessory FR CAB INP DB15 male connector connected to four balanced and shielded pairs 6m long without termination to ease the wiring of 2 audio inputs and 2 audio outputs in SYSTEL IP 12 Max 6 per units per system are required 17 AEQ SYSTEL IP SA 3 SYSTEM INSTALLATION 3 1 Control elements 3 1 1 Configuration PC and control web server The system software comprises two Microsoft Windows compatible applications On one hand a setup utility that allows the user to define studios programs assign circuits to studios and control the agenda among other functionalities On the other hand a web server that provides service to the producers and studio operators control terminals They will only need to use a web browser to operate the system These two applications make use of the same shared database file so it is most reasonable to install them on the same machine There is no need to use a dedicated PC for Systel IP as both pieces of software can be installed
102. ons for one two and up to 6 faders A call queue is established on each fader E AEQ SYSTEL IP 12 o When two or more faders exist one of them can be locked as VIP pink color to make it stable on air meanwhile the rest of calls give pass to the following one in the rest of faders JONATHAN PAYO o A Zus Mr Jonathan Payo is a leader of Gypsy O a community He Il answer listener s ON AIR 04 18 AN LEVEL AEQ SYSTEL IP The different variables are filled in by default or can be preconfigured by the Windows Mem Vleit Setup application and can be called from gt the menu option Change Program amp Studio Auto Answer manaset Leve From an operative point of view there are sub variations depending on the work role of each person involved in the system operation For example producer controller and presenter Their visualization and operation rights are preconfigured although they can be changed using the Setup software 5 4 Control screen right common area general controls Change Program amp Studio 5 4 1 Call queue control window It is possible to configure the audio outputs and application operative logic within the Setup Software in order to use 1 2 or up to 6 faders where the queue ordered calls will be distributed Once calls are classified the operator will queue them By clicking on the red button with an arrow on it the next call in the queue will
103. operator handset is to be connected it must be plugged in the 4 pin RJ9 handset connector provided by the SYSTEL IP HS Physically each pin carries the signals described below Pin 4 Pin 5 BLUE BLUE amp WHITE Pin 3 Pin 6 WHITE amp GREEN GREEN Pin 2 Pin ORANGE WHITE amp BROWN Pin 1 Pin 8 WHITESORANGE BROWN RJ45 connector pinout 13 AEQ SYSTEL IP Pin 1 48V DC Pin 5 SPK Pin 2 48V DC Pin 6 MIC Pin 3 MIC Pin 7 CONTROL Pin 4 SPK Pin 8 OV DC Shield OV DC Highlights The use of shielded cable is mandatory and the shield must be properly connected to provide adequate power supply and interference suppression The handsets can be installed up to 300m far from the SYSTEL by using CAT5E or superior cabling Two LEDs can be found besides each connector The yellow one indicates that there is power supply and the green one indicates that a SYSTEL IP HS handset is connected at the other end of the wire Instead of using handsets 1 to 4 IP phones can be connected to the switch in the LAN port In this case each IP phone will take one of the twelve available IP lines away 2 2 2 4 Digital inputs and outputs o The connectors used are DB15 female with the following pinout S 1 00000000 0000000 15 g Identification of the DB15 connector DIGITAL I O 1 2 Pin 1 AES1 IN Pin 9 AES1 IN Pin 2 GND Pin 10 GND Pin 3 AES1 OUT Pin 11 AES1 OUT Pin 4 GND
104. os GC Unnumbered 83 AEQ SYSTEL IP IP configuration using a serial cable If you don t have a DHCP server in your network a new IP address can still be configured In order to do this just connect the serial ports of the gateway and PC using the cable provided by Patton and then run any terminal application Windows 7 doesn t provide one but you can download this free one for instance Real Term from http realterm sourceforge net e Set baud rate to 9600 No Parity None no flow control e The default login is user administrator with an empty password After that type in the following sequence of commands at the prompt in order to configure the IP address 192 168 0 175 as an example login administrator password SN4112 JO EUl gt enable SN4112 JO EUFconfigure SN4112 JO EUI cfg context ip router SN4112 JO EUI ctx ip router interface eth0 SN4112 JO EUI if ip ethO ipaddress 192 168 0 175 255 255 255 0 SN4112 JO EUI f ip ethO copy running config startup config SN4112 JO EUI f ip ethO Now you are ready to continue with the additional configuration either using text based telnet interfaces or via web Step 2 Generation of a configuration file AEQ can provide a configuration file that you can use as a Starting point to change your particular data yourself The example file presented in this manual configures two FXO lines in the SN4112 as independent SIP lines with UDP ports 5060 and 5061 respec
105. oupled GPI 4 GPO Main characteristics Universal power supply 100 240 V 50 60 Hz 25 VA Quiet operation natural convection cooling Weight 3 5 Kg 7 7 lbs Width 482 mm 19 1U rack height 44 mm 1 75 Depth 170 mm 6 7 SYSTEL IP 12 Engine for 12 IP lines Inputs and outputs DB15 female type multiple pole connectors two inputs and two outputs per connector 4 SYSTEL IP HS handset RJ45 connectors 8 analog balanced inputs 8 analog balanced outputs 4 digital AES EBU AES3 or SPDIF inputs 4 digital AES EBU AES3 or SPDIF outputs 1 IP port WAN for 12 VoIP lines 1 IP port LAN for control 3 DB15 connectors each one includes 4 optocoupled GPI 4 GPO Main characteristics Universal power supply 100 240 V 50 60 Hz 50 VA Quiet operation natural convection cooling Weight 5 Kg 11 Ibs Width 482 mm 19 2U rack height 89 mm 3 5 Depth 330 mm 13 SYSTEL IP HS Remote powered preamplifier Handset Includes 48V remote powered preamplifier with electret micro powered output RJ45 input connector for dedicated Cat 5 or better cable RJ9 connector for included microtelephone or standard operator microtelephone electret microphone Dimensions and weight Weight 0 5 Kg 1 1 lbs Width 85 mm 3 33 Height 44 mm 1 757 Depth 220 mm 8 66 SYSTEL IP ST Support for 10 Ipad Adequate for most 10 Tablets iPad Homologated fo
106. pe localhost 288 The login screen will show up AEQ SYSTELIP Account login Please enter your username and password P to login By the fault the user name is ADMIN with an empty PASSWORD You should add a password to the Administrator user ADMIN and create users and user groups depending on the roles of people in the station As an example some users and user groups are already created all of them without password Once you have been given access the system will inform about the active Systel system for the studio we are now and allows us to choose the studio program and intercom names 595 AEQ SYSTEL I P AEQ SYSTELIP Device SYSTEL IP1 172 26 38 100 Studio STUDIO_1 Program PROGRAM_1 Intercom HANDSET 1 A i Lines 8 Studio Ctrl Y If more than 30 seconds elapse before entering the studio program and intercom name the browser will disconnect from the system and it must be re opened Make sure to choose the intercom actually associated to the work place where the application is being open otherwise you won t be able to communicate with the studio partners If the system doesn t allow opening of the requested program this may be because another studio is already using the lines we are specifying We can then go to Lines amp Studio Ctrl and close the program using those lines that could have been left open inadvertently Usar Stud Program Intercom ADMINS ta Ole STUDER 1 OCH ah PRICHAM R
107. ple SYSTEL IP1 SYSTEL IP2 and SYSTEL IP 4 SMALL STUDIO Add button has been clicked while SYSTEL IP1 was selected and now we can create a 4 studio that we will name STUDIO 4 The right side of the screen shows the fields corresponding to the wiring assigned to the studio highlighted in the left list Name Mode Input Name Output Name and Pose In case that the selected studio has resources that are pending configuration warning messages will appear in that area AEQ SYSTEL IP 35 In that area we can create delete and modify the wiring to the corresponding studio In the example STUDIO 1 DIGITAL WIRED corresponding to the SYSTEL IP1 unit contains seven circuits Output Name DIGITAL 1L QUEUE DIGITAL 1L 1 DIGITAL 1R QUEUE DIGITAL 1R 2 DIGITAL 1L FEEDBACK DIGITAL 1L HANDSET 1 INTERCOM HANDSET 1 HANDSET 1 HANDSET 2 INTERCOM HANDSET 2 HANDSET 2 IP HAND 2 INTERCOM IP HANDSET 2 IP HANDSET 2 IP HAND 3 INTERCOM IP HANDSET 3 IP HANDSET 3 In order to add new resources to the selected studio the Add button should be used This will show a new window up that shows us all the options to assign resources Resource AAA ele Ee QUEUE from pcia Console Fader INTERCOM Mame Position 6 Input Name Output Name ANALOG 1 QUEUE ANALOG 1 1 ANALOG 2 QUEUE ANALOG 2 2 ANALOG 3 QUEUE ANALOG 3 3 ANALOG 4 QUEUE ANALOG 4 4 ANALOG 5 QUEUE ANALOG 5 A ANALOG 7 INTERCOM ANALOG 7 ANAL
108. put 1 No Output 1 No Security Level t2 No Output 2 No aai Mask 255 255 0 0 se e Input 3 No Output 3 No 2 WE 1 2 26 1 1 Input 4 No Output 4 No S DNS 0 0 0 0 External Sync AES 11 Users WAN for VOIP only INTERNAL NETWORK Active Auto 0 EE IP 192 168 0 21 Manual Mask 255 255 255 0 Sa ar aT L Gateway 0 0 0 0 Mask 255 255 255 0 Phonebook DNS 0 0 0 0 Computer IP addresses E ETS 172 26 3 2 255 255 0 0 172 31 3 222 255 255 0 0 O 192 168 0 11 255 255 255 0 192 168 7 97 255 255 255 0 amp ADMIN ADMINISTRATOR BA Version 1 0 1 20 20 01 2015 14 05 42 You can add new units manually or automatically from the Systel IP Units window as well as modify or delete existing equipment In order to add a new System IP system manually use the Add button a new window will appear that allows you to create a system from the scratch Add P lu Systel IP Setup Se Systel IP Units a Systel IP List ee El ze SYSTEL IP 1 eh SKI y A O Se E O ys e Auto Discover Add Delete Edit VOIP Test Adjust gain Firmware Sync configuration H H oe c2090570277d SYSTELIP1 Sdaef9bb066c SYSTELIP2 1P12 ee Ga O La SYSTEL IP4 SMALL STUDIO 1P4 VOIP Model 09 IP 12 IP 4 LAN DIGITAL AUDIO E IP Input 1 No Output 1 No Security Level el 2s 255 0 0 Input 2 No Output 2 No Input 3 No Output 3 No e Gateway 0 0 0 0 Input 4 No Output 4 No LA MEN 0 0 0
109. r no Y Make Call Without Reg Register Expires 3600 Ans Call Without Reg Display Name Cisco Phone User ID Password S Use Auth ID Auth ID Undo All Changes Submit All Changes 2009 Cisco Systems Inc All Rights Reserved SPA502G IP Phone In order to configure the telephone s user extension data check next section Subscriber Information The parameters to be configured are Display Name Handset name for descriptive purposes only User ID Name or user name to assign to the telephone It must match the User Name configured for the IP Handset in the Systel IP database Password It is not necessary in case of registering is in a Systel IP Use Auth ID the usual value to fill in here is No Auth ID leave blank if the previous field was filled with No DTMF Tx method select INFO option Last to make the telephone store the changed settings and reboot using them click on the Submit All Changes button 81 AEQ SYSTEL IP APPENDIX C INSTALLING AND CONFIGURING A GATEWAY For example Patton SN4112 2FXO Introduction AEQ SYSTEL can communicate with PSTN lines through gateways that translate both the call control signaling and the audio from Telephone Company to IP messages and streams As SYSTEL uses standard SIP messaging for cal control and de facto standard for audio transmission in VoIP is RTP standard gateways can be used Most of them will work p
110. r Apple Ipad 2 and Ipad 3 Minimum tablet size 22x15 5 cm 8 66 x 6 1 Maximum tablet size 28x18 5 cm 11 x 7 29 Includes a support for SYSTEL IP HS in the left side Adjustable tablet tilt Nov 2014 Characteristics subject to evolutive changes Download the latest version from www aeq es www aeq eu or www aeqbroadcast com 68 AEQ SYSTEL IP 7 A E Q WARRANTY AEQ guarantees that this product has been designed and manufactured under a Quality Assessment System Thus we guarantee that all the necessary test protocols have been followed in order to ensure the correct functionality and that the specified technical characteristics are met Both the general design and manufacturing protocols as well as those particular to this unit are properly documented 1 This warranty doesn t exclude or limit any customer s legally recognized right 2 The warranty period extends for twelve months from the date of purchase by the first customer In order to make use of this warranty the customer is required to inform of the problem to an authorized dealer any AEQ sales office or AEQ technical support department within thirty days from its appearance and within the warranty period as well as to provide a copy of the purchase invoice and product serial number At the same time AEQ Technical Assistance department s previous and express acceptance is required for the substitution or repair of the product in application of the present
111. recorder A 2 Installing the Asterisk PBX It is strongly recommended that the latest official stable version is installed among the many distributions based on Asterisk There are free and also paid versions The latter basically tend to include a PC with the installed PBX and technical support During the redaction of this manual we have based on the installation of AsteriskNOW version 3 0 a free distribution including Asterisk v11 2 1 the free graphical user interface FreePBX in its version v2 11 The installation process is quite simple Requirements e A PC with Internet connection and a CD DVD recorder to prepare the installation disk 70 AEQ SYSTEL IP SA A PC to install the PBX on Its hardware must be Linux compatible most standard configurations are Some PCs ready to host Asterisk PBX are available in the market If you still want to choose your own system we recommend that the investment is more focused on robustness than in performance For example with a good quality power supply good cooling a 4U rack unit with a good fan is usually enough and a good maintenance periodic fan filter cleaning basically a system can operate several years uninterruptedly Other desirable features include Removable SATA hard disks to ease backups Two network cards in case we want to separate the access to IP telephony WAN network to the control and setup IP network LAN network mainly due to security reasons
112. reens defining the number of lines per program operation with one or several up to six call queues Configure SIP for the communication with the IP PBX proxy gateway FXO external IP telephony provider via Internet or internal within a LAN or WAN Configure multi equipment operation shares resources between the different equipment within a network 4 1 Preparing a computer for setup and control of the system Starting the application Connect a computer with Systel Setup installed to the network SystellP Setup Click on the icon The start up screen will appear By default the user is set to ADMIN and the password is empty The user is encouraged to change it as soon as he is familiar with the application After the application is open just click on the first tab at the left and go to the SYSTEL IP Units option Check that the autodiscovery has found your SYSTEL IP unit or units if you have more than one in the same installation Accept on the discovered system and if you don t find it please check that the network parameters of both the unit and the computer belong to the same network having in mind that the default IP address of the unit is 172 26 36 250 in the case of Systel IP 4 and 172 26 35 250 in the case of Systel IP 12 E Systel IP Setup Systel IP List KS a Auto Discover 23 AEQ SYSTEL IP 4 2 SYSTEL IP SETUP Description of the screens in the Setup application After logging in
113. ronments are Windows by Microsoft and iOS for the iPad by Apple 3 1 2 1 Installation and initial configuration of the control client in Windows Google Chrome web browser is homologated for use with Microsoft Windows Vista Windows 7 and Windows 8 lt can be downloaded at http www google com intl es_es chrome In order to access the control screen the IP address and port 288 by default of the computer where the web server is installed must be typed in the URL bar of the browser for example http 172 26 5 32 288 It is recommended that the application is run maximized on the screen in order to have the maximum available space Use F11 for this purpose 3 1 2 2 Installation and initial configuration of the control client in OS The Safari Web browser that comes preinstalled in all iPad devices is homologated for Systel control In order to access the control screen the IP address and port 288 by default of the device where the web server is installed must be typed in the URL bar of the browser for example http 172 26 5 32 288 The following steps must be followed in order to use the iPad browser exclusively for SYSTEL IP control client 1 Disable multitask gestures in order to avoid that the application disappears when touching the screen with several fingers at the same time Settings gt General gt Multitask Gestures gt Disable 2 Disable automatic screen lock so it is not disabled when not used for
114. roperly however we recommend the devices made by Patton This document describes how to use a Patton gateway with SYSTEL IP Have in mind that no gateways are required when the telephony service is a SIP trunking over IP Compatible models Patton gives its family of VoIP gateways SmartNodes and are readily available internationally and also from AEQ though our dealership network Next a list of specific Patton devices for the different telecom services is provided e SN4112 JO EUI 2 analog FXO ports Configuration example http patton com products product_detail asp id 51 e SN4940 1E24V EUI 1 T1 E1 PRI port http patton com products product_detail asp id 437 e SN4634 3BIS EUI 3 ISDN BRI ports http patton com products product_detail asp id 329 e N4912 JO RUI 12 FXS VoIP IAD http patton com products product_detail asp id 364 These models are the specific units recommended by AEQ Other members of this family may be compatible but its operation should be asserted to guarantee the interoperability Configuration The following instructions example apply to Patton model no SN4112 firmware version R5 2 2009 01 14 They should be essentially the same for other Patton gateways Gateway parameters to be configured e Gateway device IP address e For analog gateways such as Patton model no SN4112 tones times etc for the telephone lines in use Usually these details are clarified once the co
115. rsal off Version 1 0 1 20 20 01 2015 14 05 42 If different parameters are required for some of the Systel IP units then the Customize Units option must be used This function allows unlocking the parameters of one unit and defining particular values for it In order to correctly configure these parameters that are provided pre configured from factory requires some knowledge of IP telephony If changes are required please consult your local support on IP telephony first fj Syste IP Setup gt fal Systel IP Units v T Studio Wiring mom a Security Level a Users Programs Lig VOIP Common SIP Parameters SYSTEL IP 1 Audio Codecs Order Audio Codecs a 6722 ae JS Sa G722 G711U de esa MN caw G726 32 G726 32 SIP Server Host SIP Server Port 5060 SIP Server Host m S Registry expiration 3600 Seconds SIP Server Port 5060 Outbound Proxy Host Register Yes 0 No dos p Outbound Proxy Port 5060 4 Registry expiration 3600 Seconds EEn soo Advanced settings RTP Ports 5004 e EJ SIP Local Port 5060 Call Transfer DTMF Tones 2 EA RTP Ports 5004 Buffer Size 20 Milliseconds Close Customize Units Call Transfer DTMF Tones 2 Packet Size 20 Milliseconds Buffer Size 20 Miliseconds os dp Packet Size 20 Miliseconds NAT Traversal Doff auto D
116. rt Internal IP 10 38 120 0 10 38 120 0 First RTP local Port Internal Mask 255 255 255 0 255 255 255 0 Last RTP local Port Nat Mode Public Address Keepalive Rtp TOS AEQ SYSTEL IP This window displays IP address and the name of Systel IP system we have connected with At the left Network area we can check all the network parameters we can configure under the Systel IP Units option The values actually set up in the Systel IP we have connected to Systel column that are different to what s stored in the software database Setup BBDD column are displayed in red At the right VOIP area all the voice traffic configuration parameters are displayed The values actually set up in the Systel IP we have connected to Systel column that are different to what s stored in the software database BBDD column are displayed in red In order to send the parameters selected by the check boxes from the Systel IP Setup software database to the selected Systel IP system we will use the Send Configuration button Send configuration BBDD gt SystellP E l E gt EN In order to read the values selected by the check boxes from the Systel IP currently connected and copy them to the Systel IP Software database we will use the Update Setup button Update Setup SystellP gt BBDD a If each and every value in the database matches those in the system the verification checkbox
117. runks should appear in this screen under the FreePBX Connections text FreePBX Connections At this point the only thing left is to define the routing rules for the outgoing calls so they are made through the provider and the routing rules for the incoming calls coming through it A 3 3 1 Incoming calls routing configuration A set of routing rules must be defined for calls coming from outside so the PBX knows where to route each one In this case as an example we will route calls coming through netelip provider to a special extension in Systel IP re W Yf Admin Applications Connectivity e Reports e Settings e Logout admin Add Incoming Route poss Op abe All DIDs toggle sort User DIDs General DIDs dd incoming Route Unused Dis FromPSTN Description 3736884562 any CID s FromSIPAEQ DID Number J 999999999 any CID Caller Number CID Priority Route Alert Info CID name prefix Music On Hold Default v Signal RINGING O Pause Before Answer Privacy Manager No v Set Destination choose one v Submit Clear Destination amp Submit 79 AEQ SYSTEL IP We will then proceed to select the Inbound Routes in the Connectivity menu The inbound routes edition screen will appear These are the fields to fill in Description DesdeNETELIP DID Number lt telephone number assigned to us by netelip gt Set Destination Extensions We consider that it is a good idea to present the
118. runtime installed for the right operation of the web server Welcome to the FMSoft uniGUT Beta Sencha Ext JS runtime confirmation is required Setup Wizard t This vd instal FMSoft wriGUI Bela Serx a Ext 25 runtime v0 93 0 936 on your computer It i recommended that you dose al other applicators before Lona Okk Next to comtinve ur Cancel to exi Setup AEQ SYSTEL IP 3 1 2 Control terminals The control terminals can consist in one two or three elements e Aweb browser based application essential element of the Systel IP that runs on a PC tablet or IPad regardless of operating system e A phone handset that incorporates a pre amplifier allowing to place it over 300 meters 327 yds from the equipment rack highly recommended e A support for a tablet and handset optional If the control terminals are mounted on tablet and that most probably won t have an Ethernet connector these can easily connect through WiFi router to the Web server control PC If the installation is very large it may be necessary to install several WiFi routers amplifiers or repeaters in order to guarantee proper coverage to all WiFi linked control terminals The control application works as a client of the server program within a web browser so a proper web browser must be installed that is adequate for the operating system of the PC or tablet used as a client At the moment of writing this manual the homologated envi
119. ry to unity gain 0 dB The green color in the display indicates that the adjusted level corresponds to the basic adjustment It may be necessary to alter it again increasing or decreasing the gain When the value is modified it turns to red indicating that this corresponds to a particular or momentary adjustment If for special reasons we notice that there is a permanent gain mismatch in the line to get a correct level we can click on the display when the correct value is visible and it will be validated and turn to green This way the adjustment is stored and will remain when another call is initiated in the same line This way If we have adjusted a line with for example 5dB gain if for some reason we need to apply a lower gain say OdB in a subsequent call the display will show 0dB in red warning that this is not the permanent adjustment for this line When clicking on the AGC button the input gains display window will show an AUTO legend while the level control fader disappears as it becomes automatic This way the system will automatically adapt itself so the incoming levels for different calls are adjusted to the nominal operating level The AGC adjustment is performed by means of the setup software In order to leave the line LEVEL adjustment screen click on Back button Note that there are two small tri color LEDs at the left of each line window indicating the presence of audio and its level allo
120. s Includes voice and control IP connectors 8 analog inputs and outputs 4 digital inputs and 4 digital outputs 4 handset ports 12 GPI and 12 GPO This should be enough to cover 4 studios Both units behave like multi line IP Phones with SIP signalling protocol Compatible with Asterisk PBX SIP Trunking and virtual PBX Analogue and ISDN lines supported through gateways Encryption algorithms include the proper ones in telephony G726 G729 and low bit rate G711 with higher quality G722 also incorporate coding with extended bandwidth to 7 kHz which characterize them as HD and makes them compatible with N ACIP AudioCodecs and SIP Phones Any AEQ Phoenix AudioCodec and most PC telephony software They are provided with configuration software which can be run in several computers on a single system web server and web client for an unlimited number of terminals IMPORTANT NOTE The default IP address configured in the units as they leave the Factory iS 172 26 36 250 in the case of Systel IP 4 and 172 26 35 250 in the case of Systel IP 12 AEQ SYSTEL IP 1 3 Functional specifications GENERAL FEATURES Operating Features SIP communications protocol compatible with VoIP trunkings Asterisk PBX SIP Phones such as Phoenix Pocket or Phoenix Lite N ACIP compliant Audiocodecs such as Phoenix Mercury Phoenix Studio Phoenix Venus or Phoenix Mobile and POTS ISDN E1 and T1 FXO Based on non blanking digital switching matrix
121. s blue and the call is active it ah will be placed in the queue when the button is clicked But if the button is blue and the call is inactive a queue or fader will be pre assigned for that line By means of the setup program the number of input channels in the console that are connected to the Systel outputs is determined for each studio The system admits between 1 and 6 queues or SYSTEL outputs to each studio console When clicking on the QUEUE button when the line is inactive the queue selection window pops up with a key for each queue previously created for that particular studio by means of the setup application Select Queue Line 01 2 Select Queue Line 04 Close Close By clicking on a queue button Q1 to Q6 a queue can be pre assigned to a line what s reflected in the QUEUE button queue 2 in the example 64 AEQ SYSTEL IP QUEUE button operation when a call is established VIP queues If the QUEUE button is pressed during Select Queue Line07 Select Queue Line 06 an established call and two or more ey ey VIP queues have been defined the VIP button will appear in the queue selection window Close Close If we click on the VIP button in the Select Queue Line 07 Select Queue Line 06 queue selection screen and after that Q2 ey on a queue that line will be assigned to that queue with the VIP attribute denoted by a deep p
122. s workers VoIP phones The main device in IP telephony is the VoIP phone or VoIP terminal specifically designed for VoIP usage that makes possible the establishment of a communication using an IP network either using a local area network LAN or through Internet The VoIP phone converts and compresses the voice signal into data packets ready to be sent through the IP network instead of using a traditional telephone connection A VoIP phone uses to be a device physically very similar to a normal phone Peculiarities of SYSTEL IP as a VoIP phone SYSTEL IP basically consists on a set of high quality IP phones 4 or 12 depending on the model digitally connected to a digital audio summing and distributing matrix with analog digital and special handset audio interfaces It is controlled through a web server system designed and configured to be managed from client applications specially adapted for utmost productivity in the radio and television broadcast stations environment Softphone VoIP adapter and IP audiocodec An IP terminal can also be a software application available for PC or smartphone Softphone that interacts with microphones and headsets speakers AEQ presents three Softphones Phoenix PC Phoenix Pocket and Phoenix Lite able to operate with IP audiocodecs and capable of using the highest quality audio compression algorithms Connecting an analog phone to a Gateway FXS or analog phone adapter is an alternative to a
123. sed from the last change This counter is quite useful to know how long the person at the other end has been waiting in a given situation IDLE the line is inactive waiting IDLE DIALING an outgoing call has been dialed and the remote end is still not ringing In most cases this is a spurious status that is DIALING not even visible in the screen RINGING outgoing call remote end is ringing RINGING 00 02 BUSY remote end is busy NO ANSWER this status disappears immediately NO ANSWER ON AIR call is being broadcast ON AIR 04 26 INCOMING incoming call Systel telephone rings WAIT on hold call either attended or unattended usually listening the program signal QUEUE queued call waiting to be put on air HANDSET X call communicating with operator of handset X in the example the name is HANDSET i Error messages In exceptional situations the following errors states may appear BlackList caller is banned by the black list No response no SIP messages have been received as a reply to the call request either by the SIP server or the final destination Unauthorized Auth ID and or password are not correct Forbidden no permission for calling Not Found call destination doesn t exist Not Acceptable call destination cannot accept the call due to compatibility reasons usually related to audio coding algorithms Unsupported codec call destin
124. static w SO Qos 4 NAT Traversal off auto static Systels SYSTEL IP 1 8 ADMIN ADMINISTRATOR Version 1 0 1 20 20 01 2015 14 05 42 42 AEQ SYSTEL I P The left area of the screen Common SIP Parameters presents the parameters that can be configured regarding both IP audio and SIP signaling Audio Codecs Order This allows the user to specify the priority applied to choose an audio codec when a unit negotiates the coding with the other end of the call This way a coding scheme close to the user preferences will be selected G 722 coding is the one providing the best voice quality while G 729 uses the lowest bandwidth The rest of audio coding algorithms provide an intermediate compromise between quality and bandwidth and are included to improve compatibility with equipment from different manufacturers SIP Server Host Name or IP address of the SIP server The SIP server is an application or device from which we receive incoming calls and to which we will send outgoing ones It is so called because it uses SIP call signaling protocol Some examples are SIP PBX Asterisk or similar gateways between traditional phone systems analog ISDN GSM etc and SIP telephone systems and VoIP providers using SIP either through Internet or dedicated networks SIP server Port Port the SIP server listens to waiting for SIP packets The most commonly used port for SIP signaling is 5060 Register This parameter specifies
125. t through Internet A 3 Asterisk PBX configuration A 3 1 General Configuration Settings menu Several parameters must be configured in the Advanced Options menu SIP and IAX allow g722 amp alaw amp ulaw SIP and IAX disallow all Several parameters must also be configured in the voicemail options submenu Voicemail Admin format wav16 A 3 2 Extensions configuration Some SIP extensions must be configured in the PBX including both the corporative extensions and those associated to the Systel IP units Ke Admin e Applications e Connec tivity e Reports e Settings e Logout admin R Add Extension Add SIP Extension EN 1000 lt 1000 gt 6000 lt 6000 gt 6001 lt 6001 gt 6002 lt 6002 gt 6003 lt 6003 gt 6004 lt 6004 gt User Extension Display Name CID Num Alias SIP Alias o Callies Outbound CID Asterisk Dial Options Ttr LI Override Ring Time Default v Call Forward Ring Time Default v Outbound Concurrency Limit No Limit v Call Waiting Enable v Call Screening Disable y Pinless Dialing Disable v Emergency CID Queue State Detection Use State v Regarding the corporative ones an extension will be defined for each IP phone If required voicemails can be activated for those extensions so that a message can be left when the called person is not available at that moment It is not within the scope of this manual to thoroughly explain how to properly configure the voice mail syste
126. that take place during an Internet call are conversion of the analog voice signal to a digital signal and compression of that signal to Internet protocol IP for its transmission The inverse process is carried out in the receiving end in order to finally recover the original analog voice signal VoIP has been in the consumer and corporate markets for a long time but it hasn t been until very recently that its usage has been widespread thanks to a better coverage lower cost per bandwidth and the improvement in technologies that guarantee the availability of the service and a call quality equivalent to traditional phone Voice and data convergence with IP or VoIP telephony Network convergence is a topic very often commented referring to integration within the same network of all kinds of communication voice data video etc Voice and data convergence in IP telephony allows for the creation of applications with great added value for the companies which multiply the usage modes and functionalities so far offered by the conventional telephone networks Advantages of IP or VoIP telephony for corporations The benefits of VoIP telephony are all related to a reduction in communications costs IP telephony allows for a reduction in the call costs or infrastructure costs Besides allows for a continuous grow of the material without obsolescence problems productivity improvements and provide new functions with added value for the company and it
127. the power switch can be found at the right side of the unit s front panel 2 1 2 Back panel description and connections AEQ 107234 e gg SS Lem we gt a IP Y ae A YA _ emm Pee as E int Ri aad ES we TELEPHONE ON AUR mmm Se E A O NO NO e re 8 gt EN KS N DIGITAL or ANALOG O ANALOG AEQ SYSTEL IP Ga gt d 2 1 2 1 Ethernet ports LAN amp WAN a SYSTEL IP 4 includes two Ethernet ports By using the LAN port the unit can be connected via IP to a local network for its configuration and control from the application installed on a configuration PC and the web control server It can also be controlled from different web clients installed in computers or tablets By means of the WAN port the unit can receive calls from a SIP proxy Asterisk or similar from Gateways or from IP phone providers by means of SIP trunkings Up to 2 IP phones can also be connected to the WAN port in order to communicate off air with the interlocutors instead of using analog handsets Physically the connectors are RJ45 10 100 BT with the following pinout Pin 4 Pin 5 BLUE BLUE amp WHITE Pin 3 Pin 6 WHITE amp GREEN GREEN Pin 2 Pin ORANGE WHITESBROWN Fin 1 WHITESORANGE Pin 8 BROWN RJ45 connector pinout 2 1 2 2 General purpose inputs and outputs GPIO The connectors used are DB15 female with the following pinout S 1 G ELEELEEE GC LELEEL
128. tifiers or phonebook entries Send a visual and acoustical RING signal Display call identifier number for the incoming calls Answer incoming calls both manually and automatically Save new contacts to the phonebook Talk by using a micro earphone or micro headphone to the person at the other end of the line Place calls on hold while they can listen the program Place calls on air so they can contribute to the program Send the signal to any of the PFL pre fader listen circuits defined in the studio in order to listen to the talker at the phone without putting him her on air end the signal to any of the Auxiliary circuits defined in the studio in order to make recordings on a device or make a party line with the lines connected to a same Auxiliary circuit Change the input and return levels to any of the studio telephone lines Display the status of any telephone line and where it is being routed to Distinguish between the producer operator and presenter roles label calls chat between the different controllers assigned to a program 5 2 How to access In order to access the control terminal the IP address and port 288 by default of the computer where the web server is installed must be typed in the URL bar of the browser for example http 172 26 5 32 288 If the server is installed the same machine where the browser is running most likely if there is only a single studio you can ty
129. tively This way a line can be assigned to a studio and the second line to a different one The labels in the file that should be substituted by your system s particular values are e IPADDRESS gateway s IP address e NETWORKMASK gateway s subnet mask e SIPREMOTEIP IP address of the system that will receive the calls coming from the analog lines that is either a SYSTEL IP system or an Asterisk PBX e SIPREMOTEPORT SIP port of the system that will receive the calls coming from the analog lines that is either a SYSTEL IP system or an Asterisk PBX e SIPREMOTENUMBER1 extension number that the gateway will pass as call destination to the remote SIP unit either a SYSTEL IP or an Asterisk PBX when passing a call received from FXO 1 port e SIPREMOTENUMBER2 extension number that the gateway will pass as call destination to the remote SIP unit either a SYSTEL IP or an Asterisk PBX when passing a call received from FXO 2 port ee e S5N4112 J0 EUT F R5 2 2009 01 14 H323 SIP FXS FXO 1970 01 01T02 09 36 SN OOAOBAO917B8 Generated configuration file cli version 3 20 webserver port 80 language en 84 AEQ SYSTEL IP sntp client sntp client server primary 129 132 2 21 port 123 version 4 system ic voice U low bitrate codec g729 profile ppp default profile tone set default profile voip default codec 1 g llalaw64k rx length 20
130. ts e Settings e Logout admin Module Administration Check Online Upload modules Reset Process Module Version Publisher Admin Custom Applications 2 11 0 0 FreePBX Enabled Digium Addons 2 11 0 3 Schmoozecom com Enabled Feature Code Admin 2 11 0 0 FreePBX Enabled FreePBX ARI Framework 21100 FreePBX Enabled FreePBX Framework 2110 11 FreePBX Enabled Recordings 33 119 FreePBX Enabled Applications Announcements 21100 FreePBX Enabled Conferences 211 01 FreePBX Enabled Core 21109 FreePBX Enabled IVR 21103 FreePBX Enabled Info Services 2 11 0 1 FreePBX Enabled Misc Applications 21100 FreePBX Enabled Parking Lot 2 11 0 12 FreePBX Enabled Queues 2 11 0 11 FreePBX Enabled Ring Groups 21102 FreePBX Enabled Text To Speech 211 07 Schmoozecom com Enabled Connectivity 12 AEQ SYSTEL IP e The language for voices installed by default is English You can find voice packets in Spanish and other languages in Internet For example voices in Spanish can be downloaded for free at http www voipnovatos es voces The packet consists on a compressed file including the audio files Those files need to be copied to the PBX hard disk in the var lib asterisk sounds folder e The display keyboard and mouse can be disconnected from now on WARNING In order to provide the PBX with a higher degree of safety it is recommended that all passwords are changed and that the web control panel is made visible only within the private LAN no
131. tx length 20 codec 2 g llulaw64k rx length 20 tx length 20 profile pstn default profile sip default profile aaa default method 1 local method 2 none context ip router interface eth0 ipaddress SS IPADDRESSSSS SSSSNETWORKMASKS S tcp adjust mss rx mtu tcp adjust mss tx mtu context cs switch digit collection timeout 2 interface sip IF_SIP_O bind context sip gateway GW_SIP_0 route call dest intertace IF FXO 00 remote SS SSSIPREMOTEIPS S S SS SSSIPREMOTEPORTSSSS address translation outgoing call to header user part fix SSSSSIPREMOTENUMBERI1SS S host part remote interface sip IF_SIP_1 bind context sip gateway GW_SIP_1 Foute call dest intertace IF FXO 01 remote 172 26 35 10 5060 address translation outgoing call to header user part fix SSSSSIPREMOTENUMBER2S S host part remote interface xo TE FXO 00 route all dest lt interrace IF SIP H disconnect signal battery reversal disconnect signal loop break disconnect signal busy tone interface fxo IF FXO 01 route call dest interface IF SIP 1 disconnect signal battery reversal disconnect signal loop break disconnect signal busy tone AEQ SYSTEL IP 85 context cs switch no shutdown context sip gateway GW_SIP_0 interface IF_GWSIP_O bind interface eth0 context router port 5060 context sip gateway GW_SIP_0 no shutdown context sip gateway GW_SIP_1 interface IF GWSIP 1 bind interface eth0 context router port 5061 context sip gateway GW_SIP_1
132. type of communications Further it significantly improves the audio quality increases the flexibility and integration with already existing telephone systems at the station The investment required is very small and will be amortized very rapidly through simple saving of costs Business telephone systems are rapidly migrating to VoIP technology integrating IP switchboards Asterisk or similar type distributed or virtual allowing access to new alternative telecommunication service providers At the same time telephony or call in systems for broadcast applications have so far been an isolated island with important operational costs and stagnant technology SYSTEL IP allows for the integration of broadcast telephony with the existing corporative IP based PBX thus avoiding the need to keep conventional phone lines exclusively for broadcast use 1 2 Description of the basic system SYSTEL IP 4 SYSTEL IP 12 The heart of the system is a 19 rack unit in two versions e Systel IP 4 one unit high rack for 4 simultaneous IP phone lines Includes voice and control IP connectors 2 analog inputs and 2 analog outputs 1 input and 1 output both selectable as analog or digital a handset port another port which can be used as the second handset or the second analog input and output 4 GPI and 4 GPO This should be enough to provide service to one or two studios e Systel IP 12 two units high rack format for 12 simultaneous IP phone line
133. uarantee that the selected position is held use the lever in the hinge itself while it is pressed towards its axis You will notice that this way you can adjust the tightness of the hinge and hence its ability to hold a position once set If you need to d change inclination again just loose the hinge using the lever to tilt the tablet change the angle and then lock it again 21 AEQ SYSTEL IP 3 1 2 5 Installing IP phones as an alternative to the analog SYSTEL IP HS handset Instead of using handsets Systel IP HS IP phones can be connected to the switch in the WAN port In this case each IP phone will take one of the available IP lines away A standard SIP phone can be used as a handset so the off air talking function is obtained Also the phone keyboard can be used to make and transfer calls and in certain situations the line can also be hanged up when the micro telephone is hung It must be installed in the same WAN network as the SYSTEL IP equipment registering it in the system itself or in an Asterisk PBX using the default 5070 port The SIP Phone model approved for this function is CISCO 303 Using the IP Handset It is similar to an analog handset with the following improvements which are available provided that the HANDSET status is configured in the software If the phone is hung it will ring and if it is not unhooked in a certain time it will change to the WAIT status When the phone is hung the HANDSET wil
134. ue 0 is recommended y RH Syste IP Setup VOIP Systel IP Units Common SIP Parameters SIP Extensions H Add ete H Studio Wiring l Audio Codecs Order Number JL Auth Id Auth Password Description G722 6001 La 6711 A 6000 E e G711U G726 32 Y Security Level SIP Server Host 9 SIP Server Port 5060 SIP Handsets E e i Register Yes No o Modify 9 Registry expiration 3600 Seconds stes ET Advanced settings SYSTEL IP 1 UserName Programs SIP Local Port 5060 Host RTP Ports 5004 EE Port 5060 gt fy HN Call Transfer DTMF Tones 2 EJ Auth Password bok Buffer Size 20 Miliseconds AAA d S Op Line Number 0 Packet Size 20 v Milliseconds Customize Units i Qos a 5 ee Handset Userame Host Port Auth Id NAT Traversal off auto static lt No data to display gt amp ADMIN El ADMINISTRATOR Version 1 0 1 20 20 01 2015 14 03 22 47 AEQ SYSTEL IP 4 2 4 Security Level ae This screen allows the creation and management of the user groups A user group is characterized by a set of restrictions to the rights giving access to some functions of the configuration and real time operation applications The admin level has all the access rights and cannot be edited or deleted All other user groups and security levels can be modified and or deleted 8 BH Systel IP Setup aoe
135. untry is selected e For ISDN gateways the used protocol e Number or text name to be shown in each line This is transferred to SYSTEL via a SIP message and is used to identify the calls and assign them to buttons as configured with SYSTEL SetUp software Patton gateways can be configured in several ways e Manually with a text based command line interface CLI using the gateway s user manual as a reference This method is not recommended due to its inherent complexity and because it is error prone e Manually using a web interface This is only recommended for true analog and SIP telephony experts e Obtaining a loadable file from AEQ If you can provide data such as your number of lines names etc we can provide a file to be loaded into the gateway in return 82 AEQ SYSTEL IP e Creating your own loadable from a model provided by AEQ or Patton More information about this option is provided below You can ask a pre configured gateway AEQ offers this service As a reference these are the links to Patton user manuals e Users Manual http www patton com manuals SCG_r57 paf e Quick start guide http www patton com manuals SN4520 SN4110 QS paf Step 1 Configuring the IP address The first step is to obtain an IP address in the gateway This can be done by any of the above described methods As any other IP device in order to give it another IP an initial temporary address must be known Fortunately this is a one
136. warranty As a consequence returns not meeting these conditions will not be accepted 3 AEQ will repair or substitute failed the hardware dongles which must be previously returned This includes the labor cost necessary to undertake this repair provided that the failure is caused by a defect in the materials design or manufacturing The repair will take place in the AEQ Technical Assistance Service installations This warranty doesn t include the shipping to the workshop nor the return 4 A warranty extension won t be provided for repaired or replaced in application of this warranty 5 This warranty won t be in force in the following cases use not according to what s indicated in the user manual violent handling exposure to humidity extreme thermal or environmental conditions or sudden changes in those conditions lightnings oxidation unauthorized modifications or connection unauthorized repairs or product openings misuse liquid or chemical products pouring 6 AEQ will be under no circumstance responsible for any kind of damages or harms direct or indirect derived from the use or the impossibility to use the product AEQ won t be responsible for any loss of information in discs that have been altered or are inaccurate nor any other accidental harm caused by the user or people handling the product 69 AEQ SYSTEL IP APPENDIX A INSTALLATION AND SETUP OF AN ASTERISK PBX FOR SYSTEL AND CONNECTION TO A SIP TRUNKING
137. will show up a list with all the Systel IP units found in the network At this very moment the user can decide whether to add one or more of the discovered units Add unit button or recognize that some of the discovered units correspond to already existing systems Link to existing unit button 4 2 1 1 Synchronize configurations a ADMIN ADMINISTRATOR Lal Version 1 0 1 20 20 01 2015 14 05 42 8 Link to existing unit Select SystellP c2090570277d 172 26 38 100 SYSTEL IP1 Sdaef9bb066c 172 26 38 110 SYSTEL IP2 172 26 38 111 SYSTEL IP4 SMALL STUDIO From the Systel IP Units main window we can connect to the different S registered Systel IP equipment After we select the desired one by clicking on the Sync Configuration button a new window will be accessed Sync configuration systet IP SYSTEL IP4 SMALL STUDIO Send configuration BBDD gt SystelIP Update Setup SystellP gt BBDD 255 255 255 0 255 255 0 0 Gateway 0 0 0 0 172 26 1 1 Dns 0 0 0 0 0 0 0 0 WAN Active False False WAN IP 0 0 0 0 0 0 0 0 WAN Gateway 0 0 0 0 0 0 0 0 Internal Auto Manual Auto Auto External Sync AES 11 LOCAL LOCAL Buffer Size Packet Time Proxy Host Proxy Port WAN Mask 255 255 255 0 255 255 255 0 E Registrar Host Registrar Port WAN Dns 0 0 0 0 0 0 0 0 Registry Expiration SIP Local Po
138. wing for a general view of each line input and output levels at a glance and adjustments can be made by going to the LEVEL screen of those lines presenting mismatches 66 AEQ SYSTEL IP 5 6 8 Partner name editable field operation The line name or URI indicator has 3 fields 1 Telephone number or URI 34916864492 2 Agenda entry name for it GYPSY INSTITUTE 3 Name given by the producer at that Mr JONATHAN PAYO moment If field 3 exists this is the one displayed When clicking on the right mid area of the field it changes to field 2 and when clicking again to field 1 cyclically If field 3 doesn t exist field 2 will appear by default type the temporary name in 5 6 9 Partner observations editable field operation Comments about the partner can be typed in Font size can be adjusted in the menu trying to find a compromise between readability and field capacity If the comments dont fit in the screen a scroll bar appears Edition mode is entered by clicking on the field 5 7 Using an IP phone instead of an analog handset as the SYSTEL IP handset The use of an IP phone allows the user to perform some operations on it concurrently with the control software for convenience In order to do this the HANDSET status must be activated by using the software Answering and hanging up a call If the telephone is hung it will ring when an incoming call is received If it is not
139. y programmed as Digital It is routed to the input connector of Channel 1 in the console which is configured to receive the left channel A console auxiliary output is sent to Input 1 of the Systel internally configured as Digital It will contain audio program without telephones FEEDBACK in Digital Input 1L HANDSET 1 is wired to provide Intercom to the studio operator Studio Name Name Mode imput Name Output Name Pos STUDIO SMALL 1 SYSTEL IP4 SMALL STUDIO ANALOG 3 QUEUE ANALOG 3 1 STUDIO SMALL 2 SYSTEL IP4 SMALL STUDIO ANALOG 3 FEEDBACK ANALOG 3 lt has one queue QUEUE 1 that is output by SYSTEL IP 4 through analog Output 3 It is routed to the input connector of Channel 1 in the console that is configured as mono A console auxiliary output is sent to Input 3 of the Systel internally configured as Digital It will contain audio program without telephones FEEDBACK in analog format HANDSET 2 is wired to provide Intercom to the studio operator If desired a 2 queue can be added in the analog studio through output 2 40 AEQ SYSTEL IP System used in a single studio Digital console Studio Name Name Mode imput Name Output Name Pos STUDIO SMALL 1 SE IP4 SMALL STUDIO DIGITAL 1L QUEUE DIGITAL 1L 1 DIGITAL 1R QUEUE DIGITAL 1R 2 DIGITAL 1L FEEDBACK DIGITAL 1L HANDSET 1 INTERCOM HANDSET 1 HANDSET 1 HANDSET 2 INTERCOM HANDSET 2 HANDSET 2 If the studio is digital we can have 2 handsets and 2 qu

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