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Manual II: Administrator`s Guide

1. Fig II 26 Extensions Management Edit Entry General Settings page Please Note Extensions cannot be detached from the line if the SIP Remote Extension service is enabled on it To detach the extension from the line disable the SIP Remote Extension service on the extension first Use Kickback checkbox enables the Kickback service on the extension for the blind call transfer When the extension transfers the call to the other extension and if there is no answer from the destination side the call will automatically get back to the extension who initiated the transfer instead of getting into the destination s voice mailbox or being disconnected Allow Call Relay enables the current extension to be used to access the Call Relay service in the QX IP PBX s Auto Attendant It is recommended to define a proper and non empty password when enabling this feature in order to protect the Call Relay service from an unauthenticated access GUI Login Allowed checkbox enables the current extension to be used to access the QX IP PBX via WEB interface by extension name and password 3pcc Click2Dial Access Allowed checkbox enables the current extension to be used with applications based on QX IP PBX 3PCC interface and QX IP PBX Click to Dial application With the Show on Public Directory checkbox enabled the details of the corresponding extension will be displayed in the User Settings table on the Main Page of the Extension s Web Management accessed b
2. Fig I1 224 IPSec Connection Wizard Add IPSec Connection page The next page of the wizard is IPSec Keying Properties which is used to select IPSec connection s security encryption settings Auto Keying requires the IKE Internet Key Exchange and ESP Encapsulated Security payload settings defined Encryption and _ IPSec Configuration Wizard Authentication parameters should be defined Go tack The Encryption drop down list offers the following standards for di selection BANES Encryption AES 128 bit Y Triple DES uses three DES encryptions on a single data E block with three different keys to achieve a higher security SCAR than is available from a single DES pass block cipher Encryption AES 2B 7 algorithm with 64 bit blocks and a 56 bit key ee AS AES 128 bit cryptography scheme is a symmetric block cipher which encrypts and decrypts 128 bit blocks of data AES 192 bit cryptography scheme is a symmetric block cipher which encrypts and decrypts 192 bit blocks of data e AES 256 bit cryptography scheme is a symmetric block cipher which encrypts and decrypts 256 bit blocks of data Fig II 225 IPSec Connection Wizard IPSec Keying Properties page The area Authentication offers the following parameters to be selected e SHA SHA1 Secure Hash Algorithm is a strong digest algorithm proposed by the US NIST National Institute of Standards and Technology agency as a Standard digest algori
3. 232 lt 20202 sip epygi com Attendant lt 20202 sip epygi com gt Diagnostic Call lt 20202 sip epygi com gt Attendant lt 20202 sip epygi com gt Diagnostic Call lt 20202 sip epygi com gt Armen Movsisyan lt Armen Movsisyan lt 093519806 i Callee 20530 sip epygi com 5060 Levon Dadayan lt 11380 gt 741200 192 168 0 209 5060 741500 192 168 0 209 5060 741500 192 168 0 209 5060 010273144 ims ucom am 5060 010273144 ims ucom am 5060 Levon Dadayan lt 11380 gt 010242532 ims ucom am 5060 010200090 ims ucom am 5060 010273916 ims ucom am 5060 010273916 ims ucom am 5060 055454177 ims ucom am 5060 0113168790440904 53 84 85 5060 010528467Qims ucom am 5060 01131687904409 4 53 84 85 5060 Ashkhen Barseghian lt 20231 gt Ashkhen Barseghian lt 20231 gt 20537 sip epygi com 5060 Ashkhen Barseghian lt 20231 gt Ashkhen Barseghian lt 20231 gt New recordings 715 Allrecordings 715 Date amp Time A 05 Aug 2014 11 51 19 05 Aug 2014 11 38 25 05 Aug 2014 10 35 49 05 Aug 2014 10 35 45 05 Aug 2014 10 34 20 05 Aug 2014 10 21 39 05 Aug 2014 10 17 34 05 Aug 2014 10 14 06 05 Aug 2014 09 53 51 04 Aug 2014 20 57 13 04 Aug 2014 20 06 13 04 Aug 2014 20 02 30 04 Aug 2014 20 00 19 04 Aug 2014 20 00 09 04 Aug 2014 19 56 19 04 Aug 2014 19 55 52 04 Aug 2014 19 54 07 04 Aug 2014 19 51 21 04 Aug 2014 19 42 04 04 Aug 2014 19 41 14 04 Aug 2014 19 39 25 5 hou
4. ae The presence of asterisks in a pattern Criterion 1 l l o The patterns without have a higher priority QX50 QX200 QX2000 SW Version 6 0 x 102 QX50 0X200 0X2000 Manual II Administrator s Guide O The total number of matching digits symbols inside and outside the braces brackets riterion The more matching digits a pattern contains the higher its priority The number of matching digits symbols outside the braces brackets Criterion 3 The more matching digits outside braces brackets a pattern contains the higher its priority Please Note This criterion is used only if several patterns take an equal but non zero value for Criterion 2 A A The total number of question marks inside and outside the braces brackets riterion The more question marks a pattern contains the higher its priority The number of question marks outside braces brackets Criterion 5 The more question marks outside braces brackets a pattern contains the higher its priority Please Note This criterion is used only if several patterns take an equal but non zero value for Criterion 4 The number of square brackets Criterion 6 The more brackets a pattern contains the higher its priority tee The number of braces Criterion 7 l l l S The more braces a pattern contains the higher its priority o The number of asterisks Criterion 8 l l o The fewer asterisks a pattern contains the hig
5. e The Available Calls Duration text field requires the maximum available duration of the calls in minutes placed with the selected routing rule Once the Available Calls Duration reaches the value defined here the current call will be disconnected without prior notice and no new call will be possible until this field is updated e The Expiration Renewal Date settings are used to configure the Expiration Date and Renewal Amount of the Available Calls Duration Expiration Renewal Date field provides selection between Periodic and Specific Date o The Periodic selection is used to define the expiration date of the allocated Available Calls Duration for the selected routing rule and has the following options e Daily Telephony Call Routing Call Recording NAT Traversa le Call Routing Local AAA Table Global Speed Dial SIP Tunnel Class of Service Call Routing Wizard O Go Back Date Time Rules Add Entry Monthly Avail v Annually Avail y Available days Available Time Period hh mm hh mm v 23 59 Month Month Month Day Day Day hh mm hh mm Call Routing Wizard O Go Back Routing Overall Calls Limitation Settings Add Entry Available Calls Duration 100 Expiration Renewal Date Fig II 142 Call Routing Wizard Routing Call Limitation Settings Edit Entry page e Weekly the preferred week start day should be selected for this option e Monthly the ca
6. QX50 0X200 0X2000 Manual II Administrator s Guide Call Routing Call Routing Local AAA Table Global Speed Dial SIP Tunnel Class of Service Call Routing Table Show Detailed View gt gt gt Hide disabled records Telephony Enable Disable Add Edit Duplicate Delete Move Up Move Down Move To So Number Patte ID State Destination Number Pattern Pattern Modification Call Settings Failover Reason s Local Authentication atras t de A mud Source Type UES URP Metric Description Caller ID Modification 1 Enabled 911 FXO None a PBX 10 Emergency Call port s Any Port 2 Enabled 97 FXO Any z PBX 10 Make PSTN Call port s Any Port 3 Enabled 8 SIP URP Yes 10 Make SIP call No sip epygi com Enabled 00 PBX URP No 10 Call To Attendant No Enabled PBX URP No 10 Call to Extensions No Enabled SIP URP No 10 192 168 74 117 5060 No NDS Number of Discarded Symbols Use Extension Settings RNSC Restrict the Number of Simultaneous Calls URP Use RTP Proxy AAA Authentication Authorization Accounting Fig II 137 Call Routing table brief preview Defining patterns in the Call Routing Table avoids registering QX IP PBX at the routing management server and gives you an option to establish a direct connection to the destination or to use a SIP server for call routing The alternating Show Detailed View and Show Brief View buttons are used to display entries in the Call Routing table in detailed and brief views
7. Attendant 00 Settings General for QX50 QX200 Attendant 00 Settings General for QX2000 Attendant 00 Settings Attendant Scenario Attendant 00 Settings SIP Attendant 00 Settings SIP Advanced Attendant 00 Settings Codecs Conference Management and Email Default Settings Universal Extension Recordings Extension Directory Receptionist Management ACD Management QX50 QX200 QX2000 SW Version 6 0 x QX50 0X200 0X2000 Manual II Administrator s Guide System Default Value G726 40 enabled iLBC G 722 G 722 1 H 263 H 263 and H 264 disabled Out of Band DTMF Transport enabled T 38 FAX enabled Pass Through FAX enabled Pass Through Modem disabled Force Self Codecs Preference for Inbound Calls disabled SRTP Policy Make unsecure calls accept anything Codecs G711u preferred G711a G729a enabled G726 16 G726 24 G726 32 G726 40 iLBC G 722 G 722 1 TDVC H 263 H 263 and H 264 disabled Out of Band DTMF Transport enabled T 38 FAX enabled Pass Through FAX enabled Pass Through Modem disabled Force Self Codecs Preference for Inbound Calls disabled SRTP Policy Make unsecure calls accept anything Display name Attendant FAX forwarding disabled Show on Public Directory enabled Percentage of System Memory 5 Display name Attendant FAX forwarding disabled Show on Public Directory enabled Percentage of System Memory 0 08 Scenario de
8. Call to Sales Department The Recording Type drop down list allows you to select whether the recording will start automatically as soon as the call is established or whether it will be activated manually by pressing the button on the phone during the call Fig II 157 Call Recording Settings Add Entry page The Maximum Recording Duration drop down list is used to select the maximum duration when the call between the defined caller and called parties will be recorded When the call recording duration expires it will be silently stopped while the call will stay active QX50 QX200 QX2000 SW Version 6 0 x 108 QX50 0X200 0X2000 Manual II Administrator s Guide The Recording To drop down list is for selecting the Recording Box extension Extensions Management to be used for storing the recordings The Description text field should contain some descriptive text related to recording rule Edit opens the Edit Entry page to modify the selected entry This page contains all the same components as the Add Entry page does NAT Traversal Settings The NAT Traversal Settings page is divided into separate pages used to configure General NAT Settings SIP RTP and STUN parameters for NAT and a page where the NAT Exclusion table may be filled General Settings The General Settings page consists of a manipulation radio buttons group to select the mode of the NAT Traversal usage for 7 the SIP traffic any incoming and outgoing SIP messages f
9. Edit Entry Extensions O Go Back e Redirection on Timeout this group allows automatic General Settings call redirection in case no action has been performed by iii e 99 the caller The group offers the following options Oo ba gt Defaut a SIP Advanced Settings Redirection on Timeout Enable Redirection on Timeout checkbox is used to A enable disable the automatic call redirection a Go To Codec Settings Call Type PBX Y Recurring Attendant Prompt Repetition Count text cao field indicates the number of Recurring Attendant Prompts to be consecutively played to the caller with no action from his her side When the Recurring Attendant Prompt is played the number of times indicated in this text field the call will be automatically redirected to the defined destination Attendant Welcome Message Y Enable Welcome Message Call Type drop down list includes possible incoming call types PBX PSTN SIP or Auto PBX selection means that the call will be redirected to the local extension SIP selection means that the call will be redirected to the SIP destination correspondingly PSTN selection means that the call will be redirected to the PSTN destination Auto Totus scenario te selection is used for undefined call types destination E independent on whether it is a PBX number SIP address A or PSTN number will be reached through Routing Choose File No file chosen Recurring Attendant Prompt Upload new Recurrin
10. L2TP Server 192 168 74 55 Fig ll 232 PPTP L2TP Connection Wizard for L2TP connection Page 2 The Start functional button initiates the selected connection s If it is a client connection then this button initiates a client activity of reaching the server The Start option is applicable for multiple connections selected at the same time The Stop functional button is used to stop the selected connection s Stopping the server connection will disconnect all connected clients and close the PPTP L2TP tunnel The Stop option is applicable for multiple connections selected at the same time PPTP Server Configurations The PPTP Server Configuration page is used to configure the PPTP server settings and offers the following components The PPTP Subnet text fields are used to enter the IP address range for the PPTP server and clients within the PPTP tunnel The value specified for the subnet mask is fixed to 24 to restrict the possible number of clients for the PPTP connection Please Note The first address specified in the PPTP Subnet will be assigned to the PPTP server others will be assigned to the clients The PPTP server subnet should be different from the L2TP server subnet otherwise a corresponding error message will appear The Authentication manipulation radio buttons are used to select the corresponding authentication protocol by which the client communicates with the server The MSCHAPv2 selection enables
11. Please Note The Overall Calling Time Limitation checkbox is not allowed for PBX PBX Voicemail and PBX Intercom destination types routing rules Set Tracing Debug Options on This Rule checkbox is used to switch events notification on the certain execution results of the corresponding routing rule When this checkbox is enabled the Call Routing Wizard Tracing Debug Options page will be displayed Require Authorization for Enabling Disabling checkbox is used to enable administrator s password authentication when enabler disabler keys are configured for the routing rule The service can be used locally from the handset see Feature Codes in Manual III Extension Users Guide or remotely from Auto Attendant see Auto Attendant Services in Manual III Extension Users Guide When this checkbox is selected administrator s password will be requested to enable disable the certain routing rule s If the administrator s password has been inserted incorrectly for 3 times no status changes will be applied to any of the routing record s even to those which have no authorization enabled Enabler Key and Disabler Key text fields request digit combination which should be dialed from the handset or Auto Attendant to enable or disable the certain routing rules in the Call Routing Table You can set the same Enabler Disabler Key for multiple routing rules the same key may be used as enabler for one routing rule and as disabler for another one this will allow
12. Signaling Type allows selecting the CAS signaling type Please Note R2 signaling compelled and non compelled can be used with an El interface both in User and Network modes QX IP PBX with E1 interface in the CAS mode detects the busy tone only in case of R2 compelled and non compelled both with and without ANI signaling types Force Update Timeslots checkbox can be optionally selected in order to apply new settings immediately This will force the timeslot s to be restarted and any active connection on the selected timeslot s will be interrupted Please Note QX does not support the Forward Digit selected on the CO when acting in the User mode with CAS Loop Start signaling type Get PSTN PBX Error Message checkbox enables notification message in case of outgoing calls to unreachable incorrect or non existent destination PE Interfaces El T1 Trunk ISDN Trunk PSTN Gateways CAS Signaling Wizard Signaling Type Settings Trunk 1 192 168 74 127 5060 Selected Timeslots 1 Allowed Call Type Both incoming and outgoing calls Signaling Type Loop Start Force Update Timeslot Get PSTN PBX Error Message Generate Progress Tone to PSTN PBX Y Enable Echo Cancellation Alternative Disconnection Mode Voice Establishment Procedure on call acceptance on channel selection on call ringing Generate Progress Tone to IP Fig IT 115 CAS Signaling Wizard Page 1 When Generate Progress Tone to PSTN P
13. The Download and Remove links appear only if a file has been uploaded previously The Download link is used to download the NTE uE message file to the PC and opens the file chooser window where the pe O AO saving location may be specified The Remove link is used to restore the default welcome message Incorrect number handling will be activated only in the following two cases 1 An attempt was made to call a non existent extension ion 2 An attempt was made to call a number not matching with any Destination Number Pattern in the Call Routing table Fig II 63 Create scenario Main menu page The User Input Options table is for configuring the action to be taken based on one of the following user choices o User Input e Any input other than in the list above 0 No input The user will press one of the following input options on the phone to activate the corresponding action The option can be selected after reaching the Auto Attendant Service and after the Welcome and or Recurring messages have been played The User Input table consists of the following functional buttons Add opens the Add Option page where the actions for previously unspecified inputs can be configured Add link opens the Add Option page where the actions for previously unspecified inputs can be configured Edit Scenario 00 MainMenu Add Option Extensions Edit link opens the Edit Option page where the actions of previously confi
14. collecting information from callers in the form of DTMF digits and based on that making the routing decision on delivering the call to proper Agent Group e Predefined ACD Agent Auto Attendant used for agent login logout and updating the current status of the agent from the phone To monitor ACD processes on the QX IP PBX Epygi provides a Statistics Monitoring and Reporting SMR application running on MS Windows PC SMSR doesn t require the 3PCC license see Feature Keys section to be installed on the QX IP PBX It displays the current status and statistics on Agent Groups and Agents builds the statistical reports and sends notifications and alerts to ACD supervisor administrator For more details and requests for this applications contact Epygi sales division www epygi com Agent Agent is the call center user answering the customers calls and reachable via QX IP PBX due to ACD To receive the calls agent needs to be logged into some Agent Group AG Agent is characterized by the agent ID password skills levels and termination phone number Agent can be logged into several agent groups at the same time and receive the calls distributed by those agent groups For easy login logout to all groups where the agent is subscribed agent should use the 83 feature code from the handset ACD allows the system administrator to define the set of skills adequate to call center profile and grade the professional capabilities of each agent a
15. only the unsecure calls will be generated and accepted e Try to establish secure calls accept anything system will try first to establish secure call but will fallback to unsecure call if party doesn t accept secure calls both secure and unsecure incoming calls will be accepted as requested by remote party with the preference given to establishing secure call e Make unsecure calls accept anything system will establish unsecure outgoing calls but both secure and unsecure incoming calls will be accepted as requested by remote party For bandwidth used by secure calls see Needed Bandwidth for IP Calls QX50 QX200 QX2000 SW Version 6 0 x 50 epygl Call Park and Directed Call Park Service The Call Park and Directed Call Park services are used to store a call on a specific number so that any other user on the system can retrieve it For example a user receives a call but wants to take it in a conference room where it is possible to speak privately Transferring the call to the conference room is not an option because the conference room it is transferred to might be in use or the user is unable to walk to the conference room in time to answer the call The user can use Call Park and Directed Call Park to place the call at a specific number and then retrieve when they reach the conference room To use the Call Park or the Directed Call Park features at least one Call Park extension should be created in the Extensions Management
16. 192 168 0 209 5060 7412109 192 168 0 209 5060 1 day 3 hour 31 min 38 sec 5 hour 30 min 20 sec 0 sec 3 1 hour 39 min 6 sec 3 1 hour 39 min 6 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec GUI Call Relay A password is empty GUI Call Relay A password is empty GUL Call Relay A password is empty GUI Call Relay GUI Call Relay 3pcc Click2Dial GUI Call Relay A password is empty GUI Call Relay GUL Call Relay d password is empty GUI Call Relay A password is empty Fig II 24 Extensions Management page 23 QX50 0X200 0X2000 Manual II Administrator s Guide The following columns are present in the table e Extension lists user or attendant extensions on the QX IP PBX This number is used for internal PBX calls e Display Name indicates an optional display name to identify the caller e Attached Line indicates the FXS or IP line corresponding extension it is attached to R is displayed in this column when SIP Remote Extension see below functionality is enabled on the extension e SIP Address displays the SIP address of the corresponding extension The column displays the full SIP address i e usernameOsipserver port when the Registration on SIP Server checkbox is selected If registration is disabled the SIP addres
17. Add the IP address into the Blocked IP list in Firewall Warning For this action to take effect the firewall should be enabled Y Discard SIP messages from IP address for Exceptions Epygi treats system security with the utmost priority and has taken an active approach to provide users with information and tools to aid in maintaining system security It is highly recommended that users of an IP based system need to be familiar with industry best practices to maintain system security Limitation of Liability and Remedies In no event shall Epygi Technologies be liable for any consequential incidental direct indirect special punitive or other damages including without limitation loss of data loss of phone calls loss of business profits business interruption loss of business information or other pecuniary loss arising out of the use or inability to use the Quadro Fig IT 197 SIP IDS Settings page Filtering Rules C Services gt Groups SIP IDS Exceptions for SIP IDS Add Delete IP address 172 30 0 0 16 Fig II 198 Exceptions for SIP IDS Table 126 epy8l Network Menu The Network menu allows you to configure the following settings e IP Routing Configuration IP Static Routes IP Policy Routes PPTP L2TP Routes e DHCP Settings DHCP Server DHCP Leases DHCP Settings for the VLAN Interface DNS Settings DNS Server Settings Dynamic DNS Settings e PPP PPTP
18. Calculate the voice energy for the last Calculate the voice energy every Switch to new Video Source if ener i Y Leave Active Y Close the Conference if Moderator did notjoinin 2 Y Close the conference if only one participant is connected recording view conference history Confirm text field Pie notification before Conferencectore P requires the confirmation of the Moderator Password Error Telepresence Settings appears if the password inserted in the Confirm text field does not match the one inserted in the Moderator Password text field Fig 1 297 Conference Settings General Settings Page QX50 QX200 QX2000 SW Version 6 0 x 187 QX50 0X200 0X2000 Manual II Administrator s Guide Participant Password can be entered to require a password for participant access to the conference It has to be entered twice for confirmation The password entered here should be used by the participant to join the conference The participant can participate in the conference only according to the rights speaker or listener granted by the moderator Max Duration sets the conference to be limited to a maximum duration in minutes Leave the field empty for unlimited conference duration With the Play Hold Music Until Moderator is Connected checkbox selected participants connected to the conference will listen to the hold music unless moderator will join the conference Automatic Speaker Detection checkbox enables the automatic det
19. Call Park and Paging Group extensions are displayed without a link in the Extensions Management table and extension pages Additionally the supplementary services configuration pages will not be accessible for this type of extensions Clicking on the Recording Box extension will move to the corresponding extension s Recording Box where the recorded calls can be managed To add an extension click on the Add button or use the Add Extension tab see below Edit opens the Edit Entry page where a newly created user or attendant extension settings might be adjusted To operate with Edit one or more record s have to be selected otherwise the No records selected error message will appear The Edit Entry page consists of two frames In the left frame settings groups are listed Clicking on the corresponding settings group displays their configuration options in the right frame Please Note Save changes before moving among settings groups Hide extensions attached to disabled IP lines functional button is used to hide extensions which are attached to the disabled IP lines When this functional button is pressed it transforms to Show all extensions functional button which is used to show all hidden extensions To enable the lines install a feature key from the Feature Keys page Add Extension Add Extension tab opens the Extensions Management Add Entry page where the type and number of the new extension should be defined This page consists
20. Defines the traphost and SNMP protocol version Configure VLANs on the LAN or WAN and assign IP address to the interface Establish VPN connection using Internet Protocol Security IPSec Establish VPN connection using Point To Point PPTP or Layer 2 Tunneling Protocol L2TP Fig II 199 Network Menu page 127 IP Routing Configuration QX50 0X200 0X2000 Manual II Administrator s Guide Routing is used to relay information across the Internet from a source to a destination Along the way at least one intermediate node is typically encountered Routing is different than bridging The main difference between bridging and routing is that bridging operates at the OSI Data Link Layer Level Two Media Access Control Layer and routing operates at OSI Network Layer Level Three QX IP PBX s IP Routing service allows you to route IP packets from one destination to another or to a specified router through QX IP PBX or a QX IP PBX VPN The IP Routing page is used to make IP Static IP Policy and PPTP L2TP routes for IP packets routing This page consists of three tables Entries in the tables are color coded according to the state of the route For example yellow indicates disabled routes green indicates successful routes and red indicates routes with an error IP Static Routes IP Static Routes are used to forward IP packets from the Network where the QX IP PBX is connected to the specified destination The IP Static Rout
21. Diagnostics Security Diagnostics Call Capture Ping Ping Target Ping Output Fig I1 265 Call Capture Active Calls page Fig Il 266 Call Capture Interfaces page Ping Traceroute amp Maintenance PING 192 168 74 36 192 168 74 36 56 data bytes 64 bytes from 192 168 74 36 seq ttl 64 time 1 071 ms 64 bytes from 192 168 74 36 seq 1 ttl 64 time 924 ms 64 bytes from 192 168 74 36 seq 2 ttl 64 time 0 916 ms 64 bytes from 192 168 74 36 seq 3 ttl 64 time 933 ms 192 168 74 36 ping statistics ts received 0 packet loss in avg max 0 916 0 961 1 871 ms Fig II 267 System Diagnostic Ping page 162 epygl QX50 0X200 0X2000 Manual II Administrator s Guide Traceroute Traceroute Target is used to enter the IP address or host name of the destination to be trace routed Diagnostics Security Diagnostics Call Capture The Start Traceroute button is used to process the router Traceroute triggering to check the Internet connection Traceroute Target Y Use ICMP In the field below these the output of the Ping or Traceroute procedure is shown Start Traceroute traceroute to 192 168 74 36 192 168 74 36 3 hops max 38 byte packets 1 192 168 74 36 192 168 74 36 2 163 ms 62 ms 0 609 ms done Output amp Maintenance p Traceroute checks the Internet connection by triggering the routers hops that are passed to reach the destination specified
22. Enable DID Service Previous Fig I1 116 CAS Signaling Wizard Page 2 CAS Signaling Wizard Routing Settings Trunk 1 192 168 74 127 5060 Selected Timeslots 1 Route Incoming Call to 00 Y CutThrough Automatic Ringing Down Y Pass Through Pound Sign Previous Fig I1 117 CAS Signaling Wizard Page 3 Attention When QX acts in the Network mode with the Attendant as a destination to route the incoming calls digit forwarding should be disabled on the PBX side Otherwise incoming digits may be mistaken as special calling codes on the QX IP PBX s Attendant Cut Through checkbox is available when signaling selected from the Signaling Type drop down list on the CAS Signaling Wizard Page 2 is different from R2 all types and is used to reconnect the call terminated by some reason e g user error network problems etc by going on hook and off hook again even if the call partner is off hook and not involved in the call Automat Ringing Down checkbox is available when signaling selected from the Signaling Type drop down list on the CAS Signaling Wizard Page 2 is different from R2 all types and allows an E1 T1 device connected to the QX to establish a hot line call automatic call without any digits dialed Pass Through Pound Sign checkbox is only available when signaling selected from the Signaling Type drop down list on the CAS Signaling Wizard Page 2 is different from E amp M FGD or R
23. IP Phone Templates IP Phones Logo ISDN Trunk Alcatel Temporis IP600 Choose File logo3 gif Alcatel Temporis IP800 Choose File Choose File No fi No fi e chosen e chosen Choose File No fi e chosen Choose File Choose File Choose File No fi No fi No file e chosen e chosen chosen Yealink SIP T22P Yealink SIP T26P Yealink SIP T28P Choose File Choose File Choose File No fi No fi No fi e chosen e chosen e chosen WARNING For Snom phone s the files must be in XML format accordi ng to Snom recommendation For Yealink phones the files must be in DOB format according to Yealink recommendation Save Fig II 103 IP Phones Logo page 72 epygl FXS Gateways The QX FXS Gateway is an analog Gateway that allows connecting analogue phones to a VoIP network The device can be used with QX IP PBXs to emulate additional FXS ports Both QX IP PBX and the FXS Gateway should be located in the same network QX IP PBX is connected to the QX FXS gateway through its MAC address The FXS Gateway Management page is used to define QX FXS Gateway devices in your network that can serve as FXS expansion modules for your QX IP PBX Additional FXS lines provided by the FXS Gateway can be connected to the IP lines on the QX IP PBX Add functional button opens FXS Gateway Management Wizard where new FXS Gateway should be defined T
24. In Case of Failover In Case if Call Failed to Establish Call Routing Call Recording NAT Traversa ble Call Routing Local AAA Table Global Speed Dial SIP Tunnel Class of Service Call Routing Wizard O Go Back amp Telephony Class of Services Add Entry Y Class1 Y Class2 Previous Fig IT 144 Call Routing Wizard Class of Services Edit Entry page Please Note Established patterns based on the Emergency Codes and PSTN Access Codes Settings in the System Configuration Wizard will be marked in bold and will be placed in the first position in the Call Routing Table Additionally they cannot be modified and deleted from the Call Routing Table The Duplicate functional button is used to create a routing pattern with the settings of an exiting one This is to avoid configuring a new routing entry completely by duplicating an existing entry with different settings To use the Duplicate button only one record may be selected otherwise the error message One row should be selected will appear The Duplicate button opens the Call Routing Wizard where all fields except the Pattern field are already filled in A Pattern for the new route will be required anyway The Move Up and Move Down buttons are used to move call routing patterns one level up or down within the Call Routing table The sequence of the routing patterns is important when making routing calls because the Call Routing table is parsed from
25. SW Version 6 0 x 59 e C Pyg l QX50 QXK200 QX2000 Manual II Administrator s Guide The Receptionist Phone Configuration Wizard Programmable Keys Configuration page is used to set the correspondence between the selected Functions and the available Programmable Keys on the IP Phone To do so assign a Function to each programmable key from the drop down list on this page The following options are available in the Functions list e Watch Ext watch the extension on the QX IP PBX and a possibility to pickup the call addressed to that extension e Call Park Ext watch the calls parked to the corresponding extensions and a possibility to retrieve the calls parked to that extension This list also contains a number of PBX services available on the QX IP PBX and accessible with the key combination see QX IP PBX s Feature Codes When configured from this page the key combinations become transparent for the IP phones too e Vmail accesses the voice mailbox of the extension to which the receptionist IP line is attached to e DND enables the Do Not Disturb service on the extension to which the receptionist IP line is attached to Receptionist Phone Configuration Wizard e CallFwd accessed Forwarding Management of the extension to which the receptionist IP line is attached to a Extensions Programmable Keys Configuration e AutoReDI auto redials the last dialed call Line Key e CallBack calls back to t
26. WAN IP Configuration PPP PPTP Configuration DNS Settings WAN Interface Configuration Email SMTP Short Text Messaging SMS lll for Protocol Ethernet WAN IP Configuration WAN Interface Configuration DNS Settings Fig II 7 Internet Configuration Wizard Getting Started page The Wizard allows navigating through the following basic configuration parameters and settings e Uplink configuration see below For Protocols PPPoE available only for QX50 QX200 e PPP PPTP Settings e WAN Interface Configuration see below e DNS Settings The Uplink Configuration page allows you to select the QX IP PBX s WAN interface connection type and its bandwidth settings These settings will make QX IP PBX available to the external network Depending on the Uplink Interface Protocol selection the page following the Uplink Configuration page is different Thus if PPPoE is selected the next page will be PPP Configuration while selecting Ethernet will bring up the WAN IP Configuration page The Uplink Configuration page offers the following components The WAN Interface Protocol radio buttons are used to choose the protocol depending on the requirements of the ISP Internet Service Provider eo PPPoE available only for QX50 QX200 turns on the PPP over an Ethernet connection o PPTP available only for QX50 QX200 turns on the Point to Point Tunneling Protocol PPTP interface used for the c
27. agent s phone If the call is not answered before this timer expires see the system will try to connect the call to another agent in that group Group Ring Timeout defines the maximum waiting time of the rm calls in the queue including connection time when the call is extracted from the queue and rings on the agent s phone until it is answered If this value for some call in the queue is exceeded then the call is being disconnected unless the call redirection is enabled from this page In that case the call will be redirected to another destination as defined here Custom Queue Messages d dit Delete Move Up Move Down Queue Message Timeout sec Play Count LudovicoEinaudiAA wav 5 1 Play Background Music RTP Channel _ a Choose Channel AA Audio Line In Fig IT 53 Extensions Management Edit Entry ACD Group Settings page Call Distribution Type defines the method of choosing the agents within the group for connecting the call The following distribution types are available e All Agents Ringing the system tries to reach all available agents in the group ringing their phones As soon as the first answers it cancels the calls to other agents similar to Many Extension Ringing on the QX IP PBX see Manual III Extension User s Guide If no one answers within Common Timeout the system either disconnects or redirects the call e Round Robin the system calls to the first available agent in the list of a
28. correspondingly The brief view displays the most important settings of the routing rules The detailed view displays all settings of the routing rules as they are configured in the Call Routing Wizard The alternating Hide disabled records and Show all records buttons are used to respectively hide or show disabled records in the Call Routing table The system does not consider the disabled records when parsing the table for the call route If the route has an Authentication or an Authentication amp Accounting selected from the AAA Required checkbox group it will have a link to the Users List in the Call Routing table The Users List page contains a list of authorized users defined from the Local AAA Table and gives the option to enable disable authentication of each user for a particular route Since the Call Routing Table may have multiple entries that could match to same pattern the table will be internally rearranged according to the rules with the following consequences e The pattern matching best to the Best Matching Algorithm will have the higher position in the rearranged list e If multiple patterns equally match to the Best Matching Algorithm the pattern with the lower metric will get the higher position in the rearranged list e If the multiple patterns with the same metric have been matched to the Best Matching Algorithm the pattern in the higher position in the table will get the higher position in the rearranged list The patte
29. i e successful missed and unsuccessful outgoing Call History If there are no any record made during last time interval the black file is archived Fig I1 250 Call History Archiving Settings page The External Backup of Call Detail Records Archive is used for configuring the Call History backup service The Send archive files to external server is used to enable disable the backup service and configuring whether the statistics should be kept locally after backing up them Two options of the Call History backup are available uploading the Call History file to the server or sending it to the mailing address The following group of manipulation radio buttons allows you to select whether the Call History files will be delivered by email or stored in some location on the server e The Send via Email radio button is used to send the Call History files via email The selection enables Email Address text field that requires the email address of the administrating person to receive the Call History files e The Send to Server radio button is used to store the Call History files on a remote server This selection enables the following fields to be inserted O The Server Name requires the IP address or the host name of the remote server O The Server Port requires the port number of the remote server O The Path on Server requires the path on the server to store the Call History files in The Send Method manipulation radio buttons allow you
30. se Import Export Scenario 00 scenario and custom messages file coma Export scenario appears when the Customized Scenario was previously configured for the current Auto Attendant The Download scenario link is used to download the scenario and voice message files to the PC and opens the file chooser window where the saving location may be specified Fig II 67 Import Export Scenario page O The Remove Scenario link removes the current Customized Scenario After pressing the Remove scenario link all configurations and uploaded voice messages will be deleted from the system O The View Download VXML Scenario link appears only when a customized scenario has been created and is used to view or download the generated script in a VXML file format The Predefined manipulation radio button selection allows you to switch the Attendant to the ACD Agent Scenario see ACD Management Attention This selection is only available if the ACD feature is previously activated from the Feature Keys page This page provides the possibility of uploading voice messages to be played in the custom Auto Attendant scenario It also removes and downloads the uploaded files to a PC Upload Custom Scenario Voice Messages Attendant 00 Custom Voice Messages enterlanguage wav The Upload Custom Scenario Voice Messages page contains a table enterdepartmert wav where uploaded custom voice messages are listed Use the Download functional button
31. 256 lAN Interface Statistics page Depending on the Watch LAN or Watch WAN Monitor link selected on the Network Status page the LAN Interface Statistics or WAN Interface Statistics page will be diaplayed LAN WAN Lan E o WAN Interface Statistics The page is automatically refreshed every minute Additionally Started at 21 Aug 2014 14 54 50 Time difference 3 sec the Refresh button allows to initiate refreshing directly Received Bytes 3026 Transmitted Bytes Status Se Received Packets 21 Transmitted Packets Receive Errors lo Transmit Errors Receive Drop Errors 0 Transmit Drop Errors Receive Overrun Errors 0 Transmit Carrier Errors Receive Multicast 10 Transmit Collisions li Refresh Clear Fig Il 257 WAN Interface Statistics page QX50 QX200 QX2000 SW Version 6 0 x 157 e C Pyg l QX50 QX200 QX2000 Manual IJ Administrator s Guide Statistics Network Transfer The Transfer Statistics page shows a user defined statistics table with the transmit receive value criteria interface type and time period It contains the following components Overview System Status Events CallHstory Conference History LAN WAN Statistics PSTN Channel Usage Time range of statistic table the drop down list includes the Transfer Statistics period in days statistics data that is to be collected and the eroticas Rain corresponding diagram charts that are to be
32. ADPCM speech coding at 24 Hitt The G 726 Standard radio buttons are used to select between 6726 40 ADPCM speech coding at 40 Kbit s rate a G 729a CS ACELP speech coding at 8 kbit s rate packaging the G 726 codewords into octets If you experience ag Eas EN AE problems with the G 726 voice quality when one of these si deacon ation ein packaging is selected try a different one O e If Use ITU_T specification is selected the ITU 1 366 2 AAL2 iiaa type 2 service specific convergence sublayer for narrow band asia Use IETF RFC services type packaging of codewords is used where packing code words into octets is starting from the most significant rather than the least significant digit in the octet RTP RTCP Port Range Min 6000 Max 6255 e If Use IETF RFC is selected the IETF RFC RTP Profile for Z Enable RTCP Support Audio and Video Conferences with Minimal Control type Save packaging of codewords is used where packing code words is Figli 164 RTP Settings page starting from the least significant position in the octet RTP RTCP Port Range e Min minimal port has to be higher than 1024 and lower than the maximal port range Only even numbers are allowed e Max maximal port has to be lower than 65536 and higher than the minimal port range Only odd numbers are allowed Since the specified maximum port has to be higher than the minimum port the error message Min port number should be less than max po
33. Add Entry page e Authentication by Login this selection is used to set the authentication based on the username and password inserted by the user upon login The Username text field requires the authentication username Only numeric values are allowed for this field otherwise the error message Incorrect Username digits allowed only will appear The Password text field requires the authentication password Only numeric values are allowed for this field otherwise the error message Incorrect Password digits allowed only will appear e Authentication by PIN this selection is used to set the authentication based on the PIN inserted by the user upon login Only digit values are allowed for this field otherwise the appropriate error message will be displayed The Expiration Date and Time drop down lists are used to set the date and time when the registration will expire The Expires in checkbox is used to enable the Expiration Date and Time feature The Description text field requires an optional description about the calling party Edit opens the Edit Entry page to modify the local AAA entry Global Speed Dial Directory The Global Speed Dial Directory page is used to define multiple speed dial rules write and save them in a file and then upload all of them at Call Routing Table Call Routing Local AAA Table S a SIP Tunnel Class of Service once Global Speed Dial Directory To compose the configuration file any text editor can
34. Auto Attendant Delay after message requires the delay in seconds after which the Recurring message will be played Recurring message indicates the file name used to upload a new custom Auto Attendant recurring message The Auto Attendant Recurring message will play after the Attendant Welcome message if it is uploaded QX50 QX200 QX2000 SW Version 6 0 x 45 QX50 0X200 0X2000 Manual II Administrator s Guide Play Count text field indicates the number of times the corresponding Recurring message will be consecutively played to the caller Interval requires the time period in seconds between consecutively played Recurring messages Browse opens the file chooser window to browse for a new custom B extensions Edit Scenario 00 MainMenu welcome or recurring message file coma Press the Save button to submit the changes or use Go Back to keep the initial data Welcome message Choose File No file chosen Attention The uploaded file needs to be in PCMU CCITT u law 8 oetyatermessages E seconds kHz 8 bit Mono wave format otherwise the system will prevent sage Choose Fie No file chosen uploading it and the Invalid audio file or format is not supported warning message will appear The system also prevents uploading if O there is not enough memory available for the corresponding extension z ae Ca E and the You do not have enough space warning message will appear en ae 1 jone No Action No Action
35. Call Type from the Allowed Call Type drop down list and the extension from the Route FXO Call to drop down list to route the FXO calls correspondingly 4 Inserta PSTN number in the same named text field to identify the FXO line 5 Enable Alternative AC Termination Mode if this is a requirement of your CO 6 Press Save to submit the FXO line settings E1 T1 Trunk Settings The QX50 0X200 0X2000 has no own E1 T1 trunks only shared E1 T1 trunks are displayed in this page if available The shared trunks lines can be edited from this page Any changes applied in this page will be automatically reflected on the QX E1 T1 gateway s that share its E1 T1 trunks E1 T1 service allows QXE1 T1 Gateway to be connected to a PBX or to the CO Central Office via E1 T1 lines using E1 T1 CAS CCS signaling QXE1 T1 Gateway can be connected to act as a User if connected to a CO or as Network if connected to a PBX If a private PBX is connected to QX E1 T1 Gateway it should be configured in network mode if the E1 T1 line from a CO is connected to QXE1 T1 Gateway it should be configured as a User The E1 T1 Trunk Settings page is used to configure the E1 T1 trunk and the timeslots settings The Trunk Settings table lists the available E1 T1 trunks on the QX IP PBX and their settings Trunk name E1 T1 mode interface signaling types Clicking on the trunk will open its Signaling Settings page Trunk CAS Signaling Settings or Trunk CCS Signaling Settings page
36. Configuration page offers the following input options Host Name requires a host name for the QX IP PBX device Domain Name requires the LAN side domain name which the QX IP PBX belongs to IP Address requires the QX IP PBX host address for the LAN interface Subnet Mask requires the QX IP PBX hosts Subnet Mask QX50 QX200 QX2000 SW Version 6 0 x o Setup Internet WAN Date andTime Email SMTP Short Text Messaging SMS System Configuration Wizard Getting Started This wizard guides you through e System Configuration DHCP Settings for the LAN Interface Regional Settings and Preferences Emergency Codes and PSTN Access Code Settings Fig II 3 System Configuration Wizard Getting Started page System LAN Internet WAN Date and Time Email SMTP Short Text Messaging SMS System Configuration Wizard System Configuration Host Name QX200 12 Domain Name epygi config loc LAN IP Configuration IP Address 172 30 4 1 Subnet Mask 255 255 255 0 Previous Fig II 4 System Configuration Wizard System Configuration page 11 QX50 0X200 0X2000 Manual II Administrator s Guide The Regional Settings and Preferences are used to select settings specific to the location of the QX IP PBX This is important for the functionality of the voice subsystem Internet WAN Date and Time Email SMTP Short Text Messaging SMS bd _ System Configuration Wizard The Regional Se
37. Entry page consists of the following components The Caller Information requires the Call Type and the caller s Address The Called Party Information consisting of the Call Type and the called party s Address Call Recording Settings The Call Type lists the available Call types Enable Disable Add Edit Delete Move Up Move Down State Caller Pattern Called Pattern Recording Type Max Recording Duration Enabled Auto Auto Always start automatically PBX indicates that the calling or called party is QX IP PBX extension lt lt lt Switch to Basic View SIP indicates that the calling or called party is located in a E SIP network external to QX IP PBX Fig II 156 Call Recording Advanced View Settings page PSTN indicates that the calling or called party is located in PSTN network external to QX IP PBX Call Recording Settings Add Entry Auto indicates any of the types listed above a Caller Information Telephony The value in the Address text field is dependent on the Call Type defined in the same named drop down list If the PBX call type is selected the QX IP PBX extension number should be defined in this field For the SIP call type the SIP address should be defined for the PSTN call type the PSTN user number should be defined here In case of Auto call type any of the addresses listed above are allowed Wildcards are A ion 1 hour applicable for this field neonnete SES
38. Epygi s sample VoXML script modify that and customize for your application The IVR voice prompts should be recorded and uploaded as usual The ACD Management page consists of 3 sub pages Skills Agents and Groups The Skills page contains a list of all available skills and their descriptions The skills defined in this page are then used in the agent management see above to assign the skill level to the agents english language knowledge french language knowlegde spanish language knowledge tech support knowledge Fig II 87 ACD Management Skills page Overview Extensions EN Agents Groups Add opens the Add Skill page where a new skill may be ACD Management Add Skill defined The Add Skill page contains the Skill text field to ee tee define the skill name and an optional Description field for the skit frena des crip ti O n of th e S kill Description french language knowlege Fig ll 88 ACD Management Add Skill page The Agents page of ACD Management contains a list of agents and the skill set corresponding to each agent Every agent is characterized by an Agent ID which should be unique in the system Agent IDs and passwords are used by the agents for Extensions logging into Agents Group see description above o Skills ACD Management Add Edit Delete Agent ID Agent Name Calling Address Skill Levels Description 103 PBX 103 i Add opens the Add Agent page where a new agent may be 1130 SP 11369
39. MAC Address DNS Settings Date and Time Settings Email SMTP Settings Short Text Messaging SMS Settings System Security Licensed Features Redundancy Settings Language Pack Extensions Management for QX50 QX200 QX50 QX200 QX2000 SW Version 6 0 x QX50 0X200 0X2000 Manual II Administrator s Guide Appendix System Default Values System Default Value Login name admin Password 19 epygiqx epygi config loc For QX50 QX200 172 30 0 1 For QX2000 192 168 0 200 Subnet Mask 255 255 255 0 Disabled Locale US TimeZone Central Time US amp Canada Emergency Code 911 PSTN Access Code 9 Ethernet For QX50 QX200 Upstream 100000 Downstream 100000 For QX2000 Upstream 1000000 Downstream 1000000 Min Data Rate 0 Assign automatically via DHCP Assigned by device MTU 1500 Bytes Dynamically by provider Simple Network Time Protocol Server and Client enabled SNTP Server ntp1 epygi com Polling interval 6 System Mail Settings disabled TLS disabled Enable SMTP Authentication disabled User Name empty User Password empty Enable SMS Service disabled Security Level Medium 3PCC support No key found ACD support No key found Barge In No key found Redundancy license key required for QX2000 only No key found DCC Pro Support No key found DCC Basic Support No key found iQall Toggling Support No key found IP Phone support No key f
40. Metric or leave the default Enter a Description if needed Enable the Filter on Source Modify Caller ID checkbox if the route functionality should be limited This is dependent on the source caller information Enable the Set Date Time Period s checkbox if a route should be functional within certain time date intervals Enable the Set Overall Calling Time Limit checkbox if the overall duration of the calls placed with the selected routing rule should be defined Enable the Set Tracing Debug Options on This Rule checkbox if the tracing debug options should be defined Press Next Select the user or attendant extension from the Use Extension Settings drop down list that the call will be placed on Specify the Destination Host and Port Number Username and Password if an IP or IP PSTN call type has been selected For the IP PSTN call type enable Multiple Logons if necessary Enable the Use RTP Proxy checkbox if needed Choose the Authentication and Accounting method from the AAA Required drop down list Choose a Fail Reason from the corresponding drop down list Configure Transport Protocol for SIP messages and SIP Privacy parameters as needed Press the Next button If the Filter on Source Modify Caller ID checkbox has been previously enabled and the destination type is different from the FXO fill in the Source Number Pattern into the corresponding text field Choose the needed value from the Source Type drop down list as well as the Nu
41. O Go Back Caller Settings The Call Type drop down list includes possible incoming call types PSTN SIP or Auto In SIP the caller connects QX IP PBX through a SIP server and PSTN means the caller is a PSTN user Auto is used for undefined call types and the destination independent on whether it is a PBX number SIP address or PSTN number will be reached through Routing amp Extensions Caller Settings Call Type SP v 11369 sip epygi loc Caller Address espero wildcard supported Login Extension 106 v Y Automatically Enter Call Relay Menu Description The Caller Address text field requires the caller s SIP address see chapter Entering SIP Addresses Correctly or PSTN number to be added to the trusted phones list The PSTN number length depends on the area code and phone number The wildcard is supported in this field If the caller address already exists in the Authorized Phones Database the error message The record already exists appears when selecting the Save button Callback Settings Y Enable Callback Callback Call Type PBX Y a 115 Callback Destination Callback Response Delay Save Fig II 95 Authorized Phones Database Add Entry page The Login Extension drop down list provides all existing extensions on the QX IP PBX When calling the QX IP PBX Auto Attendant a trusted user will automatically be logged in as the selected extension i e the extension number and its password
42. P Preferred Identity or a Remote Party ID then the CallerID on ISDN contains the number from the user name field and the Redirecting Number IE contains the original number from the identity field SIP user agent should check for the existence of the P Asserted Identity then the P Preferred Identity then the Remote Party ID to fill the identity field For the calls from ISDN to SIP with Caller ID the SIP Invite message contains P Asserted Identity field with the original number value from the Redirecting Number IE on ISDN The SIP From field contains the value from the user name When the Send Calling Party Subaddress checkbox is selected QX will send the extension number as subaddress and the value defined in the Default outgoing Caller ID field as caller ID on the outgoing call When this checkbox is disabled no subaddress information will be sent and the caller ID will be defined according to the selection of the Use Default Outgoing Caller ID checkbox see above Caller ID information along with the Subaddress can be displayed on the phone display depending on the phone and PBX settings and capabilities When the Ignore Empty Channel Identification in CALL PROCEEDING Msg option is selected QX will ignore the empty ISDN L3 Channel Identification information element in CALL PROCEEDING message and will not response with STATUS message When this checkbox is disabled QX will response with STATUS message on empty Channel Identification information e
43. PBX This information might Biaon AAA be used by remote side for authentication purposes epica ace Registration failure timeout 120 Fig II 166 SIP Settings page Enable Session Timer enables advanced mechanisms for connection activity checking This option allows both user agents and proxies to determine if the SIP session is still active The DNS server for SIP radio button group allows you to choose between regular DNS servers configured in the DNS Settings page and specific DNS servers for SIP traffic e Use default is used to apply regular DNS servers for SIP traffic e Specific is used to enable SIP specific DNS servers For this selection both primary and secondary SIP DNS servers should be defined in the SIP DNS 1 and SIP DNS 2 text fields At the least a primary DNS server should be inserted The SIP Timers radio button group is used to define the timeouts of the SIP messages retransmission e RFC 3261 will apply standard SIP timers described in the corresponding specification e High availability will apply SIP timers to shorten the call establishment registration confirmation and registration failure procedures This selection provides more firmness to the SIP connection but increases the network traffic on the QX IP PBX QX50 QX200 QX2000 SW Version 6 0 x 112 QX50 0X200 0X2000 Manual II Administrator s Guide QX50 0X200 0X2000 Manual II Administrator s Guide pya e Custom allows manually definin
44. QPe eP AAA 60 Authorized PHONES Database its 64 Gall BAG A o o E PU o Ar OE E E EE PO PE O E O 65 Interfaces Menton eta cl ado 67 QX50 QX200 QX2000 SW Version 6 0 x 4 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide DP UNG S or 68 AA o O P e RI E E ECO E erate S 69 Supported MP eee ec eee ee ee ee ee ee eee 69 Programmable Keys Gro iol ate 0 crs Lalo 4 A paulo ee eee 70 IP Phone Template gt POPE Po o oo ee E eee eee 71 IP Phones e PP o O o ee 72 FXS Gateways assi 73 FXS LIE Sos e 74 DA AAPP PP RP UN O OE E E A E CNS CT eT 74 Dra o Co A PP E O E O cr a en re a re nT eer re tr 75 HEEE E Te eT TT cr Cr rT E 75 A o o lt m oo oo o m AAA 76 EDT as 77 coria WARE CEE Servicio nadaa rinda nadaa 84 ISDN TTUDK SOS e iio 85 External PSTN Gate Way S sdsatnusicansainaiaancaiaatancassaasainataananiaacansaiaaaaanad IRREAL ARCA ARALAR SUR aii RCA ALEA DARAR CARR RARA LARREA OLA REO 89 alar PP ee A E 89 Talepho ny Meni eraon AE Uno NA 5 RI NEANS AEAEE ANE EE 90 A NO 91 Call Routing Table PAPA o _ o aani diddaiaraeadoraa 92 Call vee Lana sien oni O Uae IRE RUEc Noein Ea naIn kaa Oa Naa NOa eta nelia atari tennis 99 Local AAA Ds PPP A A o e nn 100 Global speed Dial DICC ii 100 Allowed Characters and Wildcards cussssansacudiaan dol N a its 101 Bese Mait Me Alp Ori
45. QX200 Display name none Password empty 1001 1200 extensions attached to the IP lines 1 200 Kickback disabled Call Relay disabled Login Allowed disabled 3pcc Click2Dial Access Allowed disabled Audio Line out disabled Show on Public Directory disabled Percentage of Total Memory 0 04 Registration username same as extension number Registration password empty SIP server empty SIP Server port 5060 SIP Server Registration disabled Authentication User Name undefined Send Keep alive Messages to Proxy disabled RTP Priority Level medium Do Not use SIP Old Hold Method disabled Outbound Proxy Secondary SIP Server and Outbound Proxy for Secondary SIP Server undefined Remote Extension disabled Call Queue disabled Internal Voice Mail for all extensions Configuration wizard activated Shared Mailbox undefined For all extensions except 101 and 102 Codecs G711u preferred G711a G729a enabled G726 16 G726 24 G726 32 G726 40 iLBC G 722 G 722 1 TDVC H 263 H 263 and H 264 disabled Out of Band DTMF Transport enabled T 38 FAX enabled Pass Through FAX enabled Pass Through Modem disabled Force Self Codecs Preference for Inbound Calls disabled SRTP Policy Make unsecure calls accept anything For extensions 101 and 102 Codecs G711u preferred G711a G729a G726 32 G726 16 G726 24 177 Parameter Extension Settings Codecs for QX2000
46. Recording free space provides information on the number of minutes seconds of free recording box space Refresh functional button is used to refresh the Recording Box for any latest recordings or status changes Send to FTP functional button is used to move one or more selected recordings to the FTP server configured from Recording Storage Settings in Recording Box Extension Settings page New recordings field shows the number of newly done call recordings since the user s last access to the voice mailbox All recordings field shows the number of all recordings existing in the Recording Box Recording Box table displays the following information Status indicates whether the call recording is New and not yet played New recordings are displayed in bold font Caller is the address of the caller of the recorded call Callee is the address of the called party of the recorded call Date Time is the call recording start date and time amp Extensions ESG Add Extension Add Multiple Extensions Bulk Import Recording Box 800 O Go Back Send To FTP Delete Status Ashot Sargsyan 11105 gt Artur Hayrapetyan lt 11203 gt Levon Dadayan lt 11380 gt Levon Dadayan lt 11380 gt Levon Dadayan 11380 gt 232 lt 20202 sip epygi com gt 232 lt 20202 sip epygi com gt 7412218 192 168 0 209 232 lt 20202 sip epygi com gt David Raysyan lt 20206 gt 232 lt 20202 sip epygi com
47. Request Method manipulation radio buttons allow to select the HTTP request method used by QX IP PBX the access the SMS gateway POST or GET Send Test SMS is used to send a test SMS to the defined SMS Recipient Address This button will be enabled if correct values have been submitted and saved on this page QX50 QX200 QX2000 SW Version 6 0 x 17 QX50 0X200 0X2000 Manual II Administrator s Guide pya System Security The System Security Management offers a possibility of managing the global security levels The System Security Management page includes the following components v Pending Foe Ny Events Admi Basic Setup System Security Licen The Security Level table allows selecting the Security Level defining requirements to the IP Lines password strength and the Security Report granularity The security levels are as follows 4 Setup System Security Management Security Level This allows a user to enter any SIP Registration password when configuring an IP phone e Low There are no specific restrictions on the strength of the saved password Only the critical warnings on the Call Routing Rules to PSTN and IP PSTN disabled Firewall and IDS will be generated in Security Report The Security Diagnostics tool will warn for only the most critical security issues This applies moderate password enforcement for the SIP Registration password when configuring an IP phone The Security Diagnostics tool will warn
48. Routing Wizard In Case if Call Failed to Establish a notification event is printed when the call executed with the routing rule failed The Call Routing Wizard Class of Services Edit Entry page is used to assign the defined class of services to a certain call routing pattern To use Class of Service feature for the corresponding routing rule it should be enabled from the Class of Service page The Class of Service CoS functionality allows to permit or deny the attempt of extensions to use certain types of call routing rules Suppose you want for a certain group of PBX Conference extensions to deny the right to make international calls but allow them to make local and long distance calls and for another group of PBX Conference extensions give a permission to make international calls only The classes defined in the Class of Service page will appear on this page to assign the corresponding routing rule to a certain class of service s Please Note The Class of Service feature is for PBX source type routing rules applicable only Please Note The Filter on Source Modify Caller ID option should be selected on the first page of the Call Routing Wizard to have a possibility to select the source caller type as a PBX Each routing rule can be attached to a several class of service s QX50 QX200 QX2000 Manual II Administrator s Guide Call Routing Wizard Go Back Tracing Debug Options In Case of Successful Call Y
49. SIP 2 0 UDP 192 168 70 25 5060 branch 29hG4bKEPSVBUS0cf8b942 9e81 41 el fd8e 8663dbaf63el To lt sip 74206444 192 168 74 206 gt tag 141033730164 d663el 3e93 4985 aeec ab0aae5209c3 From lt sip 131 192 168 70 25 gt tag 1410345872006b07 d6 aac0 412e f88d 63915c2cb5b0 CSeq 906 INVITE Call ID 07d06544 42b1 4174 7ea7 08d85c02e0f1 Aram_Manukyan epygi loc Allow INVITE ACK CANCEL BYE OPTIONS INFO SUBSCRIBE NOTIFY REFER MESSAGE UPDATE Contact lt sip 74206444 192 168 74 206 5060 gt Content Type application sdp Supported replaces norefersub Server Epygi Quadro SIP User Agent v5 3 67 MACRO IPPBX Content Length 250 v 0 o 854 36 IN IP4 192 168 74 206 c IN IP4 192 168 74 206 t 00 m audio 6156 RTP AVP 0 8 18 101 a rtpmap 0 PCMU 8000 a rtpmap 8 PCMA 8000 a rtpmap 18 G729a 8000 a fmtp 18 annexb no a rtpmap 101 telephone event 8000 a fmtp 101 0 15 teremt SID message buffer end Fig II 269 System Logs page This page is used to adjust system logging settings view system logs directly in your browser or download them locally to your PC The System Logs Settings page is used to adjust the system logging settings and contains the following components The Enable User Logging checkbox is used to enable user level logging This logging contains brief information about events on the QX IP PBX The Enable Developer Logging checkbox is used to enable developer high level logging This logging contains detailed infor
50. SIP and PSTN users can be added to the Authorized Phones Database Authorized Phones Database Extensions Add Edit Delete Call Type v Caller Address The Authorized Phones Database table displays all trusted Psm 505614 118 callers with their settings For example the call type caller ee address extension they automatically login with information if they have automatic access to Call Relay Menu of the Auto Attendant etc Login Extension Automatically Enter Call Relay Menu Callback Enabled SIP 11369 sip epygi loc Delay 60 sec Enabled PBX 115 Delay 30 sec Fig IT 94 Authorized Phones Database Each record in the table has an assigned checkbox The checkbox is used to edit or delete the corresponding record The No records selected error message occurs if the user activates the edit or delete button with no records being selected The error message One record should be selected appears if the user tries to edit more than one record The heading of each column in the table has a link By clicking on the column heading the table will be sorted by the selected column When sorting ascending or descending arrows will be displayed next to the column heading The Add functional button refers to the Authorized Phones Database Add Entry page where new trusted users may be entered The Authorized Phones Database Add Entry page offers two groups of input options Authorized Phones Database Add Entry
51. Server indicates the address of the SIP server The field is not limited regarding symbol usage or length It can be either an IP address such as 192 168 0 26 or a host address such as sip epygi com QX50 QX200 QX2000 SW Version 6 0 x 27 epygl SIP Port indicates the port number to connect to the SIP server The SIP server port may only contain digit values otherwise the error message SIP Server Port is incorrect will be displayed when applying the extension settings If the SIP server port is not specified QX IP PBX will access the SIP server through the default port 5060 Registration on SIP Server enables the SIP server registration option If the extension has already been registered on an SIP server its IP address will be displayed in brackets Please Note If the ITSP does not require each DID to uniquely register to the external SIP server then only enter the DID number in the User Name DID Number field The other fields are not required 3 SIP Advanced Settings This group is used to configure advanced SIP settings Outbound Proxy Secondary SIP Server and Outbound Proxy for the Secondary SIP Server settings and to define other SIP server specific settings The SIP Outbound proxy is an SIP server where all the SIP requests and other SIP messages are transferred Some SIP servers use an outbound proxy server to escape restrictions of NAT For example Free World Dialup service uses an Outbound Proxy server If an Outbound
52. Settings Advanced PPP Settings SNMP Settings Global SNMP Settings SNMP Trap Settings e VLAN VPN Configuration IPSec Configuration PPTP L2TP Configuration QX50 QX200 QX2000 SW Version 6 0 x Overview IP Routing Overview IP Routing IP Static Routes IP Policy Routes PPTP L2TP Routes DHCP Server DHCP Leases DHCP for VLAN DNS DNS Server Dynamic DNS PPP PPTP Advanced PPP Global SNMP SNMP Trap IPSec PPTP L2TP QX50 0X200 0X2000 Manual II Administrator s Guide Y Pending Configure static IP Routes for forwarding the IP packets from the network to the specified destination via specified IP address Configure IP Policy Routes for forwarding the IP packets from the specified source via specified IP address Configure PPTP L2TP Routes for forwarding the IP packets through the PPTP and L2TP tunnels Enable the DHCP Server and choose the dynamic IP address range to assign to clients List of DHCP IP addresses and host names provided by the DHCP Server Configure DHCP Server settings for VLAN interfaces Configure service provider DNS settings to resolve DNS addresses Configure DNS services for LAN connected hosts Configure Dynamic DNS DynDNS service for mapping a dynamic IP address to a host name Configure PPP PPTP connection basic settings Configure PPP PPTP connection advanced settings Configure contact details for network management server
53. The tool will save the settings ina bulk User Extension configuration file that will be ready to upload to the QX IP PBX e Import the configuration file to the QX IP PBX using the Extension Import feature Please Note The Bulk User Extensions Importer tool is applicable only for Adding and Modifying the extensions of User Extension type The extension types other than User Extension such as Auto Attendant Pickup Group etc currently are not supported by this tool QX50 QX200 QX2000 SW Version 6 0 x 52 QX50 0X200 0X2000 Manual II Administrator s Guide To configure the Extension Templates on the QX IP PBX select the Extension Template Management tab from this page The Extension Template Management page is used to configure different sets of user extension settings The Extension Template Management offers the following components Extension Template Management Extensions Extension Template Management Extension Import Settings Add Edit Delete e Add opens the Extension Template Management Add Template y Entry page where a new template can be created e Edit opens the Extension Template Management Edit Entry page where the settings of the user extension template can be configured Fig I1 72 Extension Template Management page The template file contains the common settings for user extensions which can be the same for a group of extensions The other settings which have to be different for
54. The value in this field is strictly dependent on the Call Type defined in the same named drop down list If the PBX call type is selected the Calling Address field should contain the extension number on QX IP PBX and the corresponding agent can be reached by calling on extension number located on the same QX IP PBX However it doesn t necessarily mean that the agent shall be located at that QX IP PBX if the extension is remote extension then agent s location might be far from QX IP PBX For the SIP call type the Calling Address field should contain the SIP address see chapter Entering SIP Addresses Correctly and the corresponding agent can be reached by calling on SIP address The agent with that kind of termination number might be located either at the same QX IP PBX or anywhere else in the SIP network For the PSTN call type the Calling Address field should contain the PSTN number and the corresponding agent can be reached by calling on PSTN number via some PSTN interface on QX IP PBX FXO The agent with that kind of termination number is located in the PSTN network fixed or cellular For the Auto call type the Calling Address field should contain the phone number routable through Call Routing Table on QX IP PBX The agent with that kind of termination number might be positioned in any of the above mentioned locations Pressing on the Skill Value column of the Agent Management table will lead you to the Agent Skill Levels page where the skill
55. Wizard consists of several pages and allows you to create a new PPTP or L2TP connection PPTP L2TP Connection Wizard Go Back The PPTP L2TP Connection Wizard Page 1 consists of the Ala PPTENZTP Connection following components Connection Name to74554Li Connection Type L2TP Connection Name text field requires a connection identification name The name of the connection cannot start with a digit symbol however it can contain digits further in the name Connection Type drop down list allows to select the type of the connection PPTP or L2TP Fig II 230 PPTP L2TP Connection Wizard Page 1 The PPTP L2TP Connection Wizard Page 2 consists of the following components The Peer Name text field requires the connection peer name If PPTP L2TP Connection Wizard you are about to create a client connection then the server s name owe should be defined here If you are creating a server connection AE then the client s name should be defined here Please Note When creating a connection with a Windows Server ensure that a user with the QX IP PBX s host name and Dial in access exists on the server When creating a connection with a Windows Client ensure that the Peer name specified on this page matches the Dial in connection s username MSCHAP MSCHAP SCHAPV2 MPPE 128 Bit Please Note The input in the Peer Name field should only be in Latin characters otherwise an error occurs and no c
56. a custom scenario file has been previously uploaded and is used to view or download the scenario file The Remove Scenario link is used to remove a custom scenario file and return to the default Auto Attendant scenario e The Upload VXML Scenario Voice Messages link refers to the page where voice messages used in the uploaded custom scenario should be managed The Customized Scenario radio button selection allows you to switch the Attendant to the customized Attendant scenario The Customized Scenario radio button selection enables the following components e The Create Scenario link refers to the Edit Scenario page where a new scenario for a current Auto Attendant might be created The Edit Scenario page consists of two pages for menu configurations The Main menu configuration page and the Submenus configuration page The Main menu is the menu where all incoming calls to the certain Auto Attendant will be placed first The Submenus are the supplementary menus which can be called from the other menus Both the Main menu and all Submenus can call each other This allows the opportunity to have several index levels for the Auto Attendant There are no limitations on the depth and nesting levels of menus The Main menu page consists of the following components Welcome message indicates the file name used to upload a new custom Auto Attendant welcome message The Auto Attendant Welcome message will play only once when callers enter the Customized
57. a list of user specific GUI pages is displayed Select the user in the table and press Edit to manage the permission for the corresponding user amp Maintenance Fig I1 276 User Rights Management Roles page On the Change Access Rights page Grant Access Deny Access epygi functional buttons are used to grant or deny access to certain GUI aR page s for the selected user uses User Rights Management Diagnostics System Logs User Rights Backup Restore When access to a certain GUI page is denied for a user the You are not authorized to access this page warning message will be ee dis played CGI Name Gana extensionstat General PBX information for the extension users Granted Changing access rights for role extensions amp Maintenance statistics Call Statistics Granted userstatistics Call Statistics Granted forwardadd Forwarding List Add Edit Entry Granted huntadd Call Hunting Add Edit Entry Granted sds Speed Calling Settings Granted findfollowadd FindFollow List Add Edit Entry Granted userpsw Changing Password for admin or extension Granted usersettings Extension Account Settings Granted vms Extension Voice Mail Settings Granted hold Basic Services Hold Music Settings Granted dnd Basic Services Do Not Disturb Settings Granted hotline Basic Services Hot Line Settings Granted redial Basic Services General Settings Granted addressmanagement Extension Caller ID Based Services Table Granted supserv
58. a string and is not being resolved It has no i ae advantages over the previous form Use IPSec compression Please Note The Local ID and Remote ID values are mandatory for RSA selection and are optional for Shared Secret selection ise However it is recommended to define the Local ID and Remote ID values for multiple road warrior connections Remote ID 2 Fig I1 226 IPSec Connection Wizard Automatic Keying Settings page PFS Perfect Forward Secrecy is a procedure of system key exchange which uses a long term key and generates short term keys as is required Thus an attacker who acquires the long term key can neither read previous messages that they may have captured nor read future ones Use IPSec Compression enables IPSec data compression This option is displayed only if the IPSec VPN partner supports it Language The forth page of the IPSec Connection Wizard contains IPSec o Connection Properties which serve to specify the members of Pse Conigwaton Wizard the IPSec Connection and to set the basic parameters for O Go bad encryption IPSec Connection Properties t07440M32x A group of radio buttons are used with Dynamic IP Road Warrior and Static IP Remote Gateway to select if the remote QX IP PBX or another VPN gateway device is connected to the Internet with a dynamic IP address and is acting as a Road Warrior or is connected to the Internet with a fixed IP address RATED and is acting a
59. activates FXO support for the Seren emcee IES es Recto ISS selected FXO line FXO Settings The Allowed Call Type is used to choose the allowed call directions z a EA ae SE TT TE a for the corresponding FXO line The administrator may choose mereces o Yes cintia cil dl between FXO2 es Both incoming and outgoing calls FXO 3 Yes Both incoming and outgoing calls e Enabling incoming calls prohibiting outgoing calls for the EXxO4 es Both incoming and outgoing calls Routing 7740 selected FXO line e Enabling outgoing calls prohibiting incoming calls for the selected FXO line e Enabling incoming and outgoing calls for the selected FXO Fig II 110 FXO Settings page line The Route incoming FXO Call to manipulation radio buttons group allows you to define the destination where incoming calls addressed to the corresponding FXO line will be forwarded to e Extension this selection allows you to choose the local PBX user or auto attendant extension to forward calls If an inactive extension is chosen from this list the voice mail system will ae answer the call addressed to the corresponding FXO line If the Auto Attendant extension is chosen it will become the default user for the corresponding FXO line on the QX IP PBX FXO Settings FXO 4 O Go Back Y Enable FXO Allowed Call Type e Routing this selection allows you to forward the incoming calls to the destination defined through Call Routing Table This selectio
60. address range of the possible peers behind the PPTP L2TP tunnel whereto the IP packets should be routed Network Ov IP Routing DHCP IP Static Routes PPTP L2TP Routes Add IP Policy Route O Go Back Priority From IP Static Routes IP Policy Routes PPTP L2TP Route Add PPTP L2TP Route O Go Back Route via PPTP gt to7415QX200 Y Route to 215 156 74 15 16 Save IP Static Routes IP Policy Routes Midi PPTP L2TP Routes Enable Disable Add Edit Delete Target State QX50 0X200 0X2000 Manual II Administrator s Guide DNS PPP PPTP Fig I1 203 Add IP Policy Route page PPP PPTP Fig IT 204 PPTP L2TP Routes table PPP PPTP SNMP Fig I1 205 Add PPTP L2TP Route page The Enable and Disable functional buttons are used to activate or to deactivate the selected route s At least one route should be selected to use these functions otherwise the error message No record s selected will appear DHCP Settings The DHCP Settings page provides the option of enabling a DHCP server and controlling the QX IP PBX user s LAN settings Therefore QX IP PBX LAN users will automatically be provided with the following settings using the configured parameters e IP addresses NTP corresponds to the QX IP PBX s IP address WINS server Nameserver corresponds to the QX IP PBX s IP address Domain name QX50 QX200 QX2000 SW Version 6 0 x 129 e C Pyg l QX50 QXK200 QX2
61. addresses As all QX IP PBX devices have the same default IP addresses on delivery at least one of them must be reconfigured in order to set a new IP address QX IP PBX supports several kinds of VPN connections such as IPSec and PPTP L2TP Attention It is strongly recommended not to run different types of VPN tunnels between the same endpoints simultaneously IPSec Configuration An IPSec connection includes authentication and encryption to protect data integrity and confidentiality VPNs are virtual in the sense that individuals can use the public Internet as a means of securely accessing an internal network Once the IPSec connection is established users have access to the same network resources addresses and so forth as if they were connected locally VPNs are private because the data is encrypted between two VPN gateways Encryption makes it very difficult for anyone to intercept data and capture sensitive information such as passwords The QX IP PBX can be set up to act as a VPN router when connected to the Internet with a fixed IP address or as an IPSec connection Road Warrior when using dynamic IP addresses Establishing an IPSec connection normally requires the functionality of a VPN gateway on each side of the communication line An intelligent Internet access router for example QX IP PBX delivers this function but also PCs or workstations may also be equipped with VPN gateway functionality Home offices typically prefer dynam
62. available on QX2000 The IP Lines page displays a table with the available IP lines on the QX IP PBX Entering the feature key in the Feature Keys page can enable more IP lines The IP Lines table lists all available IP lines with additional information about each of them number of the extension attached to it information about the phone type and the configuration details Each column heading in the tables is link By clicking on the column heading the table will be sorted by the selected column When sorting ascending or descending arrows will be displayed next to the column heading The alternating Hide disabled IP lines and Show disabled IP lines buttons are used to respectively hide or show the IP lines that have not been activated with a feature key To enable the lines install a feature key from the Feature Keys page By pressing on the IP line link in the Available IP Lines column the IP Line Settings page specific for the current IP line is opened This page offers a group of manipulation radio buttons that allows you to enable the IP line and to configure it to for use by the SIP phones Inactive this selection disables the corresponding IP line SIP Phone this selection configures the IP line for a SIP phone to be connected to the QX IP PBX s Phone Model drop down list is used to select the IP phone model to be used by the receptionist The drop down list excluding Other selection enables the MAC address t
63. be defined here The Action drop down list is used to select the defined user s permissions allow or deny to use the Paging Group service for the extensions included in the Paging Group table ACD Group Extension Settings For ACD Group extensions the Extensions Management Edit Entry page consists of General Settings SIP Settings SIP Advanced Settings ACD Group Settings and ACD Agents Table pages The SIP Settings and SIP Advanced Settings pages are the same as for the regular extensions described above The General Settings page is described below 1 General Settings for ACD Group extension This group requires ACD group extension s information and has the following components Extensions Management Edit Entry Extensions O Go Back Display Name is an optional parameter used to recognize the ACD Group Usually the display name appears on the called party s General Settings 700 phone display when a call is made or a voice mail is sent This information is also displayed in the ACD Management Groups table stings Display Name Subject ACD700 Password Generate Password ACD Agents Table Password requires a password for the ACD Group extension Percentage of Total Memory Fig ll 52 Extensions Management Edit Entry General Settings page for ACD Group extension The extension password may only contain digits If non numeric symbols are entered the Incorrect Password no symbol characters all
64. be used which plond eer Di Deco Me EREE Wo e chosen may produce files compatible to the CSV format the speed dial code and e destination should be separated by commas There should be a line break after each code defined The View Download Speed Dial Directory and Remove Speed Dial Directory links appear only if a global speed dial configuration file is uploaded previously Fig lIl 148 Global Speed Dial Directory page QX50 QX200 QX2000 SW Version 6 0 x 100 QX50 0X200 0X2000 Manual II Administrator s Guide The View Download Speed Dial Directory link is used to download the configuration file to the PC and opens the file chooser window where the saving location may be specified The Remove Speed Dial Directory link is used to restore the default configuration The speed dial configuration file downloaded from the QX IP PBX is in the CSV format To use the global speed dialing rules user should simply dial the speed dial code assigned to that speed dialing rule The call will be parsed through the rules of Call Routing Table To create a new Call Routing rule ony Se Sr S 15 16 17 19 20 21 22 23 24 25 26 Click on the Call Routing Table tab on the Call Routing page Press the Add button on the Call Routing Table page Specify the Pattern in the corresponding field Select the Number of Discarded Symbols and Prefix if required Select the Destination Type from the drop down list Define the
65. being called is disconnected The Busy Tone Duration drop down list is used to select the period in seconds when a busy tone will be transmitted to the FXS port e The Enable Power Disconnect Indication checkbox enables the power cycling on the FXS line when the remote party being called is disconnected Power Disconnect is applied after the busy tone transmission on the FXS line The Disconnect Duration drop down list is used to select the period in milliseconds when the FXS line power will be down The Ringer Type drop down list allows you to select the frequency of the ringer supported by the phone attached to the line Information can be found on the phone enclosure or in the phone s manual Problems with the ringer might occur if the ringer type selected here does not correspond to the one supported by the phone Please Note The supported ringer type can be found on the bottom of the phone in the Ren x xN value where N is the ringer type supported by the phone For example if N A the TypeA ringer type should be selected if N B the TypeB amp Z ringer type should be selected The Enable off hook Caller ID checkbox enables Caller ID transmission to the phone in the off hook state attached to a certain line Service is applicable to the phones supporting the Call Waiting Caller ID feature The Enable Hot Desking Capability checkbox is used to enable the Hot Desking feature on the corresponding onboard analogue FXS line Please Note Wh
66. built Interface UN y i a Y Show also as readable values Interface drop down list available only for QX50 QX200 offer ee the values abad a Receive Errors Y Transmit Errors e WAN Wide Area Network WAN events only Receive Drop ors WD Trant Drop Errors Receive Overrun Errors Transmit Collisions e LAN Local Area Network LAN events only B Received Multicast Packets Transmit Cater Errors When Show also as readable values checkbox is selected an additional table with statistics values will be displayed on the next page Fig IT 258 Transfer Statistics page The area Receive Values provides the following e Receive Bytes number of received bytes Overview System Status Events Conference History LANAVAN Statistics Network Transfer PSTN Channel Usage e Receive Packets number of received Ethernet packets tinas oh he LAN ia Current System Time Wed Aug 6 14 39 54 2014 e Receive Errors number of received packets containing Time range last 24 hours relative display errors Bytes Packets Errors e Receive Drop Errors number of received packets that have been discarded e Receive Overrun Errors number of received overrun errors that occur when the receive buffer is not large enough to hold all incoming packets This error usually appears due to a slow receiving system e Receive MultiCast Packets number of received cee broadcast packets een ile number of received bytes MW number of dropped p
67. by a moderator from GUI In this case even though the moderator activated the conference and did not join within the first X minutes the conference will be closed In all the above mentioned cases the conference will be closed regardless of the number of regular participants already joined Close the conference if only one participant is connected if enabled then the conference will be closed as soon as there is only one participant connected to the conference after the moderator left the conference If the moderator did not join yet during the first X minutes as described above the conference will stay active even if there is only one participant connected yet If the moderator is the only participant connected to conference then it will stay active Play notification before Conference close When the Max Duration M of the conference is reached the system will close the conference and M minutes before closing the conference the system will play the warning message to all participants Recording Settings The settings on this page are addressed to the conference recording configuration enabling conference recording defining the recording memory allocation internal or external storage etc The Recording Settings page offers the following components QX50 QX200 QX2000 SW Version 6 0 x 188 The Enable Recording checkbox enables an option to be used for active conferences to perform the online recordings With this checkbox sel
68. by subnet mask it could be up to 508 WINS Server defines a WINS server IP address for the QX IP PBX LAN users DHCP Advanced Settings link leads to the page where the advanced options of the QX IP PBX s DHCP server can be configured The Special Devices table on this page allows you to set a static IP address binding on the MAC address of the device in the QX IP PBX s LAN When this table is configured the devices with defined hostnames and MAC addresses will always get the same LAN IP address from the DHCP server Otherwise devices not listed in this table will get dynamic LAN IP addresses This table is also displayed in the System Configuration Wizard Add functional button opens an Add Host page where a new Static MAC address binding can be defined The page consists of the ove following components Se DHCP Settings for the LAN Interface Add Host Hostname text field requires the hostname of the device in the QX IP PBX s LAN Hostname Aastra6739i MAC Address text fields require the MAC address of the device in Wee PA the QX IP PBX s LAN o EE Static IP Address text fields require a fixed IP address of the device in the QX IP PBX s LAN Please Note If you leave this field empty the device in the QX IP PBX s LAN will get the first available IP address from range defined in the DHCP Settings page see above Fig II 207 DHCP Settings for the LAN Interface Add Host page QX50 QX200 QX2000 SW Version 6 0
69. call Y Start Recording Automatically Save Fig II 298 Recording Settings page Recording Indication selection enables voice announcements played in the conference to inform participants that the conference recording is started stopped paused or resumed When the Start Recording Automatically checkbox is selected the conference recording will start automatically as soon as the corresponding conference is activated Customization The Customization page is used to manage the voice prompts played during an active conference The page offers the following options When the Play First in Conference message checkbox is selected the system will play a You are the first participant in the conference notification message informing you that no more participants are yet connected Welcome Message parameters group allows updating the active conference welcome message played once a user is connected to the conference downloading it to the PC or removing the custom welcome message The group offers the following components Upload new welcome message indicates the file name used to upload a new welcome message The uploaded file needs to be in PCMU wave format otherwise the system will prevent uploading it and the Invalid audio file or format is not supported warning message will appear The system also prevents uploading if there is not enough memory available for the corresponding conference and the You do not have enou
70. calls in the queue when being on the line and to answer calls in the order they have been received The usage of this service is not limited to receptionist only and can also be used by the extension user if configured correspondingly The configuration of the Call Queue feature is done from the Extensions Management Edit Entry page where the length of the call queue and the call queue appearance is defined When the Call Queue service is enabled the second arriving call to the receptionist extension user will be either set into the queue if call queue appearance is 1 or will be ringing in the background of the active call if call waiting is enabled for the user and the call queue appearance value is greater than 1 If the call ringing in the background isn t answered it will be transferred to the user s voice mailbox or if no answer forwarding is enabled it will be forwarded to the corresponding destination If the call is set into the queue the caller will hear a message asking them to wait until the call will be answered Once the receptionist or extension user terminates the call the next call in the queue will ring to the user For regular FXS users indication about the callers in the queue is through the Call Waiting service see Manual III Extension Users Guide When a new caller arrives to the call queue the phone display if available of the phone connected to the FXS will display the total number of callers in the queue along w
71. defined The Access List of Extension page lists all users or a group of users if a wildcard is used and the appropriate permissions to use ja PARRA ere 299 the Paging Group through the corresponding extension Add Edit Delete PBX SIP 11369 sip epygi loc Fig IT 50 Access List of Extension page for Paging group The Add functional button opens an Add Entry page where a new user with corresponding permissions might be created This page consists of the following components Add Multiple Extensions Bulk Import Call Type lists the available call types B extensions ACCESS List Add Entry e PBX local calls from QX IP PBX s extensions e SIP calls through a SIP server Address Action wildcard supported e PSTN calls from global telephone network e Auto used for undefined call types The destination independent on whether it is a PBX number SIP address or PSTN number will be parsed through Call Routing Table Fig II 51 Access List of Extension Add Entry page for Paging Group The Address text field is used to define the address to be included in the Access List table The value in this field is strictly dependent on the Call Type defined in the same named drop down list If the PBX call type is selected the QX IP PBX extension number should be defined in this field For the SIP call type the SIP address should be defined for the PSTN call type the PSTN user number should
72. depending on the selected signaling type while selecting the B meta ESE corresponding trunk s checkbox and pressing Edit will open the ae See E Trunk Edit Entry page E1 T1 Stats link is displayed for every Trunk 1 919236874 127 5060 active trunk on the board and refers to the page where E1 T1 trunk and traffic statistics can be viewed E1 T1 Trunk Settings Start and Stop functional links are used to start shutdown the selected E1 T1 trunk s When E1 T1 trunk is shutdown state no E1 T1 calls could be placed and received Fig ll 112 E1 T1 Settings page The Trunk Edit Entry page consists of the following components Overview Pines PXS FXO E T Trunk ISDN Trunk PSTN Gateways a Trunk 1 192 168 74 127 5060 Edit Ent The Interface Type drop down list gives an option to choose between eae E1 T1 User and Network interface configuration ma a The Signaling Type drop down list allows selection of CAS Channel Associated Signaling or CCS Common Channel Signaling signaling types The same timeslot is used both for voice and data transmission in case of CAS signaling In the case of CCS signaling a single timeslot is Coding Type used for signaling data transmission on the entire trunk All other re timeslots are used for voice transmission Clock Mode The E1 and T1 radio buttons are used to select between E1 and T1 modes The T1 mode enables 24 timeslots and the E1 mode enables 32 timeslots to be used The selection
73. details link in the Details column for the calls Ie that contain T 38 FAX transmission A Mand Gi Urmuccesta Oupoing Cas CORSatngs COR Acti Aching Satngs ed FAX Statistics The FAX statistics page provides information about received and transmitted packets lost bad and duplicated packets This statistics a de minita E refers only to the T 38 FAX transmission The FAX statistics is not O ddr available for the FAX transmitted with other protocols BEA Report Fig II 252 FAX Statistics page Conference History In the Conference History page the calls are classified by conferences The Conference Call History sent via 3PCC Radius email or FTP is the same as is it shows only the PBX calls not sorted out by the conference The Conference History page consists of four tables They provide information on Conferencess Successful Unsuccessful Outgoing Conference Calls and Conference History Settings Conference History allows the collecting of conference call events on the QX IP PBX with their parameters and to search them by various criteria Only the administrator is allowed to enable or disable the conference statistic services Conferences The Conferences page lists all Conference Calls and their parameters ConfID Activation Time Conference Duration ae ven oy Participant Count Activation Reason and Activation Details n Successful Calls Unsuccessful Outgoing Calls Settings Each column he
74. during the caller is in the queue after which the call in the queue will be automatically redirected to the destination defined below QX50 QX200 QX2000 SW Version 6 0 x 29 e C pyg l QX50 QX200 QX2000 Manual II Administrator s Guide Call Type lists the available call types e PBX local calls to QX IP PBX s extensions e SIP calls through a SIP server e PSTN calls to a global telephone network e Auto used for undefined call types The destination independent on whether it is a PBX number a SIP address or a PSTN number will be reached through the Call Routing Table The Address text field is used to define the address where the call will be redirected The value in this field is strictly dependent on the Call Type defined in the same named drop down list If the PBX call type is selected the QX IP PBX extension number should be defined in this field For the SIP call type the SIP address should be defined for the PSTN call type the PSTN user number should be defined here For the Auto call type a routing pattern needs to be defined The ZeroOut Redirection radio buttons are used to enable the call redirection to the extension voice mailbox or other destination after some time spent in the queue This will avoid the caller to wait in the queue for too long e TheVoice Mail radio button selection allows the user to redirect the call to the extensions voicemail e The second radio button selection allows the callers to
75. end users Extended Warranty Statement Epygi Technologies LTD extends its Limited Warranty for an additional period of three 3 years from the date of the termination of the original Limited Warranty period proof of purchase required Epygi reserves the right to revise or update its products pricing software or documentation without obligation to notify any individual or entity Please direct all inquiries to Epygi Technologies LTD 1400 Preston Road Suite 300 Plano Texas 75093 QX50 QX200 QX2000 SW Version 6 0 x 2 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide Administrative Council for Terminal Attachments ACTA Customer Information This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA Located on the equipment is a label that contains among other information the ACTA registration number and ringer equivalence number REN If requested this information must be provided to the telephone company The REN is used to determine the quantity of devices which may be connected to the telephone line Excessive REN s on the telephone line may result in the devices not ringing in response to an incoming call In most but not all areas the sum of the REN s should not exceed five 5 0 To be certain of the number of devices that may be connected to the line as determined by the total REN s contact the telephone company to determine the maximum REN for the callin
76. field before being sent out to the external network The value in this field will be used in the MX record in the DNS server on the QX IP PBX The table on this page lists aliases for each of the device in the QX IP Fig II 213 DNS Server Settings page PBX s LAN to be resolves through the DNS server Zone epygi config loc Add functional link opens the page Add Host where a list of aliased can be defined for the certain device in the QX IP PBX s LAN The page contains the following components cl a i ca IP Address text fields require the IP address of the device in the QX Padars hoz Lo Jo Lio IP PBX s LAN Network me QXIPPBX Hostname text field requires the hostname of the device in the QX IP PBX s LAN Alias text fields are used to enter up to 5 alias names by which the device in the QX IP PBX s LAN will be resolved Fig IT 214 DNS Server Settings Add Host page Dynamic DNS Settings The Dynamic DNS DynDNS is a service that is used to map a dynamic IP address to a host name This service is used if you are connected to the Internet with a dynamic IP address and PPP DHCP client and want to allow access from the Internet to a device behind the firewall For example if you want to run your own WEB server To enable the DynDNS service on QX IP PBX you first have to choose a DynDNS provider and register at their website The Dynamic DNS Settings page provides the following components PPP PPTP S
77. installed It is used to enable custom language for QX GUI or revert back to the default language English QX50 QX200 QX2000 SW Version 6 0 x 9 e C Pyg l QX50 QXK200 QX2000 Manual II Administrator s Guide Administrator s Menus Setup Menu The Setup Menu consists of the following sections e Basic Setup For QX50 QX200 System LAN ee For 0X2000 System Configuration Wizard Overview Basic Setup For QX50 QX2 00 Internet WAN l o l System LAN Configure LAN interface and regional settings Internet WAN Configure WAN interface settings and adjust connectivity with external network For QX2000 Uplink Configuration Wizard 0 p Date and Time Configure time server and or time client Email settings for automatically generated emails events voice mails etc SMS settings for automatically generated text messages to mobile phones Date and Time System Security Email SMTP Security Settings Set security level to Low Medium or High for all passwords used in the system g Licensed Features Sho rt Text M e SSa gin g S M S Feature Keys List of licensed features that may be activated by installing a software license key Free Trial Activate a one time trial on one or many licensed features S Y stem Secu rit y Redundancy Redundancy Configure a redundant high availability system using a master and backup units connected through LAN ports Licensed Features Language Pack Upl
78. is pressed FXO Channel Usage Statistics chart appears It represents dependency between the time frame and the number of calls performed during that period Additionally it may display the maximum number of calls performed in the selected time frame QX50 QX200 QX2000 SW Version 6 0 x Status epygi FXO Channel Usage Statistics Y FXO1 Y FXO2 Y FXO3 Y FXO4 Time range of statistic table Intraday Y Incoming Calls Outgoing Calls Maximum Active Calls FXO Channel Usage Statistics Current System Time Wed Aug 6 14 44 39 2014 FXO 1 Statistics Time range last 24 hours relative display Call 15 16 17 18 19 2 21 2 2 0 Axis 1 Call otal number of incoming calls otal number of outgoing calls Maximum number of active calls Fig II 260 FXO Channel Usage Statistics page Fig II 261 FXO Channel Usage Statistics page 159 epy8l Maintenance Menu The Maintenance menu allows you to configure the following settings e Diagnostics Security Diagnostics Call Capture Ping Traceroute e System Logs System Logs Settings Remote Logs Settings Logs Archive e User Rights Management Users Roles e Backup Restore Automatic Backup Download Legible Configuration Upload Legible Configuration e Firmware Update Upload Firmware Get Firmware From Server Automatic Firmware Update e Reboot QX50 QX200 QX2000 SW Version 6 0 x amp Mai
79. is used to assign the corresponding permissions to the users Users The Users page contains a table where the Administrator and Local Administrator users are listed This page allows them to modify the passwords of available users in the table and to manage the Local Administrator s account Two levels of QX IP PBX GUI administration are available Administrator this is the main administrator s account The administrator can configure to have the factory reset safe the default password or choose not to The administrator has access to all Web GUI pages and no one else has configuration permission to adjust this account The administrator is responsible for granting access to all other user groups Local Administrator this is a common sub administrator s account The password is not factory reset safe Local Administrator can have permission to adjust each GUI page 0 Extension this account refers to all extensions created on the QX IP PBX The password for default extensions is not factory reset safe but is contained in the backed up configuration Permissions for an extension to access each GUI page can be adjusted here The following functional buttons are available on this page User Rights Backup Restore Firmware The Change Password functional button is used to change the password of the Administrator and Local Administrator user s account Select one of the available users in the table by toggling ae the corre
80. its characteristics generate the automatic i Intoriaces AN configuration file and will upload it to the SIP phone The SIP E Enable firmware version control phone will be then configured on the first available IP line of tien oa the QX IP PBX and will become completely functional Phones Default Template systemdefault Y Please Note The Plug and Play service is only available for the supported SIP phones see the list below This service will not work in case the SIP phone is already manually configured or if it is not reset after enabling the Enable PnP to IP lines checkbox Fig II 99 IP Line Settings page Enable Firmware Version Control checkbox is used to control the firmware version running on the SIP Phone attached to the QX IP PBX This service also allows you to have the new firmware automatically downloaded and installed on your SIP Phone in case your SIP phone was running an old firmware upon connecting to the QX IP PBX or when the QX IP PBX s firmware has been updated and the compatibility was changed to the higher firmware version of the SIP phone Every new firmware of QX IP PBX is compatible to a certain firmware version of each supported SIP phone If you are running older firmware on your SIP phone this service will automatically download and install the newer firmware on your SIP phone Please Note The Firmware Version Control service is only available for snom and Aastra SIP phones Attention Do not select this checkbo
81. l QX50 QXK200 QX2000 Manual Il Administrator s Guide Diagnostics The Diagnostics page gives a possibility of running Network protocol diagnostics to verify QX IP PBX s connectivity and to download all system logs for possible problems recovery The Start Network Diagnostics button is used to initiate network diagnostics i e to check the WAN link and IP a configuration to verify gateway DNS primary and secondary if A i gears Diagnostics configured servers accessibilities Start Network Diagnostics Start FXO Diagnostics Download system logs The Start FXO Diagnostics button available only for peste tes QX50 QX200 runs FXO diagnostic tests to determine the erecting configuration tica optimal value for the FXO country specific regional setting A uaa primary nameserver 292 188 0 1 Fan CSRS appropriate to your PSTN provider Once the FXO orla arar Ehe aiii diagnostic is complete the recommended value should be set emcees Parlin and ansera manually on the fxocfg hidden page Setting this value may rar vela botas Gies tc occ ks de et www epygi config loc resolved resolve echo or poor audio quality issues on FXO lines The Download system logs button is used to download all logs to the local PC as a tar archive file These logs can then be used by the Epygi Technical Support Office to determine the problem that has occurred on your QX IP PBX Fig II 263 Diagnostics page The field below
82. levels for the corresponding agent should be configured The Agent Skill Levels page consists as many drop down _ lists as Skills created in the Skills page see below For each available Skill you should select the skill level from 0 to 10 with 0 meaning the absence of that specific skill and 10 meaning the highest level matching to the corresponding agent The Groups page of ACD Management contains a list of ACD Group type extensions filtered from the Extensions Management table This page allows you to configure the ACD Group specific parameters i e a collection of agents included to the group call queue and the call distribution mechanism Any new ACD Group created in this page will automatically be displayed in the Extensions Management table QX50 QX200 QX2000 SW Version 6 0 x Extensions amp Extensions Agent 103 Skill Levels O Go Back Skill french Level 3 Skill english Level 5 Skill spanish Level 10 highest Y Skill support Level 5 Save Fig II 91 ACD Management Agent Skills page Overview Extensions Conferences Skills Agents ACD Management Add Edit Delete Group ID 104 11369 1111 108 Sales Department Tech Support 105 Marketing Department 11369 Fig II 92 ACD Group Management page 63 QX50 0X200 0X2000 Manual II Administrator s Guide Add opens the Add Group page where a new ACD Group may be created The Add Group page incl
83. managing several routing rules with the single Key The second page of the Call Routing Wizard offers different components depending on the Destination Type selected on the NE al CG EG A GEES previous page Call Routing Wizard O Go Back Use Extension Settings drop down list is applicable to SIP and IP m PSTN destination types and allows you to select the extension as di also Auto Attendant on behalf of the call that will be placed The SIP settings of the selected extension will be used as the caller information If an entry is not selected from this list the original caller information will be kept When Keep original DID checkbox is selected the called destination will receive the original caller s se information and not the information of the extension selected from o the Use Extension Settings list When the checkbox Add Remote Party ID is selected the Remote Party ID parameter is being delivered to the destination side upon call establishment procedure SIP Tunnel drop down list appears only when the SIP_Tunnel Destination Type is selected on the previous page The list is used to select the particular SIP tunnel to route the calls through the corresponding QX IP PBX A Check with 3PCC Fig II 139 Call Routing Wizard page 2 Destination Host requires the IP address or the host name of the destination for a direct call or the SIP server for calls through the SIP server This field
84. more than 6dB then the Conference Server will switch the video to a new source having the largest voice energy Leave Active checkbox will keep conference active even if all participants have left it Close the Conference if Moderator did not join in the idea of including this parameter is as follows If the conference is activated by one of the existing ways and the moderator does not join the conference within the first X minutes then the conference will be closed by the system No message will be played to the joined users in this case The conference will be closed in one of the following cases O The conference is activated by a schedule and the moderator did not join within the first X minutes after activation The only method of distinguishing the moderator from the other participants is the moderator s password If the user entered the moderator s password during the joining process then he she is a moderator There are no other means of distinguishing the moderator from the regular participant O The conference is activated by a participant when dialing in and the Activate On Dial In checkbox is enabled for that conference During the joining process the participant either did not enter any password or entered a regular participant s password In this case the same as above if the moderator did not join the conference within the first X minutes entering moderator s password the conference will be closed O The conference is activated
85. of the Administrator An error message prevents entering the wrong password Change Password Account Name admin GUI Access Password Phone Access Passwor d e The New Password text field requires a new password Od Password for the Administrator or Local Administrator i maintenance VPO e Reentering the new password in the Confirm New i Password text field will confirm the new password The New Password field is checked against its strength and you may see how strong is your inserted password right below that field Fig II 275 Change Password page QX50 QX200 QX2000 SW Version 6 0 x 165 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide Please Note The password can consist of numeric values only Up to twenty 0 20 digits are allowed A corresponding warning appears if any other symbols are inserted The Enable User and Disabled User functional buttons are used to enable or disable the Local Administrator s account Attention It is highly recommended to define a proper and non empty password on this page if the extension is being used for the Call Relay service from the QX IP PBX s Auto Attendant Roles The Roles page contains a table where the Local Administrator and Extensions users are listed This page allows you to set the permissions to the GUI pages for each user in the table User Rights Management The Edit functional button leads to the Change Access Rights me page where
86. of the RTP channel 7 oahuscchane Fig II 170 RTP Streaming Channel page The Port Number text field requires the broadcasting RTP port number The Description text field requires optional information related to RTP Streaming Channels Add Entry the RTP streaming channel O Go Back Telephony Fig II 171 RTP Streaming Channel Add Entry page Gain Control The Gain Control settings are used to define transmit and receive gains The Gain Control page offers Transmit Gain and Receive Gain drop down lists for each line that contains allowed gain values which can be set up by the administrator for every line QX50 QX200 QX2000 SW Version 6 0 x 114 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide For FXS lines Transmit Gain defines the phone speaker volume on the call Receive Gain defines the volume of the phone microphone on the call For FXO lines Voice Mail RTP Streaming Channels 3PCC Radius Client Dial Timeout Call Quality Notification Transmit Gain defines the level of voice transmitted from QX IP PBX to the FXO network Gain Control Restore Default Gains Telephony ere Receive Gain defines the volume of voice received by QX IP PBX from the FXO network Transmit Gain 6 Y Receive Gain 0 Y For Voice Mail Recording Gain defines the volume of the phone microphone upon playing voice mails or system messages Playback Gain defines the phone speaker v
87. of the following components S Bulk Import Extensions Management Add Entry Extensions The Extension text field is used to enter a new extension number If non digit symbols have been entered the error Incorrect Extension no symbol characters allowed will appear If an extension with the same number already exists in the Extensions Management table the error Extension already exists will appear Fig II 25 Extensions Management Add Entry page Please Note Extension number cannot start with the digits 0 You can add extensions of up to 20 digits long However the Call Routing Table won t be adjusted automatically you may need to manually adjust the routing rules for extensions in custom length The Type drop down list is used to select the type of the extension to be created for details see below The following values are available in this list Attendant User Extension Pickup Group Call Park Paging Group e ACD Group if the ACD feature is previously activated from Feature Keys page e Recording Box if the Call Recording feature is previously activated from Feature Keys page QX50 QX200 QX2000 SW Version 6 0 x 24 epygl User Extension Settings 1 General Settings This group requires extension s personal information and has the following components Extensions Add Extension Add Multiple Extensions Bulk Import Display Name is an optional parameter used to recognize the c
88. page 3 Confirm the deletion by pressing Yes The IP range will then be deleted To abort the deletion and keep the IP range in the list press No RTP Settings The RTP Settings page allows the administrator to configure the codec s packet size and silence suppression for each voice codec All parameters listed on this page may be modified and submitted The Codec Properties table lists all codecs with the corresponding packetization interval and information about silence suppression Edit opens the Edit RTP Settings page where the codec settings can be modified To use Edit only one codec may be selected at a time otherwise the One record should be selected error message appears The Packetization Interval is the time interval between two RTP packets of the same stream If the interval is increased the overhead is decreased but the voice quality may deteriorate as a result If the interval is decreased the network load is increased RTP Setti and the delay is reduced ettings Codec Properties Edit Silence Suppression disables RTP packet transmission in case of y Tomato Codecs Packetization Interval Silence Suppression no voice activity This feature helps to avoid extra traffic ifthe RTP G11u PCM audio coding standard 8 Ke sample rate its Gt KV data rate 20 ms stream contains no voice activity It is activated after two seconds TTE e e ee ee ee z of silence and restarted immediately if any audio appears 675 24
89. period the action has failed but will continue to try When the required action is successful QX IP PBX raises an appropriate message To Assign an Action to the Event 1 Select the checkbox of one or more events to assign an action to them 2 Press the Edit button The Edit Event Settings page appears 3 Select an action type from the Action radio buttons to notify the administrator about the event 4 Press the Save button to submit the changes or use Go Back button to abort the selected action Call History The Call History page provides information on Successful Missed Unsuccessful Outgoing Calls Call History Settings Archive and Archiving Settings Call History allows the collecting of call events on the QX IP PBX with their parameters and to search them by various criteria The selected number of statistics entries will be displayed in the Call History tables The Call History page reports successful non successful and missed incoming outgoing calls and shows the Call History settings Only administrator is allowed to enable or disable the call statistic services Successful Missed and Unsuccessful Calls The Successful Calls Missed Calls and Unsuccessful Outgoing Calls pages lists successful missed and unsuccessful incoming and outgoing calls and their parameters Call Start Time Call Duration Calling Phone and Called Phone Each column heading in the tables is created as a link By clicking on the column heading the tabl
90. phone IP Phone Templates The Manage IP Phone Templates page is used to create custom templates for the IP Phones The templates contain a set of configuration settings that are uploaded to the IP phone once it is registered on the QX IP PBX With the custom templates the most popular configuration settings may be adjusted accordingly The saved custom templates can be then configured from the Edit IP Line Settings page to be used on the particular IP phone The Manage IP Phone Templates page consists of a table where the available IP phone templates are listed The systemdefault template in this table indicates the QX IP PBX default template for all IP phones This template cannot be edited or deleted Add opens the Add Entry page where an IP phone template can be created The Add Entry page includes the following text fields e Template Name text field indicates the name of the template This name will be visible in the Edit IP Line Settings page when defining the template for the IP phone e Description text field requires optional information about the template Edit opens the Manage IP Phone Templates Edit Entry page where the selected template s settings can be adjusted F1 T1 Trunk ISDN Trunk PSTN Gateways Ov S IP Lines IP Line Settings IP Phone Templates IP Phones Logo FXS Gateways The Manage IP Phone Templates Edit Entry page allows configuration of multiple IP phones The IP phones templates Manage IP Phone Templates he
91. proper functionality of the IP phone However if you wish to IE TT reboot the IP phone later leave this checkbox unselected MAC Adderss 00 15 85 4574 19 Attached IP Lines 3 Description Epygi Receptionist Mapped IP Lines To Keys Not specified Watched Extensions 104 302 NOTE You must restart the phone before the new settings will take effect WARNING After deleting this receptionist record call queue settings of the extensions to which the lines are attached will be reset to their default values v Reboot IP phone now Previous Finish Fig II 86 Receptionist Phone Configuration Wizard Summary page ACD Management Attention The Automatic Call Distribution is an optional feature and can be activated with a feature key from the Feature Keys page Automatic Call Distribution ACD is the contact center solution designed for queuing and automatic distribution of the calls between contact center agents ACD concept and the contact center solution are based on the following building blocks Agent a call center user reachable via QX IP PBX Agent Group AG comprises the call queue collection of agents call center users and call distribution mechanism between its agents QX50 QX200 QX2000 SW Version 6 0 x 60 QX50 0X200 0X2000 Manual II Administrator s Guide s Interactive Voice Response system IVR a custom Auto Attendant on QX IP PBX answering the calls from remote callers customers
92. provided by the host itself QX50 QX200 QX2000 SW Version 6 0 x 131 QX50 0X200 0X2000 Manual II Administrator s Guide DHCP Settings for the VLAN Interface DHCP Settings for the VLAN Interface is used to establish virtual networks in the QX IP PBX s LAN or to integrate the QX IP PBX into the corporate network s virtual LAN WAN DHCP service can be activated both on LAN or WAN interfaces VLAN is useful in corporate companies to divide large networks into groups and to have devices like QX IP PBX s and IP phones in each network separated for example to separate networks for data and voice transmission Priorities may be assigned to the interfaces for packets prioritization With VLAN configuration each virtual network will be characterized with a VLAN ID tag Packets addressed to that network will be checked towards the ID and if the ID number defined in the incoming packets matched the corresponding network s ID the packets will be accepted Otherwise if the ID does not match the packets will be dropped In the same way if the QX IP PBX is integrated into the network that uses VLAN technology outgoing packets should have the ID number of the corresponding virtual network for the remote party to accept the packets from the QX IP PBX The DHCP Settings for the VLAN Interface page contains a table with all enabled VLAN interfaces created in VLAN Settings page see below and the corresponding parameters VLAN ID IP IP Routing DH
93. proxy is specified for an extension all SIP calls originating from that extension are made through that outbound proxy i e all requests are sent to that outbound proxy even those made by Speed Calling The Secondary SIP Server acts as an alternative SIP registration server when the primary SIP Registration Server is inaccessible If the connection with the primary SIP server fails QX IP PBX will automatically start sending SIP messages to the Secondary SIP Server It will switch back to the primary SIP server as soon as the connection is reestablished Authentication User Name requires an identification parameter to reach the SIP server It should be provided by the SIP service provider and can be requested for some SIP servers only For others the field should be left empty Send Keep alive Messages to Proxy enables the SIP registration server accessibility to the verification mechanism Timeout indicates the timeout between two attempts for the SIP registration server accessibility verification If no reply is received from the primary SIP server within this timeout the Secondary SIP server 2 W Extensions Management Edit Entry will be contacted When the primary SIP server recovers SIP a R packets will resume being sent to it Add Extension Add Multiple Extensions Bulk Import SIP Advanced Settings The RTP Priority Level drop down list is used to select the priority Si A low medium or high of the RTP packets sent from a ces
94. rate 8 bits 64 kbit s data rate Enabled coding at 16 kbit s rate Disabled When establishing a call the system will try this codec first If the TT coding at 32 kbit s rate Disabled coding at 40 kbit s rate Disabled remote party does not support the preferred codec the following ee codecs will be tried out strictly in the order given in the Codecs table Please Note Pay attention when configuring Auto Attendant Codecs as they are used by virtual extensions for redirecting the incoming calls Enable Disable enables or disables the selected codec Disabled codecs do not participate in codec negotiation i e they will never be iLBC internet Low Bit Rate Coder at 13 33 kbit s rate G 722 HD audio coding at 48 64 kbit s data rate 16 kHz sample rate G 722 1 HD audio coding at 24 32 kbit s data rate 16 kHz sample rate TDVC Time Domain Voicing Cutoff at 1 95 kbit s rate Video Codecs H 263 Video coding for low bit rate communication H 264 Advanced video coding for low bit rate communication H 263 Video coding for low bit rate communication Disabled Disabled Disabled Disabled State Disabled Disabled Disabled Out of Band DTMF Transport Enable T 38 FAX Y Enable Pass Through FAX Enable Pass Through Modem Force Self Codecs Preference for Inbound Calls used to for call setup At least one codec must be enabled otherwise voice communication with an extension attendant con
95. redirect the call to the specified destination instead of holding in the extension s queue The caller will then be automatically transferred to the destination specified in this page This selection activates the following fields to be inserted Call Type lists the available call types O PBX local calls to QX IP PBX s extensions o SIP calls through a SIP server o PSTN calls to a global telephone network o Auto used for undefined call types The destination independent on whether it is a PBX number a SIP address or a PSTN number will be reached through the Call Routing Table The Address text field is used to define the address where the call will be redirected The value in this field is strictly dependent on the Call Type defined in the same named drop down list Ifthe PBX call type is selected the QX IP PBX extension number should be defined in this field For the SIP call type the SIP address should be defined for the PSTN call type the PSTN user number should be defined here For the Auto call type a routing pattern needs to be defined wildcard is allowed in this field Please Note To activate the ZeroOut Redirection feature the caller should dial O digit Upload new call queue welcome message allows updating the active Call Queue welcome message played when a caller joins the extension s call queue downloading it to the PC or restoring the default one The Remove call queue welcome message functional link appears onl
96. rem corresponding extension RTP packets with higher priority will be A sent first in case of heavy traffic imeout sed 60 RTP priority level medium Y Y Do Not Use SIP Old Hold Method The Do Not Use SIP Old Hold Method checkbox enables the new TER recommended method of call hold in SIP in which case the hold request is indicated with the a sendonly media attribute rather oTolinesetinas Secondary SP Sever than with the IP address of 0 0 0 0 used before The checkbox should ee e be enabled if the remote party does not recognize hold requests Sc cs SP Eee initiated from the QX IP PBX mis A group of Host address and Port text fields respectively require the host address IP address or the host name and the port numbers of the Outbound Proxy Secondary SIP Server and the Outbound Proxy for the Secondary SIP Server These settings are provided by the SIP servers providers and are used by QX IP PBX to reach the selected SIP servers Fig II 34 Extensions Management Edit Entry Advanced SIP Settings page 4 Remote Settings This group is used to configure SIP Remote Extension functionality This is an advanced telephony feature that allows QX IP PBX users to remotely Operate QX IP PBX Users need to register a hardware or software SIP phone on the QX IP PBX by defining the QX IP PBX s global IP address and an appropriate Username Password A registered SIP Remote phone can act fully as a phone connected
97. s Guide During this period participants will be able to communicate with each other However this does not mean that the conference is activated the participants will be dialed out if any and the recording will start if configured only after the configured scheduled time comes The Send Mail before Conference Activation checkbox enables email notification delivery to the participants before the conference activation The text field requires the timeout in minutes before the conference activation when the email notifications to the conference participants with Email Address configured from the Add Participant page should be delivered This option is only valid ifthe Email Address is configured for the participant The Send Mail on behalf of text field requires an email address or a conditional name related to the conference to be transmitted in the From field of the email notifications Send Notification Mail This link is used to send an email to the participants notifying them about the start of a conference and inviting them to join The text of the notification email is being configured by the administrator QX50 QX200 QX2000 SW Version 6 0 x 193 QX50 0X200 0X2000 Manual II Administrator s Guide Appendix Software License Agreement EPYGI TECHNOLOGIES LTD Software License Agreement THIS IS A CONTRACT CAREFULLY READ ALL THE TERMS AND CONDITIONS CONTAINED IN THIS AGREEMENT USE OF THE QUADRO HARDWARE AND OPERATIONAL S
98. selected trunk s settings should be copied Tunk 3192 68 74108560 PIMP Point To Mut Poin e HEI C PON MM G 212 28 2 2 28 2 2 runk 4 192 168 74 109 5060 PTMP Point To Multi Poin to The Restore Default Settings functional link restores the default signaling settings of the selected ISDN trunk s Fig II 126 ISDN Settings page Clicking on the corresponding ISDN trunk will lead to the ISDN wizard where trunk s ISDN signaling settings can be configured The ISDN Wizard consists of several pages The ISDN Wizard ISDN Settings allows you to choose the interface type and the connection type of the selected trunk s The Interface Type drop down list allows you to select between the User and the Network interfaces If the ISDN port of the QX ISDN Gateway is connected to the CO then User interface type should be selected If the ISDN port of the QX ISDN Gateway is connected to the PBX then Network interface type should be selected in that case QX ISDN Gateway acts as a CO for that PBX QX50 QX200 QX2000 SW Version 6 0 x 85 QX50 0X200 0X2000 Manual II Administrator s Guide The Connection Type manipulation radio button group allows you to choose the connection type for the selected trunk s e PTP Point to Point In case of connection to the CO User interface type is selected on QX IP PBX choose this option if only QX is connected to the ISDN trunk from CO no other ISDN devices A EE iar are connected to th
99. system internal or external storage space depending on the configuration To use conference recording service it should be enabled from the Call Recording Settings page The maximum duration of the recorded conference can be optionally limited from the Recording Settings page Conference recording can be manipulated either from the Conference Progress page or from the handset see Feature Codes If the Recording Indication is also enabled from the Recording Settings page voice announcements will be played in the conference to inform participants that the conference recording is started stopped paused or resumed Recorded conferences are stored and are listed in the Recorded Conferences page accessible by the moderator from QX IP PBX Web Management The Recorded Conferences page displays a table where recorded conferences are listed The recorded conferences can be played and deleted from this page QX50 QX200 QX2000 SW Version 6 0 x 186 e C Pyg l QX50 QXK200 QX2000 Manual II Administrator s Guide The Recording free space field displays the free space allocated for the corresponding conference The New recordings field displays the number of new e recorded conferences in the recording box All new recordings are marked in bold Recorded Conferences Conference ID 700 Recording free space 2 day 20 hour 9 min 22 sec The All recordings field displays the number of all recorded conferences in the recording box
100. table Additionally two lists should be defined for the call park extension Park Access List for users that might park a call to the corresponding Call Park extension and Retrieve Access List for the users that can pick up calls parked to that extension By default both of these lists have entries so any PBX extension on the QX IP PBX can park the call and any destination can retrieve the parked call Any limitations to these settings should be done individually for each call park extension To make a Call Park To make a Call Park the QX IP PBX user which has been previously added to the Park Access List for at least one of the available Call Park extension on the QX IP PBX should dial the appropriate digit combination see Feature Codes in Manual III Extension User s Guide during the call The active call will go on hold while the PBX number and the SIP username if it is registered on the SIP server of the first available call park extension where the user is added will be played to him her The pickup user will be able to pick up the parked call from any destination by calling the extension where the call has been parked either by its PBX number or SIP address The authentication password will be prompted if configured of the call park extension in order to retrieve the parked call For example the Call Park extension 77 is created which has been registered on the SIP Server under the 892220 registration username The QX IP PBX user i
101. text field requires the mailing address of the IPSec connection partner The Send button will insert QX IP PBX s public RSA key into an e mail and send it to the IPSec connection partner Fig IT 228 IPSec Configuration RSA Key Management page PPTP L2TP Configuration PPTP Point to Point Tunneling Protocol is used to establish a virtual private network VPN over the Internet Remote users can access their corporate networks via any ISP that supports PPTP on its servers PPTP encapsulates any type of network protocol IP IPX etc and transports it over IP Therefore if IP is the original protocol IP packets ride as encrypted messages inside PPTP packets running over IP PPTP is based on point to point protocol PPP and the Generic Routing Encapsulation GRE protocol Encryption is performed by Microsoft s Point to Point Encryption MPPE which is based on RC4 L2TP Layer 2 Tunneling Protocol is a protocol from the IETF which allows a PPP session to run over the Internet an ATM or frame relay network L2TP does not include encryption as does PPTP but defaults to using IPSec in order to provide virtual private network VPN connections from remote users to the corporate LAN Derived from Microsoft s Point to Point Tunneling Protocol PPTP and Cisco s Layer 2 Forwarding L2F technology L2TP encapsulates PPP frames into IP packets either at the remote user s PC or at an ISP that has an L2TP remote access concentrator LAC The LA
102. that point forward and unless the user with log off the phone he may place and receive calls and use all the supplementary PBX services of the QX IP PBX QX50 QX200 QX2000 SW Version 6 0 x 75 epygl The Hot Desking feature is used to organize the user login logout on the public phones Each user should have a virtual extension configured in the Extensions Management table The virtual extensions can be configured as needed to use all the available supplementary PBX features when the user will log in from the phone with that extension The Hot Desking option should be enabled on the corresponding analog or IP lines from the IP Lines or FXS Lines page accordingly To login to the phone use the 9000 feature code for more details see Feature Codes chapter You will be prompted for the extension and the password When you login to the phone with your extension the phone becomes a fully featured phone connected to the QX IP PBX You may place and received calls with the SIP address configured in the Extensions Management page use Voice Mail services etc When you have finished using the phone logout with the 9000 feature code From that moment forward your extension becomes again virtual and is not connected to any analogue or IP line but it still can handle calls using Call Forwarding Many Extension Ringing Hunt Grouping etc services and voice mails according to the supplementary service configured on that virtual extension The phone
103. the following information e Uptime duration Period QX IP PBX is running since last reboot e Device hostname QX IP PBX device host name Status General Information e Application Software Software and file system versions of the QX IP PBX g l Uptime duration 1 hour 27 min 15 sec Device hostname QX200 12 e Language Pack this field is present only when the Application Software 5324 Release custom language pack is uploaded and it indicates the Gill status legeh JE version Fig II 236 Status General Information page Network Status The Network Status page includes the following information about Interfaces System Status Interface Name lists the Network interfaces available on the QX e y IP PBX LAN WAN and a number of PPPs depending on the d hee oa number of active PPP connections ay Interface name IP address Subnet Mask Properties Monitor Service Name Status WAN 192 168 74 12 255 255 255 0 MAC 00 FO 00 FO 80 61 Watch WAN NTP Server Running IP Address lists the IP addresses corresponding to each 3 LAN 172 30 4 1 255 255 255 0 MAC 00 FO 00 FO 80 60 Watch LAN NTP Client Running netwo rk interface DHCP Server for LAN Running Default Gateway 192 168 74 5 DHCP Server for VLAN Stopped DNS Server 192 168 0 11 Subnet Mask lists the subnet masks corresponding to each a soe As Stopped network interface DNS Running Firewall Low NAT Running Propert
104. the top down and routing will take place according to the first pattern that matches the dialed number The Move To button is used to move the selected entry to a different position in the Call Routing Table This will increase or decrease the selected pattern s priority Pressing the button will open the page where a row number should be specified together with the position the selected entry is to be placed before or after the defined row Call Routing The Call Routing page offers the following components e When the Route all incoming SIP calls to Call Routing checkbox is disabled for all incoming SIP calls QX IP PBX will first search the incoming SIP address in the Extensions Management table If found the incoming SIP call will ring on the corresponding extension If not found QX IP PBX will look for a matching routing rule in Call Routing Table When the Route all incoming SIP calls to Call Routing checkbox is enabled for all incoming SIP calls QX IP PBX will directly look for a matching routing rule in Call Routing Table and will ignore the possible matches in the Extensions Management table QX50 QX200 QX2000 SW Version 6 0 x Telephony Call Routing Table Local AAA Table Global Speed Dial SIP Tunnel Class of Service Call Routing Y Route all incoming SIP calls to Call Routing Regardless of whether the Route all incoming SIP calls to Call Routing checkbox is selected or not SIP calls from external callers will
105. this radio button is activated a button will be displayed in the main management window that PPP PPTP Settings serves to switch the Internet connection on off When accessing the Internet every station of the connected LAN has to connect to QX IP PBX first e Always connected QX IP PBX stays in the always connected mode This will allow always being online in the network IP Address Assignment radio buttons are used to define the IP address assignment for the PPP interface with the following options e Dynamic IP Address the IP address to the PPP interface will be assigned dynamically by the DHCP server e Fixed IP Address the fixed user defined IP address will be assigned to the PPP interface Dynamic IP Address Fixed IP Address The Keep Connection alive checkbox enables keeping the connection alive by sending control packets dedicated for the link state verification Fig II 216 PPP PPTP Settings page Advanced PPP Settings The Advanced PPP Settings page available only for QX50 QX200 is used to enable disable certain parts of the negotiation process during connection establishment These settings are available only if QX IP PBX has a PPPoE WAN interface Attention Disabling any of the services below may cause problems when establishing a connection including the complete connection failure The default settings should be changed only if the ISP Internet Service Provider specifically requires it or if the pe
106. to define a list of extensions that are capable to watch the current extension calls and to define the appropriate permissions a tale Watch Access List of Extension 103 Add Edit Delete This page contains the following functional buttons Add functional button opens the Watch Access List Add Entry page where extensions may be added to the Watch Access List The Watch Access List Add Entry page consists of the following components e Call Type lists the available call types PBX local calls to QX IP PBX s extensions SIP calls through a SIP server Auto used for undefined call types The destination independent on whether it is a PBX number or a SIP address will be reached through the Call Routing Table Fig II 31 Watch Access List page Watch Access List Add Entry Extensions e The Address text field is used to define the address where the call will be redirected The value in this field is strictly Call Type PBX Y dependent on the Call Type defined in the same named drop A down list If the PBX call type is selected the QX IP PBX aa ias extension number should be defined in this field For the SIP O Aw Dialog Subscription call type the SIP address should be defined For the Auto call type a routing pattern needs to be defined wildcard is allowed in this field Fig II 32 Watch Access List Add Entry page The checkboxes on this page allow to select one or more options of the Watch Access Lis
107. to drop down list Selecting the Use Default outgoing Caller ID allows you to overwrite the source caller information with the one specified in the Default outgoing Caller ID field when placing outgoing calls toward the CO The Default outgoing Caller ID field requires the caller ID for the outgoing calls from the QX through the ISDN trunk That number should be registered at the CO and can be one of the MSNs provided by the CO If this checkbox is enabled but no value is defined in the Default outgoing Caller ID empty caller information will be sent to the CO If this checkbox is disabled the source caller information will be forwarded to the CO FE Interfaces QX50 0X200 0X2000 Manual II Administrator s Guide E1 T1 Trunk ISDN Trunk PSTN Gateways ISDN Wizard Routing Settings Trunk 1 192 168 74 135 5060 MSN Number s Route Incoming Call to 11102 121 11103 111 11104 Routing with inbound destination number Y Use Default outgoing Caller ID Default outgoing Caller ID 297094 Advanced Settings Fig IT 129 ISDN Wizard Routing Settings page Select the Advanced Settings checkbox if you wish to adjust trunk s L2 and L3 Settings manually otherwise leave this checkbox unselected to use the system default values The ISDN Wizard L2 amp L3 Settings is used for advanced configuration only and contains L2 amp L3 Settings This page only appears when the Advanced Settings checkbox is selected on the pr
108. to select the remote server type TFTP or FTP In case of FTP selection the authentication username and the password need to be inserted In case if these fields are left empty anonymous authentication will be used The File Format drop down list is used to select the format in which Call History will be saved This list offers to choose between Tab Delimited Text log and Comma Separated Values csv file formats To Enable Disable the Call History 1 Enter the Call History Settings page QX50 QX200 QX2000 SW Version 6 0 x 153 pya QX50 0X200 0X2000 Manual II Administrator s Guide 2 Select or deselect the Enable Call Reporting checkbox to enable or disable statistics recording 3 If enabling the statistics the maximum number of records to be stored in the statistics table should be selected from the corresponding drop down lists 4 Press Save to apply the new configuration To Filter the Call History 1 Enter the desired criteria fields 2 Press the Filter button to search the call reports within the Call History table Please Note To return to the complete Call History table clear all search criteria and press Filter To Reset the Call History 1 Press the Clear All Records button in the Call History Settings page 2 Confirm the deletion by clicking on Yes The Call History will then be deleted To abort the deletion and keep the statistics information click on No RTP Statistics The RTP St
109. to the primary server Once the RADIUS server receives the request it determines if the sending client is valid A request from a client that the RADIUS server does not recognize must be silently discarded If the client is valid the RADIUS server consults a database of users to find the user whose name matches the request The user entry in the database contains a list of requirements username password etc that must be met to give access to the user If all conditions are met the user gets access to the QX IP PBX Network The RADIUS Client Settings page contains the Enable RADIUS Client checkbox that enables RADIUS client on the QX IP PBX Please Note The RADIUS Client cannot be disabled if there is at least one route with RADIUS Authentication and Authorization or RADIUS Accounting values configured in the AAA Required drop down list at the Call Routing Table In order to be able to disable the RADIUS Client on the QX IP PBX appropriate routes should be removed first The other RADIUS Client settings are divided into three groups 1 Registration Settings The Primary Server requires the IP address of the primary Radius EN A S erver Voice Mail RTP Streaming Channels Gain Control 3PCC Dial Timeout Call Quality Notification RADIUS Client Settings Y Enable RADIUS client The Secondary Server requires the IP address of the secondary Radius Server X Telephony Registration Settings NAT Station IP text fields require the NAT PC WA
110. to your IP phone You should then reboot your phone to make the new language pack active QX50 QX200 QX2000 SW Version 6 0 x 21 C Pyg l QX50 QXK200 QX2000 Manual II Administrator s Guide Extensions Menu The Extensions menu allows you to configure the following settings e Extensions i Add Extension Overview Extensions Overview Extensions Add Multiple Extensions ae View and manage all extensions Bulk Im ort 7 Peis Add ion Create a single extension Add Multiple Extensions Add many extensions at once E Conferences x o Bulk Import Add many extensions at once from a preconfigured list in CSV format D Conferences Conferences List conferences and access conference specific settings pages Add Conference Add Conference Create a new conference ID Email Defaults Configure notifications and activation emails for upcoming conference meetings Email Defaults Recordings Configure Music on Hold and other system messages e Recordings decian Directory Configure dial by name directory used by the Auto Attendant 6 Dire cto ry Receptionist Receptionist Configure IP phones with add on modules for additional keys o Rece ptionist Skills Configure the list of skills used by ACD for skills based routing Configure each agent of the ACD Groups Create ACD Groups and assigned agents e ACD Authorized Phones Authorized Phones Based on the caller ID incoming calls to the Auto Attendant can be
111. tone is idle the tone plays a ring splash IP Lines IP Lines IP Line Settings RUM QX50 0X200 0X2000 Manual II Administrator s Guide E1 T1 Trunk ISDN Trunk e Templates IP Phones Logo FXS Gateways Manage IP Phone Templates Add Entry FE Interfaces O Go Back Template Name Template Marketing Description Template for Marketing Department IP phones Save PSTN Gateways Fig II 102 Manage IP Phone Templates Add Entry page For snom models the General Settings page contains the following components Dial Plan String indicates a dial plan string used to match dialed digits from the handset to the certain actions e g dialing Dialog Info Call Pickup is used to switch a subscription to the status information of SIP URLs mapped as Destination Extension on the programmable keys Transfer on Onhook is used to switch the call transfer when the handset is placed on hook Call join on Xfer 2 calls when this option is enabled you will connect the newly arrived incoming call to the call on hold by pressing Xfer button When this option is disabled and you press the Xfer button you will have an option to choose the call on hold to transfer the newly arrived incoming call to or to dial a new destination manually Message LED for Dialog State Missed Calls when this option is enabled the phone will indicate missed calls and changing dialog states using the message LED Dialtone during Hold when thi
112. uploaded previously The Download Recurring Attendant Prompt link is used to download the Recurring Attendant Prompt file to the PC and opens the file chooser window where the saving location may be specified The Remove Recurring Attendant Prompt link is used to restore the default Recurring Attendant Prompt e Friendly Phones the Edit Authorized Phones Database link refers to the Authorized Phones Database page where a list of trusted external phones can be created If external SIP or PSTN users are added to the QX IP PBX Authorized Phones database they are free to access the Auto Attendant Services without passing the authentication or to use the Call Back services The VXML Scenario manipulation radio button selection allows you to upload Attendant s custom scenario file and voice messages The selections are e The Upload VXML Scenario File indicates the file name used to upload a new scenario file The uploaded file needs to be in EpygiXML format the coding standard can be found at Epygi Technical Support and is restricted to a 20KB file size Browse opens the file chooser window to browse for a custom scenario file Please Note You may upload an attendant scenario file along with the voice prompt recordings as a single file To do this create an archive file of the tar gz type containing all the necessary files and upload it from the Upload VXML Scenario Voice Messages page The View Download VXML Scenario link appears only when
113. use a piece of music not longer than one minute in order to leave enough space for user defined messages and voice mails e RTP Channel selection is used to define the channel for the broadcast streaming The RTP channels are created by the system administrator Therefore if you are experiencing problems with using the RTP channels as hold music or no RTP channels are available to select on this page turn to your system administrator for clarification s Audio Line In available only for QX50 QX200 selection uses the external radio broadcasting or any other audio resource as the hold music When selecting this option check with your system administrator if there is an external audio resource connected to the QX IP PBX Extensions Directory The Extensions Directory is a useful tool for callers to get direct access to the QX IP PBX extensions by spelling the username with the help of the phone keypad The Extensions Directory can be accessed through QX IP PBX Auto Attendant Services and it has its own manipulation buttons to browse the directory The Extensions Directory Settings page allows you to make a list of names assigned to the extensions on the QX IP PBX If the name spelled by the caller matches the one s listed in the Extensions Directory the corresponding extension user name s will be played to the caller for verifying the input and selecting the user to connect Each extension s user should record their name with the help of the hand
114. will be repeated If this checkbox is not selected the QX IP PBX will provide an IP address immediately when requested Fig II 208 DHCP Advanced Settings page The following functional buttons are available for managing DHCP options Add opens a page Add Entry page where a new DHCP server option can be defined The Add Entry page contains a group of manipulation radio buttons to select between the predefined DHCP server options or to define your own DHCP server option Overvi IP Routing DHCP DNS PPP PPTP SNMP DHCP L DHCP for VLAN DHCP Advanced Settings Add Entry e Predefined this selection allows you to select from the Predefined OPENS Option Name Time oft predefined DHCP server options Custom Options Option Code 101 The Option Name drop down list contains the most AX ption Value Type Address common DHCP server options Option Value 172522230 The Option Value text field requires the value for the selected option The type and format of the value inserted in this field is dependent on the option selected from the Option Name drop down list Save s Custom this selection allows you to define a new DHCP server options The following parameters are required to be inserted for a new option Fig II 209 DHCP Advanced Settings Add Entry page The Option Code text field is used to insert a code of the option It may have values in a range from 0 to 255 The Option Value Type drop down list is used to select th
115. will display the diagnostics results and the connectivity conditions The system should be reconfigured if problems occur during the diagnostics Security Diagnostics The Security Diagnostics page allows running the security audit and getting the security reports The Start Security Audit functional button is used for running the security audit The QX IP PBX Security Audit is a security reporting system which generates the warnings regarding the QX IP PBX s weaknesses relative to the selected Security Level The warnings may vary depending on the selected global Security Level The Security Audit will detect the security related configuration issues in Firewall IDS IP Line passwords Call Routing and extension settings The output of Security Audit may look as follows Diagnostics System Logs User Rights Amy CallCapture Ping Traceroute Start security audit Security Diagnostics Checking jit Security Audit E Start Security Audit Firewall done IP Lines done Start security audit Checking Call Routing done A fee a Call Routing done Extensions done Extensions done Users done Users aon done Settings do not correspond to selected security level You can view the complete report by clicking the Show the latest security report done Settings do not correspond to selected security level You can view the complete report by clicking the Show the latest Useful Links
116. x 130 QX50 0X200 0X2000 Manual II Administrator s Guide DHCP Advanced Settings The DHCP Advanced Settings page is used to modify the advanced options of the DHCP server on the QX IP PBX This page contains a table where a list of default DHCP server options is already defined re os More options can be added from this page as well as settings of the DHCP Leases DHCP for LAN existing options can be modified All options in the table on this DHCP Advanced Settings page are then sent to the DHCP clients Add Eat Delete e The Authoritative checkbox is used to enable disable Gateways onime 1723041 authoritative mode on the QX IP PBX DHCP server Disabling cscs aan the checkbox is recommended if several DHCP servers are used a aT on the network and the QX IP PBX should provide network servers 1723041 parameters to IP phones only bc epygi config loc Overload tftp server name 172 30 4 1 e The Ping Check checkbox enables checking the availability of Sees an IP address on the network before providing it to a client If E aai this checkbox is selected the QX IP PBX will first ping an IP Ping Check address retrieved from the IP pool and wait for a reply If no a Ping Timeoutseo a reply is received within a timeout specified in the Ping timeout text field by default 1 sec the retrieved IP address will be provided to the client If otherwise a new IP address will be retrieved from the IP pool and the procedure
117. you are sure that the image version is appropriate ea for your device press Yes otherwise press No After pressing No press Discard this image button to start upload a new image Fig II 284 Firmware Check page If you have confirmed the firmware version a new page with firmware update progress will be displayed next There are no functions available on this page just information about the Burning Image firmware update procedure At some point the connection with eae eee the device is being lost and you need to wait until the firmware o Performing software update now This process cannot be interrupted will be burned on the QX IP PBX wane A The device will reboot itself when the software update is ready You will not be automatically redirected to the Login page To WARNING If your IP phones are not in the Supported Phones list you will have to reboot them manually after firmware upgrade is finished access the QX IP PBX s Web GUI you need to connect QX IP PBX i E I ena This is the last message you will see here again and login a Attention After the firmware update all IP phones attached to the QX IP PBX should be restarted Fig IT 285 Firmware Update Burning Image page QX50 QX200 QX2000 SW Version 6 0 x 169 QX50 0X200 0X2000 Manual II Administrator s Guide Get Firmware From Server The Get Firmware From Server page allows you to get a new Firmware image from the FTP server
118. 0 0X200 0X2000 Manual II Administrator s Guide Successful Calls Missed Calls Unsuccessful Outgoing Calls Archive Archiving Settings Call History Settings Y Enable Call Reporting Maximal Number Of Successful Call Records 10000 Maximal Number Of Missed Call Records 10000 Maximal Number Of Unsuccessful Call Records 10000 Y Download All Call Detail Records Download All Call Detail Records in CSV format Clear all Records Fig II 248 Call History Settings page When the number of Call History entries exceeds the numbers specified in the Call History Settings page the oldest entries are being automatically deleted In order to keep the Call History entries safe QX IP PBX allows you to configure the Archiving Settings service of the Call History Call History Archive In the table on this page all available Call History archived files are listed The Filter button performs searching within the Archive table The Archive Record field shows the time when the Call History was archived The csv and log links in this field allows you to download the archived Call History file to the PC in a Comma Separated Values csv or Tab Delimited Text log file formats and opens the file chooser window where the saving location can be specified The Number of Call Records field shows the number of records in particular Call History archive file The External Backup Status shows the status of the backup T
119. 0 sip epysiloc C reated e 7412138 es Auto 138 138 PBX 138 Fig II 89 ACD Management page Agents page The Add Agent page contains the following components ACD Agent ID requires the number of the agent Digits are only accepted for this field The Agent ID should be unique in the system Password requires a password of the agent The agent password may only contain digits If non numeric symbols are entered the Incorrect Password no symbol characters allowed error will prevent creating the agent QX50 QX200 QX2000 SW Version 6 0 x 62 QX50 0X200 0X2000 Manual II Administrator s Guide Confirm Password requires a password confirmation If the input is not corresponding to the one in the Password field the Incorrect Password confirm error will appear Skills Groups ACD Agent Management Add Agent O Go Back Extensions Description requires an optional description of the agent ACD Agent ID 2222 John Smith Call Type lists the available call types e PBX extensions on the QX IP PBX e SIP calls through a SIP server e PSTN calls to a global telephone network e Auto used for undefined call types The destination independent on whether it is a PBX number a SIP address or a PSTN number will be reached through the Call Routing Table Fig II 90 ACD Management Add Agent page The Calling Address text field is used to define the address by which the agent can be contacted
120. 000 Manual II Administrator s Guide DHCP Server The DHCP Settings for the LAN Interface page offers the m following input options 3 DHCP Leases DHCP for VLAN Enable DHCP Server checkbox activates the DHCP server on QX EAE SANSE EE A peter IP PBX With this checkbox enabled QX IP PBX will be able to el assign dynamic IP addresses to the devices in its LAN Seen nen ner eee Give leases only to hosts listed in the static MAC address binding table checkbox enables the DHCP services only for the devices listed in the Special Devices table With this checkbox Sittings selected no DHCP services will be provided to the other devices WINS Server 0 JO 0 0 Special devices Add Edit Delete Hostname MAC Address Static IP Address Host Options Advanced Configuration Aastra67391 00 08 5d 13 bc 15 172 30 4 50 DHCP Advanced Settings Cisco525G d0 d0 fd e9 65 f0 172 30 4 51 DHCP Advanced Settings YealinkT19 00 15 65 54 3e f6 172 30 4 52 DHCP Advanced Settings PolycomVV300 00 04 f2 81 3e ef 172 30 4 53 DHCP Advanced Settings Save Fig ll 206 DHCP Settings page for LAN interface page IP Address Range defines a range of IP addresses that will be assigned to the QX IP PBX LAN users The IP range must be at least 6 otherwise the error message Address Range too small will prevent it from being saved The error message Address Range too large will appear if the IP range exceeds the allowed IP address range defined
121. 2 except for R2 DTMF When this checkbox is selected the pound sign detected in the dialed number will be passed through and will be considered as a part of the dialed number When this checkbox is not selected the detected pound sign will be considered as a Call acceleration digit QX50 QX200 QX2000 SW Version 6 0 x 79 e C Pyg l QX50 0X200 0X2000 Manual II Administrator s Guide CAS Signaling Wizard Page 4 appears only in E1 User mode when signaling selected from Signaling Type drop down list on the CAS Signaling Wizard Page 2 is R2 all types and is used to configure country settings Page consists of the following CAS Signaling Wizard components El T1 Trunk ISDN Trunk PSTN Gateways PR Interfaces Country Settings Country drop down list is used to set the location where QX is located to support the correct functionality of R2 signaling For countries absent in this list use ITU selection Selected Timeslots me Use Default Country Settings checkbox restores default mu advanced settings for the selected country When this checkbox ne is not selected next page will provide a possibility to manually A configure advanced country settings Previous Fig IT 118 CAS Signaling Wizard Page 4 CAS Signaling Wizard Page 5 appears only in E1 User mode when signaling selected from Signaling Type drop down list on the CAS Signaling Wizard Page 2 is R2 all types and when Use Default Country S
122. 3 Change values in Packetization Interval and or enable disable Silence Suppression 4 Tosavethe codec settings press Save or to keep the initial data click Go Back SIP Settings The SIP Settings provide information on the SIP receive UDP and TCP ports and allows you to select DNS server configurations for SIP and the SIP timers scheme The UDP Port indicates the SIP UDP User Datagram Protocol receive port number By default 5060 is selected and used The SIP Voll UDP port cannot be in the selected RTP RTCP port range for FXS a gt and IP lines see RTP Settings otherwise the Mapped port for SIP SIP Settings shouldn t be in RTP port range error message appears vo port 000 Telephony The TCP Port indicates the SIP TCP Transmission Control aera pes Protocol receive port number By default 5060 is selected and a ee used mam Please Note QX IP PBX will not use TCP protocol as a transport for Use defuse DNS defined inthe network sting SIP messages if the TCP Port field is left empty speci The TLS Port indicates the SIP TLS Transport Layer Security receive port number By default TLS port is not used and is empty coded to 0 TLS port number should be different from the TCP SIP timers Port number nec3261 SIP DNS 2 All timers according to the standard High availability The retry periods are shortened The Realm text field requires messaging level information to be included in SIP messages sent by QX IP
123. 50 QXK200 QX2000 Manual II Administrator s Guide Add Multiple Extensions The Add Multiple Extensions tab is used to add multiple extensions to the Extensions Management table at once The page consists of the following components Type checkbox is used to select the type of the extensions User Extension Pickup Group Call Park Paging Group ACD group Recording Box or Attendant to be created Quantity text field requires the number of extensions to be created at once For example inserting 5 in this text field will add 5 new extensions to the Extensions Management table Extensions Management Add Multiple Extensions O Go Back Type User Extension v Start from the Extension text field requires the number of the first new extension to be created Depending on the value in the Quantity A text field the next extensions to be created will have subsequent PEE E aaa numbers For example if you have inserted 41 in this text field and Automatically attach to P Line the Quantity text field contains the value 5 then extensions 41 42 StartFromthelP Line 43 44 and 45 will be added to the Extensions Management table If non digit symbols have been entered the error Incorrect Extension no symbol characters allowed will appear If an extension with the given numbers already exists in the Extensions Management table a next subsequent not used extension number will be used instead Please Note Extension cannot start with the
124. Automatic Dialer Application 32 users Oo DCC Basic Support the number for licensed Basic level DCC Conference Server Full support for Conference Server 500 users extensions Call Recording Support for Call Recording capability 32 users e iQall Toggling Support the number for licensed iQall exte nsions Video Conferencing Conferencing Enabler forthe Conference 64 users erver e Autodialer Support the number of maximum simultaneous Telepresence Support Support for Telepresence Application 16 users Auto Dialer calls e Conference Server the number for maximum conference calls e Video Conferencing the number for maximum video conference calls Fig II 18 Trial Features Activation page QX50 QX200 QX2000 SW Version 6 0 x 19 epygl Redundancy Redundancy feature is used to increase QX IP PBX device availability using second QX IP PBX as a backup unit This requires two units running the same firmware version and connected to each other through Ethernet or LAN ports depending on the device model The idea of redundancy is to ensure uninterrupted functionality of the QX IP PBX The Redundancy Settings should be configured on both QX IP PBXs One of the QX IP PBXs is configured as a master the second one as a backup unit Please Note To setup a redundant network you should first startup the master device with all attached IP phones and other devices make sure it works normally and then startup the backup de
125. BX checkbox is selected QX generates ring tones to incoming callers during E1 T1 call dialing This feature is mainly applicable to 2 stage dialing mode Enable Echo Cancellation checkbox enables the echo cancellation mechanism on the selected timeslot s When Alternative Disconnection Mode checkbox is selected the QX will play a busy tone towards the PBX CO if the call has been failed After 60 second timeout the QX will disconnect the call from PBX CO and will stop playing the busy tone Voice Establishment Procedure manipulation radio buttons group is used to select a method of voice establishment on the trunk e Oncall acceptance with this selection voice will be established after call is being accepted e On channel selection with this selection call will be accepted during channel selection This selection is not allowed for R2 signaling e On call ringing with this selection voice will be established after call is being ringing Selection enables Generate Progress Tone checkbox which is used to enable the progress tone generation upon voice establishment QX50 QX200 QX2000 SW Version 6 0 x 78 CAS Signaling Wizard Page 2 appears if the Signaling Type on the previous page is set to any of the E amp M types or to R2 DTMF The page provides the possibility of enabling the DID Service on the timeslot s and contains the following component The Enable DID Service checkbox is used to enable disable DID Direct Inwa
126. BX will first try to reach the local PSTN allocated emergency destination and if failed will dial the ITSP emergency destination Please Note If the defined ITSP is 911 compliant then you have to bind this account with the geographical address of your device If the ITSP is not 911 compliant then the public safety agency will not be able to determine the address automatically The Failover to PSTN checkbox selection will route the call to the PSTN through the local FXO line in case if the VoIP Carrier is not available When this checkbox is selected an additional entry will be added to the Call Routing Table This maintains digit transmission to the local PSTN when an IP call towards the configured VoIP Carrier cannot be established Please Note A warning message will appear when the defined Access Code already exists in the Call Routing Table or causes a conflict with entries already in the Call Routing table In this case when continuing through the VoIP Carrier Wizard the existing entry in the Call Routing table will automatically be overwritten by the new settings Call Routing Table The Call Routing Table lists manually defined routing patterns along with their parameters pattern number state routing and source caller settings RTP Proxy and Date Time period settings metric and description as well as automatically created and undeletable patterns created from the System Configuration Wizard QX50 QX200 QX2000 SW Version 6 0 x 92
127. C transmits the L2TP packets over the network to the L2TP network server LNS at the corporate side Large carriers also may use L2TP to offer remote POPs to smaller ISPs Users at the remote locations dial into the modem pool of an L2TP access concentrator which forwards the L2TP traffic over the Internet or private network to the L2TP servers at the ISP side which then sends them on to the Internet For PPTP and L2TP Connections two parties are required a Client and a Server The client is responsible for establishing the connection The server is waiting for clients it is not able to initiate the connection itself Attention L2TP tunnels have no data encryption mechanism QX50 QX200 QX2000 SW Version 6 0 x 140 e C Pyg l QX50 QX200 QX2000 Manual Il Administrator s Guide The Host Name and a Password specify each side The client should know the server s name and password the QX IP PBX server has no password and the server should set the client s host name and a password The client and server settings have to match on both sides for successful connection establishment Clients and Servers are identified by their hostnames which means that only one client can be connected to the server in the same network Servers also define the range of IP addresses that are assigned to the Server and Client hosts participating in a connection The PPTP L2TP Configuration link displays a page where a new PPTP and L2TP connection can be
128. CP DNS PPP PPTP Address Range and WINS Server This page contains the following EED GE components DHCP Settings for the VLAN Interface Enable DHCP Server Enable DHCP Server checkbox activates the DHCP server on QX en IP PBX for VLAN With this checkbox enabled QX IP PBX will be a ae es ea E able to assign dynamic IP addresses to the devices in its VLAN 55 10101006 10 10 100 254 10 10 1005 VLAN Settings Activate functional button is used to activate DHCP service on one of the VLAN interfaces in the list Only one VLAN interface can have DHCP service activated Save Edit functional button opens a page where the corresponding VLAN interface can be configured and controlled This page contains all the same components as the DHCP Server page does Fig ll 211 DHCP Settings page for VLAN interface VLAN Settings link moves to the VLAN Configuration page where virtual LAN WAN interfaces may be created DNS Settings The DNS Settings page provides the option of setting up a name server for the QX IP PBX It offers the following components The Nameserver Assignment radio buttons are as follows ee Nameserver assignment e The Dynamically by provider selection automatically configures the assignment of the name server address from E the provider party i jn Dynamically by provider e Fixed Nameserver address is a manually selected name server The Nameserver text field requires the IP address of an external name serve
129. Conference Moderator s Main Page iia 184 CONTETENCE AP o o nn 185 A tre rr TTT Tre rrr err tr rr rrr tre vr Trt rrr rrr rrr Tr rr errr ter cre rr rr Tree vrererr rr er err err ere 186 Conference Settings scisccissscsscccnsccnsesaneaccunessesenscenscenecsneeenseeaacesasansesesensdenseenesenesanseasinaeudedseaaseneeansdenedsnesdaseuaaundaaddaeddsedecensdenedeneedeanassucdsdedaedaseddsensdenesadanseanadaudsavdeeddseusseresd 187 e E on 187 o A ee ee ee ee eee eee eee 188 CUStOMIZA UO PAPA PA e o 189 Participants 1 OR pee Pere PP o ee ea ee a ee a ee Ce 190 ig Paracas Cont aora tn rere rrr rrr rn rrr tit mcr rte cmt Tt 191 Honaser Added Participants Conne nra oran ata 192 PP Po o ee ee ee 192 Send aa A o O o ee ee eee ee ee ee eee eee eee 193 Appendix Sottware License Aprre Eme it aa 194 Manual III see Extension User s Guide Describes detailed the menus available for extension users and includes further all call codes at a glance QX50 QX200 QX2000 SW Version 6 0 x 7 QX50 0X200 0X2000 Manual II Administrator s Guide About this Administrator s Guide The QX IP PBX Manual is divided into three parts e Manual Installation Guide gives step by step instructions to provision the QX IP PBX and configure the phone extensions with the Epygi SIP Server After successfully configuring the QX IP PBX users will be able to make SIP phone calls to remote QX IP PBX devices make local calls to the P
130. D Support enables the ACD Management feature which provides contact center solution for queuing and automatic distribution of the calls between contact center agents Free trial Expiration Date 2014 08 06 23 00 iQall Toggling Support Support for iQall toggling Free trial Expiration Date 2014 08 06 23 00 IP Phone support Support for additional IP Phones Free trial Expiration Date 2014 08 06 23 00 Autodialer Support Support for Automatic Dialer Application Free trial Expiration Date 2014 08 06 23 00 Conference Server Full support for Conference Server e Barge In enables the Barge In Service on the QX IP PBX The feature allows the PBX users to participate to the third party s calls while remaining imperceptible Free trial Expiration Date 2014 08 06 23 00 Call Recording Support for Call Recording capability Free trial Expiration Date 2014 08 06 23 00 Video Conferencing Video Conferencing Enabler for the Conference Server e Redundancy activates the Redundancy feature on the QX2000 Redundancy feature is readily available for the QX50 QX200 by default without a software license key e DCC Pro Support allows run with QX IP PBX the Pro level Desktop Communication Console the application description can be found at Epygi Technical Support Fig II 16 Features page e DCC Basic Support allows run with QX IP PBX the Basic level Desktop Communication Console the application description can be f
131. D User Name PIN Code Expiration Date and Time Description PIN Code lt hidden gt 08 06 2014 17 39 The Add functional button opens the Call Routing Local AAA Table AA iets Add Entry page where a new local AAA record can be created The Call Routing Local AAA Table Add Entry page offers a group of manipulation radio buttons to select the type of authorization and the following other parameters Fig II 146 Call Routing Local AAA Table page e Authentication by Caller ID this selection is used to set the authentication based on the caller s phone number which is A considered to be automatically detected The Phone Number SIP Call Routing Local AAA Table Add Entry User Name text field requires the caller s phone number or the SIP Siana username Only numeric and wildcard characters see chapter E Authentication DC 5 5 Entering SIP Addresses Correctly are allowed for this field o 5 Y are used to define a range or a quantity of numbers For nace SO eet example 2 13 17 ww a c means that the dialed number may be iia 213 214 215 216 or 217 2ww 2a 2b and 2c to match the oe ee specified phone number in the case of 2 3 7 the dialed number may be 23 or 27 to match the specified phone number The 11 15 23 38 45 pattern means that the dialed number may be 11 15 23 38 or 45 to match the pattern 11369 Expiration Date and Time Fig II 147 Call Routing Local AAA Table
132. D will be unconditionally displayed When the Enable Connect Acknowledge Option checkbox is selected QX will stop the T303 and T310 timers upon receiving the CONNECT message will send a CONNECT ACKNOWLEDGE message to the remote side and enter the active state When this checkbox is not selected QX will stop the T303 and T310 timers upon receiving the CONNECT message and will enter the active state without sending the CONNECT ACKNOWLEDGE message to the remote side P Asserted Identity The Disable P Asserted Identity radio button disables the P Asserted Identity feature for both incoming and outgoing calls The Override CLID with P Asserted Identity radio button selection enables the SIP P Asserted Identity support For the calls from SIP to E1 T1 if the Invite SIP message contains a P Asserted Identity or a P Preferred Identity or a Remote Party ID then the CallerID on E1 T1 is sent with the original Caller ID which comes from the identity field SIP user agent should check for the existence of the P Asserted Identity then the P Preferred Identity then the Remote Party ID to fill the identity field For the calls from E1 T1 to SIP with restricted Caller ID the SIP Invite message contains P Asserted Identity field with the value from the Caller ID on E1 T1 The SIP From field contains anonymous The Use Redirecting Number Info Element with P Asserted Identity radio button selection enables full support of the SIP P Asserted Identity For the
133. Encryption drop down list where the encryption method can be selected L2TP Server Configuration The L2TP Server Configuration page is used to configure the L2TP server settings and provides the following input options The L2TP Subnet text fields are used to enter the IP address range for the L2TP server and clients within the L2TP tunnel The value specified for the subnet mask is fixed to 24 to restrict the possible number of clients for the L2TP connection Please Note The first address specified in the L2TP Subnet will be assigned to the L2TP server others will be assigned to the clients The L2TP server subnet should be different from the PPTP server subnet otherwise a corresponding error message will appear QX50 QX200 QX2000 SW Version 6 0 x Network PPP PPTP sec PPTP L2TP Configuration Connec tions PPTP Server Configurations L2TP Server Configurations Subnet 10 110 120 0 24 Authentication Encryption CHAP MSCHAP MSCHAPv2 PE 128 Bit Y Fig II 233 PPTP Server Configuration page DNS PPP PPTP Psec PPTP L2TP Configuration erver Configurations L2TP Server Confiqurations Fig II 234 L2TP Server Configuration page 142 QX50 0X200 0X2000 Manual II Administrator s Guide To Specify an IPSec Connection 1 Press the Add button on the IPSec Connection Settings page The IPSec Connection Wizard will appear in the browser window 2 Selecta VPN Peer Type and assign a na
134. For Recording Box extensions the Extensions Management Edit Entry page consists of General Settings SIP Settings SIP Advanced Settings and Recording Box Settings pages The SIP Settings and SIP Advanced Settings pages are the same as for the regular extensions described above The General Settings and Recording Box Settings pages are described below QX50 QX200 QX2000 SW Version 6 0 x 40 e C Pyg l QX50 QXK200 QX2000 Manual Il Administrator s Guide 1 General Settings for Recording Box extension This group requires Recording Box extension s information and has ion Ada Multiple Extensions Bulk Import the following components Extensions Management Edit Entry Extensions O Go Back Display Name is an optional parameter used to recognize the General Settings Recording Box extension Usually the display name appears on the General Settings 400 called party s phone display when a call is made or a voice mail is gt sent NA SS Generate Password Password requires a password for the Recording Box extension D GUI Login lowes Show on Public Directory Percentage of Total Memory 5Y Save Go To Recording Box Go To Codec Settings Fig II 57 Extensions Management Edit Entry General Settings page for Recording Box extension The extension password may only contain digits If non numeric symbols are entered the Incorrect Password no symbol characters allowed error will prevent ma
135. Guide The Password text field requires the authentication password Please Note The User Name and Password should match both 2 A A O A een on master and slave QX IP PBXs for the successful SIP tunnel SIP Tunnel Settings Tunnels to Slave Devices Add Entry establishment Oco bac Telephony SIP_Tunnel_7415 The Symmetric NAT checkbox should be selected when the slave joo QX IP PBX is located behind the symmetrical NAT Y Symmetric NAT Save Fig II 151 SIP Tunnel Settings Tunnels to Slave Devices Add Entry page The Enable Tunnels to Master Devices checkbox enables the QX IP PBX as a slave device and allows connecting to the master QX IP PBX via SIP tunnel When this checkbox is enabled the Tunnels to Master Devices table needs to be configured Call Routing Table Call Routing Local AAA Table Global Speed Dial SIP Tunnel Class of Service SIP Tunnel Settings Tunnels to Master Devices Add Edit Delete Telephony SIP Tunnel Name T r i Registration State Registration Date Time The link Tunnels to Master Devices moves you to the page where a list of master devices needs to be defined SIP_Tunnel_7415 k 7 192 168 74 12 6 Registered 08 13 2014 10 45 07 Fig Il 152 SIP Tunnel Settings Tunnels to Master Devices page The Tunnels to Master Devices page consists of a table where master devices are listed with the corresponding authentication parameters Add functional button leads to the Add Entry page wh
136. IP calls through them and accessing peers located behind the slave QX IP PBX or recognized by it This enables the master QX IP PBXo to locate the slave even when the network settings like IP address SIP port and other settings are changed on the slave QX IP PBX When the SIP Tunneling service is enabled virtual tunnels between the master and its slaves are created A possibility to use the created SIP tunnels will be automatically enabled in the Call Routing Table Optionally a SIP tunnel can be mutually established on two QX IP PBXs allowing to route SIP calls back and forth A QX IP PBX can be at the same time configured both as a slave and as a master to the same remote device i e the slave QX IP PBX can act as a master for the master device it is registered on For example the QX IP PBX 1 can act as a slave for the QX IP PBX 2 In its turn the QX IP PBX 2 can act as a slave for the QX IP PBX 1 With this configuration and the corresponding routing rules added in the Call Routing Table on both devices the SIP calls will be routed from QX IP PBX 1 to QX IP PBX 2 and vice versa The SIP Tunnel Settings page is used to enable the QX IP PBX asa slave or master device for SIP tunneling The page consists of the following components SIP Tunnel Settings Y Enable Tunnels to Slave Devices The Enable Tunnels to Slave Devices checkbox enables the QX IP PBX as a master device and allows you to configure the SIP tunnels to the slave QX IP PBXs W
137. ISON 1 WAN 2 ISON 2 j AUDIO gt epygi Bee E OO l_a M Manual II Administrator s Guide Edition 1 November 2014 SW Release 6 0 7 and higher e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide Notice to Users This document in whole or in part may not be reproduced translated or reduced to any machine readable form without prior written approval Epygi provides no warranty with regard to this document or other information contained herein and hereby expressly disclaims any implied warranties of merchantability or fitness for any particular purpose in regard to this document or such information In no event shall Epygi be liable for any incidental consequential or special damages whether based on tort contract or otherwise arising out of or in connection with this document or other information contained herein or the use thereof Copyright and Trademarks Copyright O 2003 2014 Epygi Technologies LTD All Rights Reserved Quadro and QX are registered trademarks of Epygi Technologies LTD Microsoft Windows and the Windows logo are registered trademarks of Microsoft Corporation All other trademarks and brand names are the property of their respective proprietors Emergency 911 Calls YOU EXPRESSLY ACKNOWLEDGE THAT EMERGENCY 911 CALLS MAY NOT FUNCTION WHEN USING QUADRO OR QX AND THAT EPYGI TECHNOLOGIES LTD OR ANY AFFILIATES AGENTS SUBSIDIARIES PARTNERS OR EMPLOYEES ARE NOT LIABLE FOR SUCH CALLS Limite
138. If the agent doesn t answer within Ringing Timeout the system tries to reach the next agent with the highest composite skill etc If the call is not answered within Common Timeout the system either disconnects or redirects the call Enable Redirect checkbox is used to enable the call redirection to the other destination after some time spent in the queue This will avoid the caller to wait in the queue for too long This checkbox selection enables the following components Call Type lists the available call types e PBX local calls to QX IP PBX s extensions e SIP calls through a SIP server e PSTN calls to a global telephone network e Auto used for undefined call types The destination independent on whether it is a PBX number a SIP address or a PSTN number will be reached through the Call Routing Table The Redirect Address text field is used to define the address where the call will be redirected It might be within the scope of ACD like the address of another ACD agent or out of scope like the address of some voice mailbox The value in this field is strictly dependent on the Call Type defined in the same named drop down list If the PBX call type is selected the QX IP PBX extension number should be defined in this field For the SIP call type the SIP address see chapter Entering SIP Addresses Correctly should be defined for the PSTN call type the PSTN user number should be defined here For the Auto call type a routing patte
139. Licensed Features GUI page the video codecs will be available on the QX IP PBX s Conference Codecs GUI page Please Note Administrator should enable only one codec at a time either H 263 or H 264 Video Conferencing provides possibility to view particular participant based on switching modes In general there are two switching modes for each phone e Manual allows participant to switch between video capable participants manually by dialing 900 or 900 a participant will see the next or previous participant who has video capability enabled In the context of manual switching next and previous means the order of entrance to the conference bridge so the first caller will be the first video capable participant connected to conference QX50 QX200 QX2000 SW Version 6 0 x 53 epygl e Automatic In this mode QX IP PBX determines the speaker or loudest participant and will automatically switch the video stream to show that speaker As a result all the video phones which are in automatic mode will see the speaking participant If participant does not have a video phone then the other participants will see a black screen Please Note Users can switch between manual and automatic mode by using 6600 0 90 and 900 By default Automatic Speaker Detection is switched off From the Conference Settings page administrator can enable or disable the default mode for video conferencing see Automatic Speaker Detection Conferen
140. Line Status for any shared ISDN Trunks on the QX IP PBX displays the state of the B1 and B2 channels and the information about the active calls on them This page includes a group of static and dynamic parameters Static parameters are always displayed Dynamic parameters appear only when an event takes place on the channel Static Parameters e B channel the state of the channel enabled or disabled e State the current state of the channel free busy or N A Dynamic Parameters e Caller Party this parameter appears when a call is received and indicates the caller address e Called Party this parameter appears when a call is placed and indicates the destination address e Call Duration current call duration in seconds The Line Status for shared E1 T1 Trunk displays the list of available timeslots in E1 mode 30 active timeslots both for CAS and CCS signaling types in T1 mode 24 timeslots for CAS signaling and 23 timeslots for CCS signaling type and their settings Route Incoming Call to Allowed Call Type and Timeslot State When Timeslot is in the call information about call direction incoming or outgoing Caller Party Called Party and Call Duration is displayed QX50 QX200 QX2000 SW Version 6 0 x 146 QX50 0X200 0X2000 Manual II Administrator s Guide epygl Memory Status The Memory Status page includes tables with the available User Space information for each extension These tables display the space used by
141. Min Data Rate text field requires the amount of upstream bandwidth that ought to remain for data applications even if voice applications use the entire available upstream bandwidth The value selected here needs to be smaller than the upstream bandwidth and is measured in kbit s QX50 QX200 QX2000 SW Version 6 0 x 13 QX50 0X200 0X2000 Manual II Administrator s Guide The WAN IP Configuration page is only displayed if Ethernet or PPTP has been selected to be the uplink protocol It offers the following components The Assign automatically via DHCP radio button selection switches to automatic retrieval of the WAN IP address from a DHCP server at the ISP uplink Please Note DHCP referred to here is the one that runs on the provider s side and not the QX IP PBX s personal DHCP server The Assign Manually radio button switches to the manual adjustment of IP settings This selection requests the following parameters IP Address requires the IP address for the QX IP PBX WAN interface Subnet Mask requires the subnet mask for the QX IP PBX device WAN interface Default Gateway requires the IP address of the router where all packets are to be sent to for example to the router of the provider The WAN Interface Configuration page may be used to modify the MAC address of the QX IP PBX This might be necessary if the ISP Internet Service Provider requires a specified MAC address for example for authentication This page o
142. Model Other SIP UserName locext107 Model Other SIP UserName locext108 Model Other SIP UserName Eyebeam109 Model Other SIP UserName locext110 Model Other IP Line 9 SIP UserName locext111 Model Epygi QXFXS24 00 f0 00 f0 81 85 Template systemdefault IP Line 10 SIP UserName locext112 Model Epygi QXFXS24 00 f0 00 f0 81 85 Template systemdefault Fig l IP Lines FXS FXO E1 T1 Trunk ISDN Trunk PSTN Gateways IP Line Settings IP Phone Templates IP Phones Logo FXS Gateways IP Line Settings IP Line 1 O Go Back Inactive SIP Phone Phone Model Grandstream GXP1400 MAC Address po 0b 62 cd cbf bb Password Generate Password Transport Use Session Timer lt use default gt Y Use template Y Enable Hot Desking Capability Hot Desking Automatic Logout Never After 0 hour s 0 min At 16 Y 12 Y AdvancedWebReboot WebReboot WebReboot AdvancedWebRestart AdvancedWebRestart 97 IP Lines page Fig Il 98 IP Line Settings Edit page For automatic SIP phone configuration the SIP phone should be reset rebooted The appropriate configuration will then be automatically downloaded from QX IP PBX to the SIP Phone Please Note For automatic configuration some SIP phones may require additional actions to follow the restart For example by default the IP Dialog SIP Tone II is in a non auto provisioning mode so it should be manually enabled on the phon
143. N The NAT service should be enabled on the QX IP PBX to provide the possibility of Port Forwarding in the Incoming Forwarding filtering rules The Port Forwarding function will be unavailable if NAT is disabled on the QX IP PBX Outgoing Traffic The Outgoing Traffic filter is for outgoing traffic The rules here allow or deny QX IP PBX s LAN users to reach external services Management Access Management Access is used to enable management access to the QX IP PBX from the Internet A host on the Internet can be allowed to reach the QX IP PBX Call Control Access Call Control Access is used to enable the access from the call controlling application from the Internet to the QX IP PBX The call controlling applications can be used to remotely initiate and handle calls on the QX IP PBX and to subscribe for certain event notifications from the QX IP PBX QX50 QX200 QX2000 SW Version 6 0 x Filtering Rules Custom Outgoing Management Access Call Control Access SIP Access Blocked IPs Allowed IPs Filtering Rules Allow or deny access from the Internet to services in your LAN If you don t host any services for example a webserver or gameserver you don t need to configure anything here Y Firewall Enable Disable Add Edit Delete State Service Action Restricted IP Forward to IP Description Enabled User QuadroFXS26GW Allowed Group MyPC 172 30 4 253 80 Fig II 183 Filtering Rules page Firewall Filtering Ru
144. N number e Duration the voice mail duration e Date the date the voice mail was received Fig II 169 Voice Mail Recording Codec page To insert the predefined tag to the subject line you should simply click on the corresponding tag The following format should be maintained to create a flexible subject Example Voice mail received from VM_DISPNAME VM_DATE In this example all email subjects will contain a static text Voice mail received from following by the display name of the caller and the date voice mail is received FAX to E mail format drop down list is used to define the format of the FAX document received in the voice mail and to be attached to the email in case user has enabled Send new voice messages via e mail option from his personal Voice Mail Settings TIFF or PDF formats may be selected here RTP Streaming Channels The RTP Streaming Channels page is used to configure channels where the broadcast RTP streams are transmitted These channels may be then configured to be used as hold music see Manual III Extension User s Guide or any other type of music played to the caller The RTP Streaming Channels page consists of a table where RTP channels are listed Add opens the Add Entry page where a new RTP channel can be added The Add Entry page includes the following text fields ny acorn Gineees Add Edit Delete The RTP Channel Name text field requires the name or the SN Local RTP Port number
145. N IP address If no ps 1921687425 NAT Station is specified here QX IP PBX s IP address will be sent to eS the RADIUS server Secret Key is used to insert the secret key between the Radius client and the server Contact the Radius server administrator to get the secret key for your QX IP PBX The Confirm Secret Key field is used to verify the secret key If the entered Secret Key does not correspond to the one in the Confirm Secret Key field the error message The Secret Key does not match Please try again will appear Retry Count allows you to select the number of attempts authorized before canceling the registration ination RADIUS Server Receive Timeout allows you to select the timeout in seconds between two attempts to register Encoding Type allows you to select the encoding type PAP or CHAP that should be unique on both the client and the server sides for the establishment of a successful connection Encoding type should also be requested from the Radius Server administrator APN The Authorization Port text field requires the port number on the oo RADIUS server where QX IP PBX is to send the authentication requests The Accounting Port text field requires the port number on the RADIUS server where QX IP PBX is to send the accounting messages Fig II 174 Radius Client Settings page 2 Authentication Settings The Enable common login for all users in time of by Phone authentication checkbox enables custom set
146. N Parameters page The Keep alive interval text field provides the options to select the time interval in seconds for keeping NAT mapping alive The value should be in the range of 10 to 300 seconds The NAT IP checking interval text field indicates the interval in seconds between the NAT IP checking attempts used to distinguish the possible NAT IP address changes and to perform registration on the new host The value should be in the range of 10 to 3600 NAT Exclusion The NAT Exclusion Table lists all possible IP ranges that are not included in the NAT process but may be accessed directly IP addresses that are not listed in the NAT Exclusion Table are accessed over NAT For example if a QX IP PBX user needs to make SIP calls within the local network as well as outside of that network all local IP addresses are required to be excluded from NAT traversal settings by being listed in this table Otherwise a malfunction may occur in SIP operations The NAT Exclusion Table page offers the following input options Each record in the table has a corresponding checkbox assigned to its row The checkbox is used to delete or to edit the corresponding record Only one record may be edited at a time An error message will appear if no selection is made or more than one is selected Each column heading in the table is a link By clicking on the column heading the table will be sorted by the selected column When sorting ascending or descendin
147. NMEA anmNeampAnan 159 MAMAS MET aa 160 DIAS NO SUES AP A AP enn 161 aa nn o O o A E ea ieee ae mire naiametnce nee nceas 161 CAICARA rana 161 mn Penn A e 162 Traceroute in A AAA AA 163 SY SLC LOGS o 2 AAA 163 SP A E E ee 163 is UU oo e 164 Logs ATCHIV ia 164 User Rights ee 165 A o A o E O A 165 0 e AS E E ee o AP nn o An 166 Bach RSCG G sicansancasanerisarsaeaneannsancassessaansaneaneaansasatancessacersoanaanenaaateneaneannanaanacapeananaieainanaaaaanaaesntanraldaanmasniaataas aitansaanateanesdneabtdaseaastianieasaaasarannandtancaisieaeantaananaanaanenaaasanapae 166 A tomatic BACKUP peeeneeererteecenertre state RFE rer ape E A ere tr reine errr ee E E ee eee ee 167 Download LSslble Contgura do ssena aaaea AANA eA eA 167 Upload Ke able Contrato amaidimEa nahin M RANI E E 168 Firmware Update ina AAA AAA AAA AAA AAA 168 Mircea 169 Gee Firmware Pron erecto 170 Aae Pithiwale Udaondo doi 171 REDO OL eE PAEA EEA E sei E E E T A T EE 171 Registration Fori o o o o PP lt A 172 Appendix PBX Services for QX IP PBX s AMM 173 Appendix Conference Services for Moderators and Participants ammm 174 Appendix System DS ic 176 Administrator E 176 Extension Settings rrisin ESE SEEE aa anew a EEEE o e 182 Append Moder tor s MENUS jini nanan di idiota 184
148. NMP The Enable Dynamic DNS checkbox selection enables the dynamic DNS service Dynamic DNS Settings The Dynamic DNS service is provided by third party companies You need to register at their web sites to get the information to fill in below The User text field requires the username specified during the registration at the DynDNS provider ore info as well as links to a list of Dynamic DNS service providers to use The Password text field requires the password specified during the registration at the DynDNS provider The Max time between updates text field requires entering the period between two updates in hours The values entered in these fields should be greater than 24 otherwise the error message TZO Connect tion Type Update interval times smaller than 24 hours are too small will DHS Cloak Tite appear Normally whenever you set up a connection to the Internet aie the DynDNS is updated at least once in the period indicated in this fi el d i Create custom HTTP GET request ii Basic Authentication The Use predefined service radio button leads to the manual configuration of the DynDNS service The selection enables the following optional settings The Service drop down list contains the provider list where the administrator needs to select the one that it has been subscribed to Fig ll 215 Dynamic DNS Settings page The Host text field requires the name of the host on the Internet The TZO Connectio
149. NTABILITY FITNESS FOR A PARTICULAR PURPOSE LACK OF VIRUSES AND ACCURACY OR COMPLETENESS OF RESPONSES CORRESPONDENCE TO DESCRIPTION OR NON INFRINGEMENT THE ENTIRE RISK ARISING OUT OF THE USE OR PERFORMANCE OF THE LICENSED MATERIALS REMAINS WITH YOU 8 LIMITATION OF LIABILITY AND REMEDIES IN NO EVENT SHALL THE LICENSOR OR ANY OTHER PARTY WHO HAS BEEN INVOLVED IN THE CREATION PRODUCTION OR DELIVERY OF THE LICENSED MATERIALS BE LIABLE FOR ANY CONSEQUENTIAL INCIDENTAL DIRECT INDIRECT SPECIAL PUNITIVE OR OTHER DAMAGES INCLUDING WITHOUT LIMITATION LOSS OF DATA LOSS OF BUSINESS PROFITS BUSINESS INTERRUPTION LOSS OF BUSINESS INFORMATION OR OTHER PECUNIARY LOSS ARISING OUT OF THE USE OF OR INABILITY TO USE THE LICENSED MATERIALS EVEN IF THE LICENSOR OR SUCH OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES YOU AGREE THAT YOUR EXCLUSIVE REMEDIES AND THE LICENSOR S OR SUCH OTHER PARTY S ENTIRE LIABILITY WITH RESPECT TO THE LICENSED MATERIALS SHALL BE AS SET FORTH HEREIN AND IN NO EVENT SHALL THE LICENSOR S OR SUCH OTHER PARTY S LIABILITY FOR ANY DAMAGES OR LOSS TO YOU EXCEED THE LICENSE FEE PAID FOR THE LICENSE MATERIALS The foregoing limitation exclusion and disclaimers apply to the maximum extent permitted by applicable law 9 Compliance With Laws You may not use the Licensed Materials for any illegal purpose or in any manner that violates applicable domestic or foreign law You are responsible for compliance with all domest
150. Ne ee ee o so o o no re ee ee 18 Het alo omen tr tr aT tran A tetra art tr TT Cen Tt Ten Te cn eT ere TT er Te rr aaa 19 A E E EE E T E E A E E E 20 AN BU ARC Palk crscccnnnoni a oni a N N R N 20 Update Lan tages toc Promesa EAER aaa 21 EXTENSIONS MEL E E E il 22 Extensions Manag ere AAA AA An 23 add JSC SIGH acariciar a a a a a 24 USer Extension Setn CA A ae tet gece en vated lh neh A OO nen 25 Fick Group Ex CuSO Seti aro 32 Gal Par Ee Slt serine Srs tii rapto 34 Paging Group Extension Settings ssim iniii iaie a iaia a aaa iaa aai ada aaar aaa a aiaia ree meee seer re a ee er 36 ACD Group Extension Settings a ere err 37 Recording Box Extension gC LANES sarren aaa aaae aAA Aaaa aaa aaa aaa araara araa araara aara araara arra TOT 40 Pecore BOX raae A A A E aaa yaaa aaa eens 42 Pea Ee e a E EO ae eter ire Mem tire mre ree errr 43 EXCeNSION COUCCS E A AN N N ka i oR ACE prpaa praia 49 Call Par anid Directed Cal Park Ser VICE usada 51 Barge IM Service OE A o 51 Ada Multiple Extension 52 User Extension Bulk IMI DOLE ansia arado aaa AARAA ERa aaa a 52 Conferences masa AAA AAA AAA 53 Conterences Manantial 54 AO COME rencor iaa panic 55 A AP O E O E o aieaen aaa nTanaNn 55 Universal Extension Recordings ee cieiicec s 55 Universal Extension Recordings Hold MUSIC msn 56 Extensions Directory ci 56 Receptionist Manage AAA 2 2 AP ne 57 Ig Pq
151. OFTWARE PROGRAM INDICATES YOUR ACCEPTANCE OF THESE TERMS AND CONDITIONS IF YOU DO NOT AGREE TO THESE TERMS AND CONDITIONS YOU MAY NOT USE THE HARDWARE OR SOFTWARE 1 License Epygi Technologies LTD the Licensor hereby grants to you a non exclusive right to use the Quadro or QX Operational Software program the documentation for the software and such revisions for the software and documentation as the Licensor may make available to you from time to time collectively the Licensed Materials You may use the Licensed Materials only in connection with your operation of your Quadro or QX You may not use copy modify or transfer the Licensed Materials in whole or in part except as expressly provided for by this Agreement 2 Ownership By paying the purchase price for the Licensed Materials you are entitled to use the Licensed Materials according to the terms of this Agreement The Licensor however retains sole and exclusive title to and ownership of the Licensed Materials regardless of the form or media in or on which the original Licensed Materials and other copies may exist You acknowledge that the Licensed Materials are not your property and understand that any and all use and or the transfer of the Licensed Materials is subject to the terms of this Agreement 3 Term This license is effective until terminated This license will terminate if you fail to comply with any terms or conditions of this Agreement or you transfer posses
152. P PBX and apply it manually on the remote side The Download Current CA Root Certificate link is used to download the actual CA root certificate in a crt format Advanced Settings Telephony a Telephony sip TLS Certificates Host Aliases for SIP o Info The host name configured in Dynamic DNS Settings is automatically added to this list Add Edit Delete QX1234 SIP SIP Aliases TLS Certifica Generate and Install New CA Root Certificate Download Current CA Root Certificate Country Name 2 letter code State or Province Name full name Locality Name eg City Organization eg company Organizational Unit eg section Common Name eg Root CA Email address Validation period days CA Key Password Confirm Password ATTENTION After pressing Generate Certificate and Install new certificate will be installed and the system will reboot Generate Certificate and Install Fig II 167 Host aliases for SIP page Fig II 168 Generate and Install New CA Root Certificate page The Advanced Settings page allows you to configure the following settings Voice Mail Common Settings RTP Streaming Channels Gain Control 3PCC Settings RADIUS Client Settings Dial Timeout and Call Quality Notification Voice Mail Common Settings The Voice Mail Recording Codec page is used to configure the codec for the Voice Mail recording and other settings related to the voicemail to email and FAX to email s
153. P Server Configuration Subnet 172 31 2 0 24 Display notification for all events except Login and Firmware Update events Those events have a Do nothing action assigned Additionally Fan Control critical and major failures have a Flash LED action assigned Enable Call Reporting enabled 100 entries for all type of calls Percentage of Total Memory used for Archive 0 Enable Call Detail Records Archive Collection disabled Call Detail Records Archive Structure Archive by records count Call Records Count 50 Time Interval 10min Send archive files to external server Send and delete from archive File Format Tab Delimited Text log Enable Call Reporting enabled 100 entries for all type of calls User Logging enabled Developer Logging enabled Log Lines to Show 25 Comment undefined Disabled Users admin enabled localadmin disabled Roles Extension all accessible pages for extension except for Extension Voice Mail Profiles Local Administrators all accessible pages for localadmin 181 Parameter Automatic Backup Automatic Firmware Update Extension Settings Parameter Voice Mail Settings Voice Mail Profiles Group List Speed Calling Account Settings Basic Services General QX50 QX200 QX2000 SW Version 6 0 x QX50 0X200 0X2000 Manual II Administrator s Guide System Default Value GUI Access Password Old Password empty New Password empty Confirm Ne
154. P Static Route O Go Back Route to 192 Via IP Address 192 168 Save Fig II 201 Add IP Static Route page IP Policy Routes allow IP packets forwarding to the specified router depending on the source IP address as well as defining the priority for the current routing rule The IP Policy Routes table displays all specified IP policy routes with their parameters Target State for the state of the route enabled or disabled Actual State for the state of the route connection up down or erroneous Priority for the route priority Route From is where the subnet routed packets come from and Via IP Address is where the router IP address incoming packets should be routed through Add opens the Add IP Policy Route page to establish a new policy route Enable and Disable are used to activate or to deactivate the selected route s Ove IP Routing DHCP DNS PPP PPTP SNMP IP Static Routes PPTP L2TP Routes IP Policy Routes Enable Disable Raise Priority Lower Priority Add Edit Delete Route from Via IP Address Target State Actual State Priority Fig II 202 IP Policy Routes table Raise Priority and Lower Priority are used to increase or decrease the priority of the selected policy route s by one At least one route should be selected to use these functions otherwise the error message No record s selected will appear QX50 QX200 QX2000 SW Version 6 0 x 128 The Add IP Policy Route p
155. PBX callers when entering Recording Box Recording Box Settings 400 Ask Password on Remote Access checkbox selection enables the Recording Box Settings Y PRA Recording Storage Settings w Play welcome message password protection for remote SIP or PSTN callers when entering Maximum recording count 300 v Recording Box max imum recording duration 1 hour Y Recording Announcement Play Welcome Message checkbox is used to enable disable the Go To Recording Box Pay announcement when starting recording welcome message played when entering the Recording Box Go To Codec Settings Upload new recording announcemen t message Choose File No file chosen Maximum recording count drop down list indicates the maximum number of call recordings allowed to be stored in the corresponding extensions Recording Box If the limit is reached some call recordings should be deleted from the Recording Box to be able to make more recordings Fig IT 58 Extensions Management Edit Entry Recording Box Settings page Maximum Recording Duration drop down list is used to select the maximum duration of the single call recording for the selected Recording Box extension When the call reaches the selected duration the recording will be automatically stopped while the call will stay active Recording Announcement group allows updating the active recording announcement played in the call when call recording starts downloading it to the PC or restori
156. PS APRA poo o PP PP o a PP 122 e A o o o sn 123 Service Pool Configuration iaa dido io poo 123 CAI APP Om A o E o EE 124 IP ao O ini UM Aaa CA o o PP oo 124 SIP TDS S UU GS sanccscsicsnsiccncascancansanasicacsnecncancancansancnsancencadiataanasnsaaaseancannansancsnanaddntacancaieansancsnancenaascaacidealeunesaasceacancancsnnnasnaasdusasicansanesaannenaasanaeanaaiduaandanesseaneancaneansnansasidabeden 126 PSE VES Menard idad da adam eaten E A AA ad 127 PP Routine A cs 128 Static ROUTES lisa 128 IP Policy nO bo ch Be erecta PE AP Eo oo enn 128 ol gy ld eh 01 eee E ree ene ee ee ere ee ee E E 129 A o o o gq 129 DNA EE OO O O e ener 130 DHCP AUY nce SCs oa aia 131 DHCP LEASES A A o e ra o eo eE Oe 131 DHCP Settings tor the VLAN tersa ia 132 DNS SONES isisa EEE EAA ERA A An 132 cn o oO OO APP ee rr ee ee eae ee ae 132 Dynamic DNSSetUNgS AAPP nn 133 PPP PPTP Settini ca 134 A RPE Sc lies A ao 134 SNMP SUMING ii 135 l pa SNMP Se tan Simancas ademas aman aaa 135 SNMP Trap settings a ais 136 VLAN COMU 136 VEN Conta ra Oos S E ciedledaNaieasAkNaRtiseaTeaNnANaNRNDRENTLRNNS 137 PP ee O EEC O er 137 Pele C IPF Cone dou caricia 140 SLATS MENU AAA ction usu AAA AA A ededinudedvededdutasnisuncsinnsnandeadienks 144 SY SECT SCA a DE OI AAA O Eo PP o andanan 145 General oO aiii 145 pa o ooo Po eee eee ieee 145 Ai SLUNG e ncen tenses cae U O A eE E T 145 Memory eL E POCA MERA a
157. Phone database The caller will then be disconnected from the QX IP PBX s Auto Attendant and the defined Call Back destination will receive a call from the QX IP PBX within the next 45 seconds Answering the incoming call the caller will be reconnected to the QX IP PBX s Auto Attendant Please Note For both Permanent Call Back and Non Permanent Call Back the detected caller number must correspond to the one configured for trusted caller In case of PSTN call back at least one PSTN line must be available on the QX IP PBX There must be network connectivity and the destination must be reachable QX50 QX200 QX2000 SW Version 6 0 x 66 pyg Interfaces Menu The Interfaces menu allows you to configure the following settings e IP Lines IP Line Settings IP Phone Templates IP Phones Logo FXS Gateways e FXS Lines FXS On board Line Settings Diagnostic Loopback e FXO Settings e E1 T1 Trunk Settings e ISDN Trunk Settings External PSTN Gateways Authorization Parameters QX50 QX200 QX2000 SW Version 6 0 x EE Interfaces Overview Overview IP Lines P Lines IP Line Settings IP Phone Templates FXS Gateways EXS On board Diagnostic Loopback E1 T1 Trunk E1 T1 Trunk ISDN Trunk ISDN Trunk PSTN Gateways PSTN Gateways Authorization Parameters QX50 0X200 0X2000 Manual II Administrator s Guide E1 T1 Trunk ISDN Trunk PSTN Gateways Configure IP phones for each extens
158. QX IP PBX etc Arrived call is being added to the end of the AG queue if there are no available online agents to answer the call immediately For connecting to the agents always the call at the top of the queue is being selected The call queue settings are configured from the ACD Group Settings see ACD Group Extension Settings Each agent can have of the following states online offline away busy or DND Do not Disturb for details see ACD Agents Table accessible from ACD Group Extension Settings If the same agent is logged into different agent groups he she may have different states in different groups except for DND status If the agent has DND state in some group then his state will be the same for all other groups The state of the agent can be updated either by administrator from the ACD Agents Table with the exception of DND and busy states or by agent from the handset except for busy state The agent for changing the state to online offline away from the handset needs to call the predefined Auto Attendant see Attendant Extension Settings and on attendant s prompt enter the agent ID password and the status code The state changes from online to busy or vice versa automatically when the agent starts or finishes conversation Calculation of Composite Skill Grade Usually before the call arrives to the agent group it is first answered by ACD specific IVR The main function
159. Recording or Recording or Plavback or Playback Recording or Recording or Playback Playback P Outgoing Playback Playback Find Greeting 1ng Blocking Name Out of Office Me Follow Blocking Message Message Message Message Me Welcome Message Message QX50 QX200 QX2000 SW Version 6 0 x 900 Administrator s Logout 173 epygl Appendix Conference Services for Moderators and Participants QX50 0X200 0X2000 Manual II Administrator s Guide This chapter describes the feature codes for the Conference Services that enable the moderator and participants to manage call conferences from the phone Conference Services accessible during the conference Invite Participant To invite a participant dial 1 Participant s SIP address or 1 Routing Number Service is available for Moderators only Get the number of participants in the conference Plays information about the total number of participants in the conference at the certain moment Get the state of recording Plays the state of conference recording started stopped or paused Lock the conference Locks the conference When conference is locked nobody can dial in any more Service is available for Moderators only Unlock the conference Unlocks the conference Now participants are allowed to dial in to the conference Service is available for Moderators only Dial out to all users with dial out settings enabled Initiates the dial out to all participants c
160. SMTP Host requires the IP address or host name of the Simple Mail Transfer Protocol SMTP server This SMTP server is part of your mail server that you normally use to receive and send mails SMTP Port requires the SMTP host port number Basic Setup Mail Sender Address text field requires the source address System LAN Internet WAN DateandTime Short Tet Messaging SMS for the QX IP PBX notification emails The email address setup defined here should be an existing valid email address registered on the selected SMTP server or it should have permission to use that particular SMTP server for e mail SMTP Host 1926802 transmission SMTP Port 5 Mail Sender Address QX200 12 epygi loc System Mail Settings Y Enable Mail Recipient Address text field requires an active email address where system emails will be delivered The e mail recipient here can be a QX IP PBX administrator or someone responsible for network and system problems Mail Recipient Address levon_Dadayan epygiarm am Mail Recipient Address CC astghik_sanasaryan epygiarm am Y The server requires a secure connection TLS Y Enable SMTP Authentication Mail Recipient Address CC text field requires an active terest fas email address where a carbon copy CC of the system e mails a ed e will be delivered Send test mail The server requires a secure connection TLS must be selected if the specified SMTP server requires secure connection using TLS If th
161. STN and to access the Internet from devices connected to the LAN e Manual II Administrator s Guide explains all QX IP PBX management menus available for administrators only It includes a list of all System Default Values e Manual III Extension User s Guide explains all QX IP PBX management menus available for extension users A list of all call codes can be found there too This guide contains many example screen illustrations Since QX IP PBXs offer a wide variety of features and functionality the example screens shown may not appear exactly the same for your particular QX IP PBX as they appear in this manual The example screens are for illustrative and explanatory purposes and should not be construed to represent your own unique environment QX IP PBX s Graphical Interface describes to the QX IP PBX s graphical user interface and explains all recurrent buttons Administrator s Menus explains the Administrator s management pages according to the menu structure shown on the main page of the QX IP PBX management Appendix PBX Services for QX IP PBX s Administrator explains PBX features for administrator accessible from the handset Appendix System Default Values lists all factory defaults Appendix Moderator s Menus explains all menus that can be accessed and configured by conference moderators Applicable if the Conference Server and or the Video Conferencing features are activated on the system Appendix Software License Agreeme
162. T1 connection will be established on the first available Se TEI while in non automatic mode a specific TEI may be reserved for aoe the connection In this case both call partners need to specify the ia same TEI in their settings ISDN L2 Timers E1 T1 Trunk ISDN Trunk PSTN Gateways Excessive Ack Delay T200 The SAPI Value text field requires an additional Service Access Point fle Timer T203 Identifier SAPI value digit values from 1 to 62 that is used to ISDN L3 Timers support additional interface between ISDN Layer 2 and Layer 3 T302 Timer Leaving this field empty default value only Call Control and Layer 2 Titas management procedures will be activated igi nnels Timeslots When Alternative Disconnection Mode checkbox is not selected QX D Channel Tinslt For Transmit Receive IP PBX will disconnect the call as soon as disconnect message has been received from the peer otherwise when checkbox is selected Bearer Establishment Procedure on progress indication with QX IP PBX s user may hear a busy tone when peer has been pe ees _ i Calling Party Type of Number Unknown v disconnected Called Party Numbering Plan ISDN telephony numbering plan Y Calling Party Numbering Plan ISDN telephony numbering plan Y In the Network Mode PBX connected Switch Type primary_dss1 v Generate Progress Tone to PSTN PBX None e If Non Automat mode is selected the same TEI address should be specified on both sides QX IP PBX and PB
163. T28P Yealink SIP T41P Yealink SIP T32G Yealink SIP T38G Yealink SIP T42G Yealink SIP T46G Yealink W52P Yealink VP 2009 VP 2009P Yealink VP530 QX50 0X200 0X2000 Manual II Administrator s Guide Please Note In the model s list the Polycom phones with sign are also presented as Polycom xx Pre 3 3 0 due to backward incompability of UCSoftware 3 1 1 configuration It is recommended to use Pre 3 3 0 models with Application SIP software 3 2 2 0477 The Programmable Keys Configuration page is used to assign a function to the programmable keys of the IP phone The design of this page depends on the IP phone model However independently on the IP phone model this page contains a number of the Programmable Keys and Functionality drop down list assigned to each of them 70 pya The following options are available in the Functionality drop down list Watch Ext watch the extension on the QX IP PBX and a possibility to pickup the call addressed to that extension E1 T1 Trunk ISDN Trunk PSTN Gateways IP Lines IP Line Settings IP Phone Templates IP Phones Logo FXS Gateways o Park Answer Ext on the phone can be visible as dc E R E PkA Ext PrkA Ext PrkAn Ext or PrkAns Interfaces sb Ext watch the calls parked to the corresponding lheit extensions and a possibility to retrieve the calls Key Fundtion Vawe d parked to that extension LineKey1 Watch Ext 105 Y w105 This list also contains a
164. TN cal a a Aug 2 12 18 13 sec P W 192 168 74 126 1 extension o 32x 1 odec uality 1 excellent needed leave the text fields empty The From field must ed IAS j indicate an earlier date and time from that which is indicated 06 Aug 2014 12 14 19 6s 103 PSTNGW 192 168 74 126 41164 a Quality 1 excellent in the To field Otherwise the error message Minimal date Aug 2014 12 12 54 PSTNGW 192 168 74 126 1 Grandstream GXP1400 103 Codec PCMU Quality 1 excellent i i lt 165 gt PSTN call should be less than maximal date prevents filtering and a gt ea searching 05 Aug 2014 12 13 02 C 2 104 Codec PCMU Quality 2 good e The From and To drop down lists offer a search by the Call PETERE n Duration specified by the list of values The field From must AZ m m are indicate a shorter duration than the field To Otherwise the a m w Sii error message Minimal duration should be less than maximal duration prevents statistics filtering Previous 1 2 3 Next e The text fields Calling Phone and Called Phone require the NA calling and called party s SIP address extension number or PSTN number as search criterion Wildcard symbols are allowed here Fig II 247 Call History Successful Calls page The Call History Successful Calls Missed Calls and Unsuccessful Outgoing Calls tables are lists of successful missed and unsuccessful incoming and outgoing calls and their parameters Call Start T
165. The update process takes about 5 minutes Normal operation will be stopped during that time not amp Maintenance Fig II 287 Firmware Update page The Image Check field will display invalid if the image does not correspond to the hardware version The Current Software Version field shows the old software cia ve rsio n The New Software Version fiel d shows the new versio n It is recommended to backup the configuration prior to upgrading the firmware k You can do that right now by clicking the following link Download Configuration of the S O ftware 1 mage Warning Make sure the Firmware Update process is not disrupted until it is completed A power down while upgrading may cause serious damage The update process takes about 5 minutes Normal operation will be stopped during that time Last Status 2014 08 07 03 26 New firmware version 5 3 25 is downloaded Image check valid This page needs to be confirmed in order to continue image Panes updating If you are sure that the image version is appropriate for your device press Update otherwise press Discard corertcatwareverson EEE Do you really want to proceed Update Discard Fig II 288 Firmware Update page If you have confirmed the firmware version a new page with firmware update progress will be displayed next There are no functions available on this page just last status about the firmware update procedure At some point the connection with the d
166. Timeout 5000 0 120000 msec digits in a range from 0 to 255 required to establish a call When field has 0 value system uses either timeout defined in the Incoming digits timeout field or the End of Address eine la messages to establish a call Independent on the value in this field the message End of Address always causes the call establishment Dialing Delay Timeout 500 0 2000 msec Unused A B C D text fields require to configure unused C and D bits of E1 T1 CAS signaling A and B bits are predefined Fields may have either 0 or 1 values Previous Fig l 119 CAS Signaling Wizard Page 5 Invert A B C D text fields are used to invert the ABCD status bits in time slot 16 before TX and after RX If bit is set to 1 the router inverts it before transmission and after the receipt End of DNIS I 15 checkbox is used to enable End of DNIS service Collect Call checkbox is only available when Brazil is selected in the Country drop down list on the previous page of the wizard and when the PBX attached to the QX supports this feature When this checkbox is selected and in case of incoming calls always the called destination will pay for the call Option is particularly applicable when calling from the mobile phone Checkbox should be selected when the appropriate feature is enabled on the PBX The Allow Timeslot Blocking checkbox indicates whether the system should use blocked timeslots to make outgoing PSTN calls If this chec
167. User Rights Backup Upload Firmware Get Firmware From Server Automatic Firmware Update Firmware URL text field requires the path of new firmware image din Firmware update which located on the FTP server It is recommended to backup the configuration prior to upgrading the firmware You can do that right now by clicking the following link Download Configuration Username and Password text fields require the FTP server dd aie td auth enticatio n p aram eters 4 The update process takes about 5 minutes Normal operation will be stopped during that time amp Maintenance Firmware URL ftp 192 168 0 2 Pub Images QuadroA20 5 3 25_2014_Aug You should save changes before Download or Download and Update User Name Levon_Dadayan Password Download Download and Update Fig II 286 Firmware Update page Pressing the Download functional button a new page with firmware download process will be displayed a Firmware update This page displays non editable information about the image It is recommended to backup the configuration prior to upgrading the firmware validity Last Status shows that firmware download process is T You can do that right now by clicking the following link Download Configuration a ia r F Warning Make sure the Firmware Update process is not disrupted until it is completed running and whether the new firmware version is downloaded or A power down while upgrading may cause serious damage
168. X Dial plan for the incoming calls Incoming Called Digits Size e If Automat mode is selected the user on PBX side will Incoming Interdit Service have the opportunity to set any mode related to TEI a assignment in PBX configuration This will allow PBX Send ALERT Message on Call Ringing connection to the QX without providing the TEI address ici from QX In the User Mode CO connected the TEI assignment is dependent on CO settings P Asserted Identity Disable P Asserted Identity Override CLID with P Asserted Identity e Select Non Automat mode and insert the same TEI address provided by CO Use Redirecting Number Info Element with P Asserted Identity e Select any mode related to TEI assignment if automat TEI searching mode is selected on CO side Two groups of timers need to be provided These settings are adjusted according to the Service Provider requirements Fig II 121 Trunk CCS Signaling Settings page ISDN L2 Timers e The Excessive Ack Delay T200 text field configures the period in milliseconds digit values from 500 to 9999 between transmitted signaling packet and its acknowledgement received e The Idle Timer T203 text field configures the period in milliseconds digit values from 1000 to 99999 for E1 T1 client idle timeout ISDN L3 Timers e The T302 Timer text field requires the value for the T302 timer in milliseconds digit values from 0 to 15000 and indicates the time frame system is waiting f
169. a The Reset Statistics button is used to reset the chart and the table if enabled QX50 QX200 QX2000 SW Version 6 0 x 158 QX50 0X200 0X2000 Manual II Administrator s Guide PSTN Channel Usage The trunk checkboxes are used to select the port number s over which the FXO traffic chart will be built At least one Trunk checkbox should be selected otherwise error message appears Please Note The PSTN Channel Usage page is not available for QX2000 The FXO Channel Usage Statistics page consists of following components used to define the chart parameters Trunk checkboxes are used to select the FXO line number s over which the FXO traffic chart will be built At least one Trunk checkbox should be selected otherwise error message appears Time range of statistic table drop down list includes the period in days statistics data that is to be collected and the corresponding diagram chart that is to be built Incoming Calls and Outgoing Calls checkboxes are used to select whether the FXO traffic statistics for only incoming or outgoing or for both type of calls should be displayed in the diagram chart Maximum Active Calls checkbox is used to have the number of maximum active calls displayed in the diagram chart At least one of these checkboxes should be selected otherwise error message appears Show button is used to generate an FXO channels usage diagram chart over the parameters selected above When this button
170. a moderator on the QX IP PBX login page After logging in as a moderator the page Conference Progress is displayed Here you may see the active conferences and the participants From this page you may also access the settings of the conference to operate and perform actions that are available only to the moderator of each conference e Conference Progress e Recorded Conferences 9 Conference e Conference Settings General Conference Progress Conference Progress Conference ID 888 Description Daily Conference Recording SIP Address 888 Duration 0 sec Customization Conference Waiting Unlocked Recording Disabled Status Participants Schedule Terminate Lock Unlock Start Resume Send Notification Mail Participants Active All 0 2 Add Delete Dial Out Set Speaker Set Listener Lecture Mode SIP Address Tel Participant Video Participant Participant Number Type Allowed Indication Status Name Request to Speak John Smith 113690sip epygi loc Speaker Yes 5 Joining Alice 5986744 Speaker Yes Not Active Dawson Refresh in 15 seconds Fig II 293 Conference Progress page QX50 QX200 QX2000 SW Version 6 0 x 184 QX50 0X200 0X2000 Manual II Administrator s Guide Conference Progress The Conference Progress page displays information about the conference including the list of participants and allows moderator to manage the conference The following read only data is displ
171. about critical and medium security issues This applies very strict password criteria for the SIP Registration password when configuring an IP phone e Medium The minimum strength of the IP Line passwords should be good The Security Report will generate warnings on all unsecured Call Routing rules IP Line passwords Firewall level if it is set to lower than Medium and disabled IDS Epygi treats system security with the utmost priority and has taken an active approach to provide users with information and tools to aid in maintaining system security It is highly recommended that users of an IP based system need to be familiar with industry best practices to maintain system security Limitation of Liability and Remedies In no event shall Epygi Technologies be liable for any consequential incidental direct indirect special punitive or other damages including without limitation loss of data loss of phone calls loss of business profits business interruption loss of business information or other pecuniary loss arising out of the use or inability to use the Quadro e High The minimum strength of the IP Line passwords should be strong The Security Report will generate warnings on the IP Line passwords disabled IDS unsecured SIP and unsecured Routing Rules to SIP PSTN and IP PSTN and also regarding the Firewall level if it is set to lower than High Fig II 15 System Security Management page Licensed Feature
172. ached from the Conference Progress page Participant Type list is used to select the type speaker or listener of the participant in the conference QX50 QX200 QX2000 SW Version 6 0 x 190 QX50 0X200 0X2000 Manual II Administrator s Guide Confirmation Type list is used to set the password protection for the participant joining the active conference Star selection allows the participant to accept the conference invitation by pressing the button Only participants connected to the conference with the moderator password will be provided with the permissions to manipulate the conference Please Note Confirmation Type should be selected to none when the Participant Type is listener A group of checkboxes on this page allow configuration of participant specific settings e Allow Video checkbox will allow participant to join the video conference This checkbox is not available on this page when it is reached from the Conference Progress page 2 When the Dial Out checkbox is selected the participant will be automatically dialed out when the conference is activated e Activate On Dial In automatically activates the conference when this participant joins the conference call This checkbox is not available on this page when it is reached from the Conference Progress page o Participant Indication enables the beep indication during the conference when this participant joins or leaves the conference 0 Nested Conference sho
173. ackets receive due to lack of resources The area Transmit Values provides the following m a otal number of dropped packets transmit due to lack of resources Statistic as readable values e Transmit Bytes number of transmitted bytes time hour RX Bytes RX Dropped RX Multicast TX Errors TX Dropped 0 0 e Transmit Packets number of transmitted Ethernet packets e Transmit Errors number of transmitted packets containing errors e Transmit Drop Errors number of transmitted packets that have been discarded e Transmit Carrier Errors number of transmit carrier errors that occur due to a defective or lost connection on 1066757 o o o o o o o o o o o o o o OoO o o o 0 0 0 elalalealealealelalelelelelsalelo lola 5 aoe e olojojojolololojojolojojojolololojololjo the Ethernet link e Transmit Collisions number of transfer errors that 3631312 occurred during a simultaneous packet transmission from an both sides recumustes 42513479 Reset Statistics Fig II 259 Transfer Statistics Diagram Chart To see the Transfer Statistics Diagram Charts select the desired criteria and click Save to generate the corresponding chart and the table showing the transfer statistics values if enabled The letters M millions and K thousands used in the legend of the displayed diagrams show the total number of specified criteri
174. actions on some events on the QX IP PBX or remotely modify QX IP PBX s settings SNMP Settings page is divided into two pages Global SNMP Settings and SNMP Trap Settings Global SNMP Settings are used to enable the SNMP agent on the QX IP PBX to select the SNMP protocol version for communication with the administrating application and to define the community for administrating application to connect the QX IP PBX Global SNMP Settings Enable SNMP checkbox is used to enable SNMP agent on the QX IP PBX System Location text field requires optional information to describe the network where SNMP management is performed System Contact text field requires optional information about the contact person responsible for the SNMP management in the defined network Field may indicate the point person s name Overview CP DNS PPP PPTP email address phone number or other contact information Global SNMP Settings Enable SNMP v1 2c checkbox is used to enable SNMP v1 2c protocol version for the messaging between QX IP PBX s SNMP Y Enable SNMP agent and the administrating application If this checkbox is not selected SNMP v1 will be implied niran SNMP v1 v2c Read Only Community text field is used to insert A caaaom the community description public private etc for the read only Enable SNMP v1 2 Read Write Access management like gathering information events statistics etc SNMP v1 2c Read Write Community public ab
175. ading in the tables is created as a link By clicking on Conference History Conferences the column heading the table will be sorted by the selected column Upon sorting ascending descending arrows will be displayed Numberof Records Conf Total Duration Conf Maximum Duration Conf Average Duration Conf Minimum Duration close to the column heading 5 1 min56 se 50 sec 23 sec ae Status The Activation Reason column indicates whether the participant is a key member to start the conference i e when participant dials into the conference the conference is getting automatically activated and the dial out participants if any are called to join the conference see Conference Progress Activation Reason Activation Details dd Month yyyy hh mm ss T h e A ctivatio n D etails co l umn p ro vi de S 1 nfo r mati on ab ou t h OW th e ConfID Activation Time Conference Duration Participant Count Activation Reason Activation Details conference Call 1S activated 500 31 Oct 2014 17 34 47 8 sec Trigger 7412106 192 168 0 209 5060 500 31 Oct 2014 17 18 30 6 sec Trigger 7412106 192 168 0 209 5060 The Filter button performs searching within the statistics tables sc cia unica 500 31 Oct 2014 17 15 13 43 sec Trigger 7412106 192 168 0 209 5060 The search may be done with several criteria at the same time 500 31 Oct 2014 17 13 54 50 sec Trigger 7412106 192 168 0 209 5060 The following search criteria are availabl
176. ae ad gt A 147 A 147 ac AA o E EEE PP 148 DP Lmes Registration Scene E ceed om cede cmc ween eek E A 148 License Statis meee ener eo OO O OO O O eee ere er eee ene eee eee Te ne ee ee ee ee re ene er re cn rr terry 148 Events 149 Syste EVENtS AA aa aaa ao e o Ri 149 a AA zp o er eer er eee eee meee etree ener 149 Call TISCOLY siicatacasaiarcansaiernaaiaacaraariqataanseanaqnaacagaadiaacantasiansaiaaatandiaacainaiaanaanadsnseaaiaacaienisanaaiaanausaaiadanadataaacdiaainaaadiaucainaiaddatansinlaandiautainaisaraataataidusasaaaieuaunaiaataunaaiautaidassnenninaaas 150 Successful Missed and Unsuccessful Calls aa 150 OMS os a CLOS tancia 151 CTS AI ara 152 ALCOI SEL nai 153 aleja alo AA A e e EP Oe o 154 A no o o re eo o o 155 COMPETENCE History sao ARAN AAN TRATAR RARA SARTRE RAR AUT CARA AURA URREA AREA ROA 155 SR e SEO EEE o O E Om E S 155 pliccess ul Calls and Unstccesstal Outgoing Calls ricas 156 Conterentce History oe So NON 156 LAIN VV a E A A AE T E E eRaaid lpnaaearirORiON 157 LAN amd WAN Interface als CUS acca ce cx cast sn sepnctesenzacdescetsieciccvetned eqasresccnestacdcectuacs ansevesd eg ate sdenesencicerenacaencavandsqee taana EASA IRERE TAAR AEREE EAE EAE a RE aeS 157 1018 E e AP o o EEE 0 158 Network Transtet coccio E E ea 158 QX50 QX200 QX2000 SW Version 6 0 x 6 e C Pyg l QX50 QX200 QX2000 Manual IJ Administrator s Guide Poa Camel Yee ancora MEI eA AERA AIAN RARE naa IAMAInER
177. age 3 The Display Name text field allows you to replace an original caller s ID with the custom display name for the corresponding routing rule This field is optional and when it is left empty an original caller ID will be displayed on the called destination s phone otherwise the name inserted here will appear on the phone This field is not available for PBX Voicemail destination type routing rules The Remove Display Name checkbox is used to remove caller IDs from calls made with this routing rule This checkbox is not available for PBX Voicemail destination type routing rules The Next button will open the Call Routing Wizard Page 4 where different information about source caller will be required depending on the selected Source Type For the SIP source type the Source Host text field will require one or more IP addresses or host names of the SIP server where the caller is registered or the caller s device if they are direct calls separated by a space In case of FXO ISDN or E1 T1 source types selected Source Port ID drop QX50 QX200 QX2000 SW Version 6 0 x 97 pya QX50 0X200 0X2000 Manual II Administrator s Guide down list will require to select the FXO line number or ISDN E1T1 trunk correspondingly and on the next step a list of timeslot s used to receive calls from the defined caller The Call Routing Wizard Date Time Rules Add Entry page appears if the Set Date Time Period s checkbox previously had bee
178. age offers the following input options Priority requires a numeric value from 1 to 252 to define the priority of the routing rule The lower the number the sooner the routing rule will take effect higher priority From requires the packet source IP address and subnet mask of the specified destination to match with the rule Via IP address requires the IP address of the subsequent router for IP packet forwarding PPTP L2TP Routes The PPTP L2TP Routes allow IP packets forwarding through the PPTP and L2TP tunnels of the QX IP PBX If PPTP L2TP connections do not exist on QX IP PBX VPN routes cannot be generated The PPTP L2TP Routes table displays all generated VPN routes with their parameters Target State for the state of the route enabled or disabled Actual State for the state of the route connection up down or erroneous Route To for the subnet where the incoming packets should be routed Via Tunnel for the VPN tunnel incoming packets should be routed through and Tunnel State for the actual state of the route tunnel up or down The Add button opens the Add PPTP L2TP Route page where a new VPN route can be generated The Add PPTP L2TP Route page offers the following components Route Via contains the available PPTP and L2TP connections on the QX IP PBX A connection selected from this list will be used to route the IP packet from the QX IP PBX s LAN to the peer behind the PPTP L2TP tunnel Route To requires the IP
179. al and whether it is a PSTN call PBX call or IP call Fig II 238 Status Lines Status page Call Start Time shows the call start date and time Call Duration shows the current call duration RX Codec shows the codec used to encrypt the incoming packets TX Codec shows the codec used to encrypt the outgoing packets If RX and TX codecs are the same only one Codec field will be displayed For IP Line Status the following dynamical parameters appear on this page Username shows the IP phone s client name registered on the QX IP PBX Last Registered shows the date and time the corresponding IP phone has been last registered on the QX IP PBX Expires In shows when the last registration of the IP phone will expire Binding IP Address shows the IP address of the IP phone within the QX IP PBX s LAN network The list of supplementary services provides the following additional status information for each telephone line Enabled or Disabled For Incoming and Outgoing Call Blocking Speed Calling Hiding Caller Info Voice Mailbox and Group List services the number of Entries will be displayed in the corresponding service table For Voice Mail Service the voice mailbox configuration mode is displayed here This allows administrator to view the status and to be notified about services running on QX IP PBX for every line The services are designed as links that guide the administrator to the corresponding service page of the selected user The
180. all will automatically be parked to that extension The pickup user will be able to pick up the parked call from any destination by calling the extension where the call has been parked either by its PBX number or SIP address The authentication password will be prompted if configured of the call park extension in order to retrieve the parked call Please Note The Call Parking is valid for the period defined in the Call Park Extension Settings By default it is 15 minutes During that time hold music if configured will be played to the parked party When the Retrieve Timeout expires the phone that initiated the call parking will start to ring If no one picks up the parked call or if the phone is off hook the parked call will be automatically disconnected Please Note Anyone who wishes to retrieve the parked call will be requested to pass a password authentication if the password is defined for the call park extension to resume the parked call The parked call will be disconnected if an incorrect password has been inserted and authentication has been rejected To avoid unexpected calls received on the extension used for call parking it is recommended to use virtual extensions for the Call Park service Barge In Service Attention The Barge In service is an optional feature and can be activated with a feature key from the Feature Keys page The Barge In service on the QX IP PBX allows the PBX users to participate to the third party s calls
181. aller Usually the display name appears on the called party s phone display when a call is made or a voice mail is sent Extensions Management Edit Entry Extensions O Go Back General Settings General Settings 103 SIP Settings Password requires a password for the new extension The extension password may only contain digits If non numeric symbols are entered the Incorrect Password no symbol characters allowed error will prevent creating the extension A ji ne SIP Advanced Settings Display Name Subject John Smith assword Pm Generate Password icensin If you are unable to define a strong password press Generate Password to use one of system defined strong passwords The Y GUI Login Allowed 3pcc Click2Dial Access Allowe Password field is checked against its strength and you may see how Selatan isting snow on Pubic Directory o Go To Line Settings Use Master Extension strong is your inserted password right below that field Go To Codec Settings Master Extension 101 Y Percentage of Total Memory 1 Y Confirm Password requires a password confirmation If the input is not corresponding to the one in the Extension Password field the Incorrect Password confirm error will appear Y Allow other users to Barge In to this extension Edit Call Barge In Intercept Access List Edit Watch Access List Attached Line lists all free lines an extension may be attached
182. allers should follow the voice instructions for configuring a new entry or reconfiguring existing entries in Authorized Phone database When system accepts the inserted settings the corresponding entry will be logged to the Authorized Phones Database The caller will then be disconnected from the QX IP PBX s Auto Attendant and the defined Call Back destination will receive a call from the QX IP PBX within the next 45 seconds Answering the incoming call the caller will be reconnected to the QX IP PBX s Auto Attendant Please Note The detected caller number must correspond to the one applied by the caller In case of PSTN call back at least one PSTN line must be available on the QX IP PBX There must be network connectivity and the destination must be reachable Non Permanent Call Back configuration service allows trusted caller to organize one time Call Back to the defined destination In this situation no entry will be logged to the Authorized Phones Database By calling QX IP PBX s Auto Attendant and entering the Auto Attendant menu the caller can use 00 menu see Feature Codes to modify the Call Back destination for already registered callers in the Authorized Phones Database The system will ask to login by dialing the number and an appropriate password for the QX IP PBX s extension that is used as login extension in the Call Back settings After login caller should follow the voice instructions for reconfiguring the existing entry in Authorized
183. allow you to select the frequency of checking for a new update Maintenance Y Enable Automatic Firmware Update Server Name ftp epygi com Server Port 21 Update Method Check and notify choose this selection if you only wish to a be notified about the new available firmware on the server With this selection on the indicated weekday Check for updates Check and notify_v EveryDay_v at 000_v and time on daily or weekly basis the QX IP PBX will sli check for a new firmware available on the server The way of notification is configured from the Events page Check and update choose this selection to check and automatically install the new firmware on the QX IP PBX as it becomes available on the server With this selection on the indicated weekday and time on daily or weekly basis the QX IP PBX will check for a new firmware available on the server will automatically download and install it on the QX IP PBX Fig II 290 Automatic Firmware Update page The Check Update Now button is used to manually initiate Check and notify or Check and update actions The action to be executed depends on the options selected above Reboot The Yes Reboot Device button is used to reboot the QX IP PBX Please note that the session with the QX IP PBX will be closed i e the QX IP PBX GUI should be newly opened and a f Reboot new login will be required afterwards Reboot Device You are about to reboot the device Yo
184. an be done manually Please Note Independent on the selected server type there should be an auto update folder in the root directory of the server QX IP PBX will check for any new firmware in that specific folder only Besides the firmware bin file the auto update folder should contain supplementary file s to point to the correct firmware file The detailed instructions on the functionality of automatic firmware update as well as server configuration are described in the Automatic Firmware Update document which you can find at the Epygi Web support portal This page consists of the following components The Enable Automatically Firmware Update checkbox selection enables the automatic firmware update service on the QX IP PBX The Server Name the IP address or hostname the Server Port and the Update Method should be defined The Update Method drop down list provides a possibility to choose among FTP HTTP or HTTPS methods For some of these selections authentication Username and Password can be entered User Rights Upload Firmware Get Firmware From Server MUENTE Automatic Firmware Update 1 Info Feedback will be given via the event system Please Note In order to use Epygi s public ftp server leave the Server Name Server Port Update Method User Name and Password text fields to their default values ftp epygi com 21 ftp and anonymous respectively use blank for password Check for updates options
185. and to define the appropriate permissions This page is only available when the Barge In Service is enabled from the Feature Keys page This page contains the following functional buttons Extensions Add Extension Add Multiple Extensions Bulk Import Add functional button opens an Add Entry page where extensions may be added to the Call Barge In Intercept Access List This Call Barge In Intercept Access List Add Entry amp Extensions Call Type PBX page requires the extension number in the Address text field that Adaress fios wilara supported will be allowed to intercept calls The wildcard is supported in the ani Address field to add a group of extensions with one entry da Atow sargetn The checkboxes on this page allow to select one or more options of the Barge In Service and Call Intercept for the extension Allow Listen In ie TERR Fig II 29 Call Barge In Intercept Access List e Allow Barge In e Allow Intercept Attention Barge In Call Intercept calls are not displayed in Active Calls table on the Administrator s Main Page nor are registered in the Call History are X 109 Call Barge In Intercept Access List of Extension 103 X 125 X 130 X 114 Fig II 30 Call Barge In Intercept Access List Add Entry QX50 QX200 QX2000 SW Version 6 0 x 26 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide Watch Access List The Watch Access List page is used
186. apter Allowed Characters and Wildcards QX50 QX200 QX2000 SW Version 6 0 x 93 epygl Number of Discarded Symbols requires the number of symbols that should be discarded from the beginning of the routing pattern The field should be empty if digits do not need to be discarded Only numeric values are allowed for this field otherwise the error message Error Number of Discarded Symbols is incorrect digits allowed only will appear le Call Routing Local AAA Table Global Speed Dial SIP Tunnel Class of Service Call Routing Wizard O Go Back Routing Call Type Add Entry Y Enable Record Prefix requires entering the symbols letters digits and any characters supported in the SIP username that will be placed in EN front of the routing pattern instead of the discarded digits The pe o wisi ance following tags can be used for this field ne Destination Number Pattern y Enabler Key 555 Y Filter on Source Modify Caller ID Y Set Date Time Period s Y Set Overall Calling Time Limit Y Set Tracing Debug Options on This Rule Fig II 138 Call Routing Wizard page 1 e lt callerid range gt used to apply the complete or a part of caller ID the caller s number detected during the call as a prefix For example lt callerid 1 3 gt indicates that the first 3 digits of the caller ID will be considered as a prefix lt callerid 3 end gt indicates that the caller ID from its 3rd digit and up to th
187. are assigned allocated to specific extensions View recent system notification messages Determine the action to be taken for events Display current list of successful calls originated or received List of missed unanswered calls Outgoing call attempts that did not complete Download current Call Detail Records or configure the number of call records to save Chronological display of archived Call Detail Records CDR Options for archiving call records View call records specific to conference calls View call records of outbound calls originated from a conference bridge View call records of unsuccessful outgoing call attempts made from a conference bridge Download conference call detail records or configure number of call records to save Show current activity of the LAN Local Area Network port Current activity of the WAN Wide Area Network port Show the activity of LAN or WAN ports over a period of time Show the activity on the on board PSTN FXO E1 T1 or ISDN channels over a period of time Successful Calls and Unsuccessful Outgoing Calls Fig II 235 Status Menu page Conference History Settings e LAN WAN LAN and WAN Interface Statistics e Statistics Network Transfer PSTN Channel Usage QX50 QX200 QX2000 SW Version 6 0 x e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide System Status General Information The General Information page includes
188. arm am The Enable Automatic Backup checkbox enables automatic Sendto server severname 9216874203 backup mechanism on the QX IP PBX GS Path on Server QXBackup Send Method The following group of manipulation radio buttons allows you to select whether the backup files will be delivered by email or stored in some location gt m TFTP e The Send via Email radio button is used to send the automatically backed up files via email The selection enables Email Address text field that requires the email address of the administrating person to receive the automatically backup files Backup Interval Selection Every Day Y at 19 00 Y e The Send to Server radio button is used to store the automatically backup files on a remote server This selection enables the following fields to be inserted Fig II 279 Automatic Backup page The Server Name requires the IP address or the host name of the remote server The Server Port requires the port number of the remote server The Path on Server requires the path on the server to store the backup files in The Send Method manipulation radio buttons allow you to select the remote server type TFTP or FTP In case of FTP selection the authentication username and the password need to be inserted In case these fields are left empty anonymous authentication will be used The Backup Interval Selection drop down lists is used to select the frequency and the time when the automatic backup of t
189. arty Numbering Plan and Calling Party Numbering Plan drop down lists indicates correspondingly the numbering plan of the called party s and calling party s number The Route Incoming Call to drop down list contains Attendant routing agent with two kinds of call routing possibilities and all extensions of QX and allows selecting the destination where incoming calls will be routed to Choosing the Routing with inbound destination number selection will request the authentication if enabled and then will automatically use the initially dialed number to connect the destination without any additional dialing Attention When QX acts in the Network mode with the Attendant as a destination to route the incoming calls to digit forwarding should be disabled on the private PBX side otherwise incoming digits may be mistaken as a special calling codes on the QX IP PBX s Attendant Switch Type is another configuration parameter that depends on the Service Provider when acting in the User mode and the private PBX capabilities when acting in the Network mode The Generate Progress Tone to PSTN PBX drop down list contains the options for sending progress ring back tone to callers from the PSTN PBX The following options are available in the list QX50 QX200 QX2000 SW Version 6 0 x 83 QX50 0X200 0X2000 Manual II Administrator s Guide e None configures the system to send ALERT messages without the Progress Indicator information element IE
190. ation The IP Pool Configuration is used to add groups of IP addresses that EEES have the same restriction criteria When adding a new filtering rule ep See groups may be used instead of several IP addresses IP Pool a esos Configuration offers the following components View makes hidden groups visible Hide makes group members hidden and adds the HIDDEN comment f Fig ll 193 IP Pool Configuration page in the member column Add opens the Add Group page where a new group may be added This page consists of the Group Name text field requiring the group name and the Group Description text field requiring the optional group description as well as standard Save and Go Back buttons to apply or abort changes Edit opens the Edit Group page where the service parameters can be modified It provides the same components as the Add Group page To operate with Edit only one record may be selected otherwise the error message One row must be selected will appear Please Note Changing a group name will also change the references to this group including groups where this group is a member of and all affected filter rules enabled and disabled ones in all chains Deleting a group will also delete any reference to the corresponding group including filter rules and member relations to the other groups IP Pool Configuration Add Group EpygiGroup Firewall roup Description Epygi Members Clicking on the Group name wi
191. atistics page provides detailed information about the established call is provided When QX IP PBX serves as an RTP proxy this page displays two groups legs of RTP statistics For example when calling from an IP Phone attached to the QX IP PBX s IP line to an external SIP destination or from one external SIP destination to another through the QX IP PBX s Auto Attendant Each group of parameters describes characteristics of a piece of RTP stream composing an overall SIP session Normally one leg describes the RTP stream from caller to the QX IP PBX and the other leg describes the RTP stream from QX IP PBX to the destination Quality estimated call quality which depends on RTP statistic Below is the legend for Call Quality definitions on the displayed RTP Statistics excellent RX Lost Packets lt 1 amp RX Jitter lt 20 good RX Lost Packets lt 5 amp RX Jitter lt 80 satisfactory RX Lost Packets lt 10 amp RX Jitter lt 150 bad RX Lost Packets lt 20 amp RX Jitter lt 200 very bad RX Lost Packets gt 20 or RX Jitter gt 200 epygi RTP Statistics Local 192 168 25 50 9120 Remote 192 168 70 218 53330 Quality 1 excellent RX Codec Received Packets Received Packet Size RX Lost Packets RX Jitter RX Maximum Delay RX Delay Increase Count PCMU 0 0 RX Delay Decrease Count 0 TX Codec Transmitted Packets PCMU 109 Transmitted Packet Size 160 Loca
192. authorized to access an extension for e Authorized Phones features such as voice mail or call relay and call back Fig IT 22 Extensions Menu page QX50 QX200 QX2000 SW Version 6 0 x 22 QX50 0X200 0X2000 Manual II Administrator s Guide Extensions Management The Extensions Management page is used to create a variety of extensions and auto attendants on the QX IP PBX From this page by clicking on the user extension the Administrator can go to the extension settings pages When this page is accessed for the first time after the QX IP PBX s initial boot up or the default configuration settings restore an intermedi is displ i atore a termediate dd d Sp ayed Choose Extensions Length Extensions The Choose Extensions Length page is used to define the extension settings applicable to all extensions on the QX IP PBX This page disappears once being saved Leave Current Length 3 Change Length The Choose Extensions Length page consists of a radio button selection Save Fig IT 23 Extensions Management Add Entry page e Leave Current Length radio button selection is used to leave the current length of extensions on the QX IP PBX Per default the extensions length on the QX50 QX200 is 3 and on the QX2000 is 4 In front of this selection the actual configured length of extensions is displayed e Change Length radio button selection is used to change the actual length of extensions on the QX IP PBX This sel
193. ayed on this page Conference ID the unique ID on the conference Info Text displays the text uploaded in the Info File from Customization page In the picture illustration on the right Conference Conference Progress Conference Progress Conference ID 888 Description Daily Conference SIP Address 888 Duration 0 sec side the Info Text says WELCOME to EPYGI s Uniocked Recording Disabled CONFERENCE Description any descriptive information about the conference optional Participants Add Delete Dial Out Set Speaker Set Listener Lecture Mode SIP Address the SIP address of the conference SIP Address Tel Number Type Participant Video Allowed Participant Indication Participant Name Request to Speak Duration the time the current conference is active John Smith Conference Status the conference status active not active Alice or waiting If the conference is active the information whether the conference is locked or not and the recording status recording started recording paused and recording stopped is also displayed herein 11369 sip epygi loc Speaker Yes Joining 5986744 Speaker Yes Not Active Refresh in 15 seconds Fig II 294 Conference Progress page The following buttons are available on this page to manage the active conference Activate available for an inactive conference only and used to activate the conference T
194. be greater than the longest prefix defined in the Incoming DNIS Prefix text field otherwise the error message will appear The Description text field requires an optional description for an E1 T1 dial plan entry The Restore Default Settings functional button is used to restore the locale specific E1 T1 dial plan entries ISDN Trunk Settings The Integrated Services Digital Network ISDN is distinguished by digital telephony and data transport services offered by regional telephone carriers ISDN involves the digitization of the telephone network which permits voice data text graphics music video and other source material to be transmitted over existing telephone wires The ISDN Basic Rate Interface BRI service offers two B channels voice transfer and one D channel signaling data transfer The BRI B channel service operates at 64 kbit s and is meant to carry user data The BRI D channel service operates at 16 kbit s and is meant to carry control and signaling information although it can support user data transmission under certain circumstances The ISDN service allows QX ISDN Gateway act as a user or as a network If connected to a private PBX the QX ISDN Gateway should be configured in the network mode If an ISDN trunk from the CO Central Office is connected to the QX ISDN Gateway it should be configured as a user QX supports the MSN Multiple Subscriber Number service i e it can be subscribed to multiple numbers from the CO and
195. be deleted To abort the deletion and keep the PPTP L2TP connection in the list click No rrF QX50 QX200 QX2000 SW Version 6 0 x 143 epy8l QX50 0X200 0X2000 Manual II Administrator s Guide Status Menu The Status Menu consists of the following sections e System Status ae 3 ball Administrator admin General Information Network Status Lines Status Memory Status Hardware Status SIP Registration Status IP Lines Registration Status License Status e Events System Events Event Settings e Call History Call History Settings Call History Archive Archiving Settings e Conference History Conferences Successful Missed and Unsuccessful Calls Overview Overview System Status General Network fil Status Events Event Settings Call History CDR Archive Archiving Settings Conference History Conferences Successful Calls Unsuccessful Outgoinc Calls CDR Settings Statistics Network Transfer Display system host name uptime duration and firmware release View system interface settings and active services Display configured IP Line FXS and FXO ports Display available and allocated system memory Display status of various interface ports Display extensions registered to an external SIP server List configured IP lines and the registration status of the IP phone List the software licenses that
196. becomes no more assigned to your extension and is now available for other users to login and use it FXO Settings The FXO Settings are used to configure the FXO support that allows QX IP PBX to connect to other PBXs or analog telephone lines The number of available FXO ports is dependent on the type of your QX IP PBX QX50 has two FXO lines and the QX200 has four FXO lines available The QX2000 has no own FXO lines only shared FXO lines are displayed in this page The FXO Settings allows you to limit incoming or outgoing calls for the selected FXO line if required Depending on configuration of the FXO gateways multiple shared FXO ports from one or more FXO gateways may be available on the QX IP PBXs thus giving you the option to use them simultaneously The administrator may assign a default recipient for each FXO line where calls from the Central Office PSTN will be routed The assigned recipients become the QX IP PBX default users If the QX IP PBX Auto Attendant has been selected as a default user a caller from the PSTN needs to go through the attendant menu to reach the desired extension If the FXO service is disabled the Allowed Call Type Route Incoming Call to and PSTN number columns are set to N A Clicking on the FXO line number will open the FXO Settings FXO page where the FXO line settings may be modified The FXO Settings FXO page consists of the following components The Enable FXO checkbox selection
197. box is used to enable the redundancy functionality on the QX IP PBX Active Device Mode drop down list is only present on backup device and is used to adjust the behavior of the backup device eae during unavailability of master device When Active is selected vs backup device will become master once the original master device became unavailable When Passive is selected backup device stops Z Enable Redundancy its synchronization with the master device and will not take over A the control even when the original master got failed unless Swap Master Device button is pressed on the master QX IP PBX The Passive mode is used for firmware update or language pack ee updates on master device when a reboot is required After the reboot of master device the Active Device Mode on the backup device should be changed back to Active to restore the redundant network functionality Virtual IP Address Redundant Device Virtual IP Address Synchronization Interval Backup Device GUI Access Port Swap Master Device Save Fig IT 19 Redundancy Settings page Redundant Group ID text field unique ID values 1 and up identifying master and backup devices The same value must be set on both QX IP PBXs Virtual IP Address text fields require the virtual IP address of the device where the configuration is done Virtual Subnet Mask text fields require the virtual subnet mask of the device where the configuration is done These two parameters identify an alt
198. call activation to be defined Number of Records displays the current amount of conference Call History entries in the table For Conferences and Successful Calls pages Total Duration Maximum Duration Conf Average Duration and Minimum Duration statistics are organized at the top of the table The Records per page are used to select the number of displayed conference call statistic records per page The Previous and Next can be utilized to switch between these pages The Download Call Detail Records Conferences links are available below for all Conference Call History tables and allows you to download the displayed conference Call History in a text file Successful Calls and Unsuccessful Outgoing Calls The pages Successful Calls and Unsuccessful Outgoing Calls lists successful and unsuccessful outgoing calls and their parameters ETT ucts going cate Sting ConfID Activation Time Call Start Time Call Duration Calling Conference History Successful Calls Phone and Called Phone Each column heading in the tables is created as a link By clicking on the column heading the table will be a 2 ay 20 rour min sorted by the selected column Upon sorting ascending descending arrows will be displayed close to the column heading From Calling Phone The Details column is only present in Successful Calls table and a ta provides the following information de Monty emm dd Month yyyy mms e Brief information abo
199. caller ID indication either with the phone handset on hook or if the called party is already busy with another call or operation handset is off hook For internal calls caller ID notification in FSK can show up to two lines of identifiable parameters on the called phone s display The first line shows the caller s extension number The second line shows the caller s nickname if indicated in the configuration For external IP calls caller ID notification in FSK can also show up to two lines of identifiable parameters on the called phone s display The first line shows the caller s user name The second line shows the caller s nickname if indicated in configuration If the nickname is not available and there is a display name provided by the caller party the second line will display it otherwise the URL in the format username Ohost will be displayed For calls from the PSTN network the entire caller ID message will be shown DTMF Standard The DTMF standard supports caller ID indication only if the phone handset is on hook phone is free and ready to accept calls This standard also has caller ID notification conditions but they are non configurable Caller ID notification in DTMF can show only one line of identifiable parameters on the called phone s display For internal calls it is the caller s extension number For external IP calls it is the caller s user name For calls from the PSTN network caller ID will only display the calle
200. calls from SIP to E1 T1 if the SIP Invite message contains a P Asserted Identity or a P Preferred Identity or a Remote Party ID then the CallerID on E1 T1 contains the number from the user name field and the Redirecting Number IE contains the original number from the identity field SIP user agent should check for the existence of the P Asserted Identity then the P Preferred Identity then the Remote Party ID to fill the identity field For the calls from E1 T1 to SIP with Caller ID the SIP Invite message contains P Asserted Identity field with the original number value from the Redirecting Number IE on E1 T1 The SIP From field contains the value from the user name The E1 T1 Stats are not available in shared mode Incoming Interdigit Service The Incoming Interdigit Service is used to configure E1 T1 dial plan for the incoming calls from CO PBX to the QX This service allows you to speed up the call establishment procedure by detecting the prefix The calls will be speed up by the timeout defined in the Incoming Digits Timeout text field Incoming Interdigit Service When the system detects incoming dialed number starting with any Add Eat Delete Restore Deft Setting of the prefixes listed in the Incoming Interdigit Service table it will os Incoming DNI Pref Incoming DMI Size a wait for the rest of the digits as specified for the corresponding prefix in the Incoming DNIS Size text field see below Once all digits are received the syst
201. ccording to each defined skill The skill grading range starts from 0 and goes up to 10 with 0 meaning the absence of that specific skill and 10 meaning the highest level The termination phone number defines the phone assigned to agent In other words the calls on some termination number assigned to agent should be answered by that agent The agent may have only one termination number and changing that number will result in answering the calls to that agent in different location Agents are being managed from ACD Agents Table see ACD Group Extension Settings Agent Group Agent Group AG is actually a QX IP PBX extension with enhanced capabilities The type of that extension in QX IP PBX configuration is ACD Group see ACD Group Extension Settings Except for regular attributes intrinsic to extension like extension number SIP user name etc it is characterized also by the collection of agents included into that group call queue and the call distribution mechanism These agent group specific parameters of extension are being configured from ACD Group Settings or ACD Agents Table accessible from ACD Group Extension Settings Call Queue of Agent Group Agent Group receives the calls from customers via means existing currently on QX IP PBX For example it may receive the direct call through ITSP on SIP number DID number assigned to AG receive a call through ACD s IVR on AG s extension number external call through Call Routing Table on
202. ces Management The Conferences page displays a table with the existing conferences on the system This page allows you to create new conferences and manage the existing ones The following columns are present in the Conferences table e Conference ID indicates the unique ID of the conference This number is used from Auto Attendant to reach the conference The Conference ID is also used as the username for the moderator when logging into the QX IP PBX e Display Name any optional information about the conference e Description any descriptive information about the conference e SIP Address displays the SIP address of the conference e Status indicates the status of the conference Active Non Active or Waiting Clicking on the conference status link will display the Conference Progress page with detailed information about the conference status participants in the conference and description of each participant This page additionally allows the administrator to drop a participant from the conference or invite new participants It also allows the moderator to start stop resume pause the conference recording and to terminate the conference e Percentage of System Memory indicates the conference related memory space in percents dedicated to conference recordings and the conference specific custom system messages ae Gea Conferences Management e Codecs column lists the short information full information is wanes seen in the to
203. cessed by the extension s login see Manual III Extension User s Guide Besides this the details of the extension will be displayed in the Public Directories on the snom and Aastra SIP phones Leave this checkbox unselected if the extension is reserved or not used or when the extension serves as an intermediate unit for call forwarding etc Retrieve Timeout text field requires a timeout in minutes during which the parked call will stay active i e the parked user will remain on hold e If the Customize push back number checkbox is not enabled and the call park retrieve timeout expires the hold music stops playing to the parked user and a new call is being placed towards the extension initiating the call park If the extension initiating the call park does not answer the call the caller which has been recently parked will reach the extension s Voice Mailbox if enabled otherwise will be disconnected e If the Customize push back number checkbox is enabled and the call park retrieve timeout expires the hold music stops playing to the parked user and a new call is being placed towards the push back number configured in the Customize push back number field If the push back number configured in the Customize push back number field does not answer the call the caller which has been recently parked will reach the extension s Voice Mailbox if enabled otherwise will be disconnected The Customize push back number field consists of the following comp
204. checkboxes of the corresponding connections that should to be deleted stopped started from the Connections tables Click on the Delete Stop Start button from the table s menu to perform the corresponding operation for the selected IPSec connection s If deleting confirm it with pressing on Yes The IPSec connection will be deleted To abort the deletion and keep the IPSec connection in the list click No nrp RSA Key Management The RSA Key Management sub page is used to see the current RSA key and to generate a new one This page contains the following components The public key is displayed in the RSA Public Key text field so that the user may inform their IPSec connection partner about it for example via fax The user has the option of generating a new pair of keys by specifying the key length with the corresponding radio buttons Generate a new 1024bit RSA Key and Generate a new 2048bit RSA Key and then clicking the Generate Button PPP PPTP SNMP Ov PPTP L2TP IPSec Configuration Connec tion RSA Key Management A valid RSA key should fit to following requirements 1024 bit RSA public key Email this key to peer OsAQNtH6iQseeuUjq 2484havBeO RSA key doesn t start with 0s eee E G25Rx3H8Sg7liHCgnh8jINmcZCp Send e RSA key doesn t end with Generate a new 1024 bit RSA key Generate a new 2048 bit RSA key e RSA key contains symbols other than Alphanum Generate The Email this to the peer
205. cked by SIP UA Reason No Such Line Configured Date 05 Aug 2014 19 37 13 Disabled All Blocked 172 30 4 52 Blocked by SIP UA Reason No Such Line Configured Date 05 Aug 2014 15 42 00 Filtering Rules Fig II 188 Filtering Rules page View All Incoming Forwarding Outgoing Management Access Call Control Access SIP Access Blocked IPs Filtering Rules This allows trusted hosts to reach your network and vice versa If a host also appears in the Blocked IP List the Blocked IP List has a higher priority and the traffic will be blocked Y Firewall Enable Disable Add Edit Delete State Service Action Restricted IP Enabled All Allowed 192 168 0 0 16 Description Fig II 189 Filtering Rules page The table displayed on the bottom of this page shows the filters selected above specified by their State enabled or disabled the selected Service the set Action allowed or blocked the IP addresses the filters apply to if Restricted and the destination of port forwarding Redirect to in case of Incoming Traffic Port Forwarding With the exception of View All the table offers the following functional buttons e Enable is used to enable the rule If no records are selected the error message No record s selected will appear e Disable is used to disable the rule If no records are selected the error message No record s selected will appear e Add opens a filter specific page where new rules may be defin
206. configured as well as PPTP and L2TP server settings can be adjusted The page consists of 3 sub pages Connections The Connections page lists all existing connections are listed characterized by their Connection Name Type of the connection PPTP or L2TP the Client Server mode the State of the connection and the Remote Hostname IP the IP address or the hostname of the connection peer The state of the PPTP and L2TP Connections except for the Stopped state is established as a link that refers to the page where logout information about the connection status is displayed Logs can be useful to determine Sa Stop Aaa Edt Delete problems on PPTP or L2TP connections failure Network Connection Name Type Client Server Remote Host IP to74554Li L2TP Client 192 168 74 55 Connected Add functional button leads to the PPTP L2TP Connection tsoon Serve 1921687415 Connected Wizard page where a new connection can be established DHCP DNS PPP PPTP PPTP L2TP Configuration Connections PPTP Server Configurations L2TP Server Configurations Please Note After creating a PPTP server connection PPTP connections between devices placed on the QX IP PBX LAN and external devices will no longer be possible The PPTP pass through service for incoming and outgoing traffic will be automatically disallowed once a PPTP server connection is created Fig II 229 PPTP L2TP Configuration Connections page The PPTP L2TP Connection
207. ctive This checkbox enables the Enable TEI Remove Procedure and Permanent TEI Value checkboxes With the Enable TEI Remove Procedure checkbox is selected the trunk will lose the assigned TEI when entering into passive mode on the Layer 2 With the Permanent TEI Value checkbox is selected the trunk will keep the assigned TEI when entering into passive mode on the Layer 2 or when QX detected ISDN link DOWN signal from carrier These checkboxes are present only for connection types different from PTP Point to Point selected on the first page of ISDN Wizard In case if PTP Point to Point connection type is selected on the first page of the ISDN Wizard these two checkboxes are replaced with a TEI Address text field that requires the channel number digit values from 0 to 63 for connection establishment between the CO and the ISDN client Channel Selection is used to select between the Preferred and Exclusive B channel selection methods For Preferred channel selection the CO answers to the call request by the first available timeslot With the Exclusive channel selection the CO should feedback only by the timeslot asked in the call request The Bearer Establishment Procedure drop down list allows selecting the session initiation method on the B channel One of the following options can be selected for the transmission path completion prior to receipt of a call acceptance indication e on channel negotiation at the destination interface e on progre
208. d Warranty Epygi Technologies LTD Epygi warrants to the original end user purchaser every Quadro and QX to be free from physical defects in material and workmanship under normal use for a period of one 1 year from the date of purchase proof of purchase required or two 2 years from the date of purchase proof of purchase required for products purchased in the European Union EU If Epygi receives notice of such defects Epygi will at its discretion either repair or replace products that prove to be defective This warranty shall not apply to defects caused by i failure to follow Epygi s installation operation or maintenance instructions ii external power sources such as a power line telephone line or connected equipment iii products that have been serviced or modified by a party other than Epygi or an authorized Epygi service center iv products that have had their original manufacturer s serial numbers altered defaced or deleted v damage due to lightning fire flood or other acts of nature In no event shall Epygi s liability exceed the price paid for the product from direct indirect special incidental or consequential damages resulting from the use of the product its accompanying software or its documentation Epygi offers no refunds for its products Epygi makes no warranty or representation expressed implied or statutory with respect to its products or the contents or use of this documentation and all accompanyi
209. d all logs Fig II 270 System Logs Settings page System Logs System Logs Settings Remote Lo Remote Logs Settings 2 Enable Remote Logging Telnet via 645 port Y Enable Call Controlling Log Y able SIP User Agent Log Enable Media Stream Log Enable DSP Log Y Enable SIP Registration Log Y Enable System Messages Log Maintenance Enable FXO Agent Log En E El Y Enable FOIP Log Y Enable Voice Mail System Log Enable SPMG Agent Log En Enable SIP Subscription Log able Presence Log Y Enable ACD Log Save Fig II 271 Remote Logs Settings page System Logs System Logs Settings Remote Logs Settings System Logs Archive E Delete amp Download Unpacked size on disk 08 Sep 2014 02 Sep 2014 18 Aug 2014 07 Sep 2014 30 Aug 2014 22 Aug 2014 26 Aug 2014 21 Aug 2014 01 Sep 2014 23 Aug 2014 27 Aug 2014 09 Aug 2014 10 Sep 2014 Fig II 272 System Logs Archive page 164 QX50 0X200 0X2000 Manual II Administrator s Guide User Rights Management The User Rights service sets restrictions on the GUI access for various users permits or denies the access to certain Web GUI configuration pages and creates multilevel user management of the QX IP PBX The feature is useful to the ISPs in order to set the restrictions for certain customers to manage the QX IP PBX s configuration The User Rights Management page consists of two pages The Users page is used to manage the available users on the QX IP PBX The Roles page
210. d be deleted from the IP Pool Configuration table 2 Press the Delete button on the IP Pool Configuration page 3 Confirm the deletion by pressing on Yes or cancel the deletion by pressing on No QX50 QX200 QX2000 SW Version 6 0 x 125 epygl SIP IDS Settings The SIP IDS Settings page includes the following components Enable SIP IDS checkbox selection allows to prevent the SIP attacks The Add the IP address into the Blocked IP list in Firewall checkbox allows to block SIP attacker s IP address SIP attacker s IP address will be blocked by QX IP PBX Firewall and will be added on the Firewall Blocked IP List table The Discard SIP messages from IP address for checkbox allows to discard the accumulated SIP messages from the QX IP PBX SIP cash after defined timeout default timeout value of Discard SIP messages from IP address for service is 32 seconds The Exceptions link leads to the Exceptions for SIP IDS page where user can require the trusted IP address es that can t be blocked Add opens the page Exception IP Add Entry where a trusted IP address can be established Delete removes the selected exceptions from the Exceptions for SIP IDS table QX50 QX200 QX2000 SW Version 6 0 x Y Firewall QX50 0X200 0X2000 Manual II Administrator s Guide Firewall Filtering Rules Custom Services gt Groups SIP IDS System Security Management Y Enable SIP IDS Actions to perform after the detection Y
211. d entered in the text fields above and then uses the Basic Authentication method to notify the provider about the user authentication settings Most of the DynDNS providers require an authentication for security Authentication parameters can be provided in the URL text field to be used for the HTTP get request The Basic Authentication checkbox can be selected if no authentication parameters to be provided PPP PPTP Settings The PPP PPTP Settings page available only for QX50 0X200 is used to establish a connection over the DSL link or any other type of uplink to the ISP A connection is needed to set up and make or receive calls through PPP over Ethernet The connection may be configured for manual setup or always up Once a connection has been established between the QX IP PBX and the provider QX IP PBX users will be able to make and receive calls at any time The PPP PPTP Settings page offers the following components The PPTP Server text fields are only enabled when QX IP PBX is running with the PPTP interface and require the IP address of the PPTP server The Encryption drop down list is only enabled when QX IP PBX is running with the PPTP interface and it is used to select the encryption for the traffic over the PPTP interface Authentication Settings require the Username and Password used for the authentication on the ISP server Dial Behavior radio buttons enables the following selections DNS PPP PPTP SNMP e Dial Manually if
212. d receive calls Instead inactive extensions have a voice mailbox available to store the messages from callers QX50 0X200 has two available FXS lines Attendant extensions are dedicated to the IVR system on the QX IP PBX These extensions are used by callers to reach QX IP PBX s users and use the remote access and call relay services It is possible to create Auto Attendants with the custom scenarios By default QX IP PBX has one Auto Attendant extension 00 which is undeletable Attention QX50 is limited to 200 extensions QX200 is limited to 400 extensions and QX2000 is limited to 2400 extensions The Extensions table is a list of all extensions and their parameters Extensions Conferences Authorized Phones Ov V Add Extension Add Multiple Extensions Extensions Management Recordings Directory Receptionist Bulk Import Extensions Total extensions count 123 208 Add Edit Delete Show all extensions Use Epygi SIP server Extension y Display Name Attached Line SIP Address Percentage of System Memory External Access Attendant FXS1 FXS 2 IP Line 1 IP Line 2 IP Line 3 IP Line 4 Rem107 IP Line 5 R IP Line 6 IP Lime 7 QX50 QX200 QX2000 SW Version 6 0 x 741200 192 168 0 209 5060 10 11 7412101 192 168 0 209 5060 7412102 192 168 0 209 5060 7412103 192 168 0 209 5060 7412104 192 168 0 209 5060 74121050192 168 0 209 5060 74121060192 168 0 209 5060 7412107 192 168 0 209 5060 7412108
213. d when a caller joins the extension s call queue e Call Queue Message played when a caller is being held in the queue QX50 QX200 QX2000 SW Version 6 0 x 55 QX50 0X200 0X2000 Manual II Administrator s Guide The Universal Extension Recordings page consists of a table where the universal voice messages are listed Recordings Directory An Upload functional link is present for each voice message e Universal Extension Recordings recording that is not uploaded in the table and it is used to upload PA the custom system message When a message is uploaded the Upload functional link is replaced by Download and Remove Hold music no message is uploaded Voice Mail regular greeting no message is uploaded Voice Mail out of office greeting no message is uploaded functional links respectively These are used to download to the PC usa palas no message is uploaded and to remove the uploaded system message Outgoing call blocking no message is uploaded i Call Queue Welcome Message no message is uploaded The Memory Allocation group includes a drop down list used to Call Queue Message nn mestaye s uploaded specify the Percentage of System Memory for the universal Maso Aloci lan extension recordings The maximum value in the drop down list is Percentage of System Memory 1 _v equal to the maximum available space for voice messages on QX IP PBX Fig II 78 Universal Extension Recordings pag
214. de 3 Recording Storage Settings This group contains recording storage settings and is divided into rg Yare two groups Extensions Management Edit Entry The Modes radio buttons selection is used to choose the storage Extensions option once the call recording is done Following options are available E General Settings EESIN Recording Storage Settings 400 e FTP Mode this option will send immediately recordings to nae see the FTP server and delete from device This option will keep Recording Storage setings FTPMode immediate send recording to FP ser your device memory the most free gt Empleloca Mode Keep recordings locally When local space is full or maximum recording count is reached stop recording calls and generate an even O Simple Local Mode this option will keep recordings locally yates Mee Keep recordings locally When local space is full or maximum recording count is reached delete the oldest recordings Stop recording when local space is full and generate an event Keep recordings locally When local space is full or maximum recording count is reached move the oldest recording to FTP server e Cyclic Local Mode this option will keep recordings locally diia When local space is full delete the oldest recordings pues e Mixed Mode this option will keep recordings locally When local space is full or when Maximum recording count is Directory on Server S date_year S date_month S date_da
215. digit 0 You can add extensions of up to 20 digits long However the Call Routing Table won t be adjusted automatically you may need to manually adjust the routing rules for extensions in custom length Quantity 5 Fig II 71 Extensions Management Add Multiple Extensions page Start from the SIP User Name text field requires the SIP server registration user name for the first extension to be created Depending on the value in the Quantity text field the next extensions to be created will have subsequent SIP user names For example if you have inserted 30201 in this text field and the Quantity text field contains the value 5 then the 5 newly created extensions will correspondingly have the following registration SIP user names 30201 30202 30203 30204 and 30205 This user name is used for the registration on the SIP Server and should be unique on the SIP server This field length is limited by 20 symbols and is not limited regarding the use of symbols If an extension with the given SIP user name already exists in the Extensions Management table a next subsequent not used SIP user name will be used instead The Automatically attach to IP Line checkbox selection is used to automatically attach extensions to IP Lines Start From the IP Line text field requires the number of the new IP Line to be created The error message One or more IP Lines in the specified range are already attached to existing extensions appears if an IP line
216. digit patterns and set up options for call routes Call Routing Send all incoming SIP calls to the Call Routing table SIP Tunnel Settings en Local AAA Table Authentication table used with Call Routing for callers to pass authorization before being allowed to call out Global Speed Dial Common speed dial directory for all extensions Class of Service SIP Tunnel Create a SIP Tunnel between two locations best usage is to register a site with a Dynamic IP address to a site with a static IP address Call Reco rding Settin gs Class of Service Create Class of Service names that can be assigned to extensions to match rules in the Call Routing table Call Recording O NAT Traversal Settin gs Call Recording Choose extensions that will have Call Recording enabled NAT Traversal General Settings SIP Parameters General SIP Parameters STUN Parameters NAT Exclusion NAT options needed to make external SIP calls on the internet when on a private network Configure NAT traversal settings for SIP messages Configure NAT traversal settings for RTP packets voice and video Configure STUN server settings used for automatic NAT traversal IP adresses and subnets to exclude from NAT traversal needed for local or VPN connected subnets RTP Parameters Choose voice and video codecs or modify RTP port range used on this device STUN Parameters NAT Exclusion SIP Configure SIP ports and other general SIP setti
217. displays a table with the IP Lines registration information on the QX IP PBX General Network Lines Memory Hardware SIP Registration IP The table lists the IP lines and remote extensions registered on the QX IP Status IP Lines Registration Status PBX The table indicates the actual IP addresses of the remote devices the usernames by which the devices have been registered on the QX IP PBX as nn nn well aS the registration status informatio n Line locext104 es 172 30 0 254 06 Aug 2014 11 39 33 30 min 16 sec Line 3 Rem104 Yes 192 168 70 17 06 Aug 2014 11 43 00 33 min 43 sec Subscription Count field indicates used and allowed number of subscriptions for all IP phones registered on the QX IP PBX Subscriptions Subscriptions count ed allowed 3 100 Used Subscriptions distribution BLF MWI Total 0 3 3 are events originated by IP phones when watching other extensions on the QX IP PBX and when monitoring voice mailbox for new received voice mails Fig II 242 Status IP Lines Registration Status page When the allowed number of subscriptions is reached no new subscriptions are possible Typically the number of subscription should be keep reasonably below the maximum allowed number to avoid losing subscriptions Thus in case the actual subscription number is close to the limit configuration of IP phones should be adjusted to decrease the number of total subscriptions on the QX IP PBX Used Subscript
218. dress 00 15 65 45 74 19 Attached IP Lines 3 Based on the selected IP phone model and the inserted MAC Y Use Session Timer Address the IP phone can be automatically configured by simple Y Use Kickback reset reboot for more information about IP phone configuration refer to the corresponding IP phone s users manual The Attached IP Lines text field requires the numbers of QX IP PBX s IP lines used by the receptionist The IP lines should be separated by commas Fig l 83 Receptionist Phone Configuration Wizard Phone Model The Use Session Timer enables the SIP session timer for the IP lines specified in the Attached IP Lines text field This checkbox enables advanced mechanisms for connection activity checking This option allows both user agents and proxies to determine if the SIP session is still active The Use Kickback checkbox enables the kickback service on the corresponding receptionist When this service is enabled if receptionist transfers the incoming calls to the extension and if there is no answer or if the called extension is busy on another call the call is returned to the receptionist s phone instead of getting into Voice Mail Service or being disconnected To use this service receptionist should simply transfer the incoming call to the local extension In case of no answer or busy the call will automatically get back to the receptionist When this service is not enabled the incoming call will reach t
219. e Download Call Detail Records Conferences Fig IT 253 Conference History Conferences page e The text fields ConfID From and To are used for the search by ConfID Activation Time ConfID requires the unique ID of the conference For From and To fields the data must be entered in the format dd mm yyyy hh mm ss The time criteria are optional if it is not needed leave the text fields QX50 QX200 QX2000 SW Version 6 0 x 155 QX50 0X200 0X2000 Manual II Administrator s Guide e C Pyg l QX50 QX200 QX2000 Manual Il Administrator s Guide empty The From field must indicate an earlier date and time from that which is indicated in the To field Otherwise the error message Minimal date should be less than maximal date prevents filtering and searching e The From and To drop down lists offer a search by the Conference Duration specified by the list of values The field From must indicate a shorter duration than the field To Otherwise the error message Minimal duration should be less than maximal duration prevents statistics filtering e The From and To drop down lists offer a search by the Participant Count specified by the list of values The field From must indicate a shorter count than the field To Otherwise the error message Minimal count should be less than maximal count prevents statistics filtering e The text fields Activation Reason and Activation Details require the reason and the details ofthe conference
220. e Please Note Changing the Percentage of System Memory on this page will stop any recordings of universal extension voice messages from the handset Universal Extension Recordings Hold music The manipulation radio buttons on this page allows you to select the way custom hold music will be provided e Default Music enables the default music If the option is selected the text field Upload Recording will be disabled Recordings File selection is used to upload the hold music file The following option is available under this selection Edit Universal Extension Recordings Hold music Extensions Go Back Upload Recording text field can be used to type the path where hold music file is located If hold music file is browsed with the help of file chooser this field displays y dl Upload recording the path of the browsed file Choose File button is used t to browse for the hold music file RTP Channel Choose Channel QKChannel Default music Audio Line In The music file needs to be in PCMU CCITT u law 8 kHz 8 bit Mono wave format otherwise the system will prevent uploading the file and display the warning message Invalid audio file or format is not supported The system will refuse uploading also if there is not enough memory available for the corresponding extension and will then announce You do not have enough space Fig I1 79 Universal Extension Recordings p Hold musicage Please Note It is recommended to
221. e Refer to the user s manual of the corresponding SIP phone for instructions on performing a factory reset or reboot on any of the supported phones what additional configurations are required for a specific SIP phone and how to manipulate with the GUI e The Use Session Timer enables the SIP session timer for the corresponding IP line This checkbox enables advanced mechanisms for connection activity checking This option allows both user agents and proxies to determine if the SIP session is still active e The Use Template drop down list is used select a preconfigured custom template for the IP phone When the Use default is selected in this drop down list the template selected on the IP Line Settings page will be used QX50 QX200 QX2000 SW Version 6 0 x 68 QX50 0X200 0X2000 Manual II Administrator s Guide e The Enable Hot Desking Capability checkbox is used to enable the Hot Desking feature on the corresponding IP line e The Hot Desking Automatic Logout section is used to configure Hot Desking functionality expiration on the corresponding IP line This may be useful when someone who logged in to the public phone with the extension attached to this line forgot to log out after using it With this option enabled once the expiration time arrives the extension will automatically log out from the public phone The following options are available e Never the extension will never expire and will remain logged in to the
222. e Access List This page is used to define a list of callers that are allowed to retrieve a call parked to the corresponding call park extension If the caller is not in the Retrieve Access List for the corresponding call park extension it will not be able to pickup a call parked to this call park extension By default this table contains an Auto entry which allows any caller to pickup the call parked to this extension Attention If you modify the Retrieve Access List by adding new callers do not forget to remove the default Auto entry from the list for the new configuration to take effect The Add functional button opens an Add Entry page where a new caller can be added to the list This page consists of the following components Call Type lists the available call types e PBX local calls from QX IP PBX s extensions e SIP calls through a SIP server e PSTN calls from global telephone network e Auto used for undefined call types The destination independent on whether it is a PBX number SIP address or PSTN number will be passed through Call Routing Table Extensions Extensions Extensions Extensions Management Edit Entry Park Access List 50 ings Add Edit Delete Extension 105 109 Add Multiple Extensions Bulk Import Extensions Management Edit Entry O Go Back Retrieve Access List 50 Ings Add Edit Delete Address Auto SIP 11369 sip epygi l
223. e Gateway allows access from all stations connected to the local network to the remote VPN gateway device local QX IP PBX and remote subnet are not included The checkbox is disabled when This device lt gt Internet lt gt NAT lt gt Peer is selected from the VPN Network Topology drop down list on the first page of the IPSec Connection Wizard This device lt gt Remote Subnet allows access from the local QX IP PBX to all stations of the remote LAN local subnet and remote VPN gateway devices are not included The checkbox is disabled when This device lt gt NAT lt gt Internet lt gt Peer is selected from the VPN Network Topology drop down list on the first page of the IPSec Connection Wizard Local Subnet lt gt Remote Subnet allows access from all stations of the local network to all stations of the remote LAN VPN gateway devices are not included In this case the local and remote subnet IP addresses and subnet masks have to be entered in the corresponding text fields Local Subnet IP and Remote Subnet IP More than one of the above checkboxes may be selected to specify the desired communication relations The Stop Connection if not successful checkbox allows you to stop the IPSec connection attempts if the partner is still unreachable after the timeout period If the checkbox is not selected the system will continue to try to reach the IPSec connection partner To Delete Stop Start an IPSec Connection Select one or more
224. e Send Test Mail button to send a test e mail with the configured settings A SMS Settings Short Text Messaging The SMS Settings are used to configure the SMS parameters that will allow QX IP PBX to send the voice mail notifications or event notifications via SMS to the extension user s mobile phone Every extension user can enable voice mail notifications when a new voice mail is received and they can to define their own mobile numbers from the Voice Mail Settings or to set the certain Events notification to be delivered per SMS However for QX IP PBX to deliver SMS notifications the SMS service should be enabled and SMS settings should be configured from this page Enable SMS Service enables the SMS service on the QX IP PBX User Name and Password text fields require the authentication settings of the SMS server SMS Sender Address requires the source address for the QX IP PBX notification SMS The address defined in this field will be seen in the From field of the SMS delivered to the mobile phone Address 37455894521 nt Address 37455894521 SMS Recipient Address requires a destination mobile number for a test SMS SMS Gateway manipulation radio buttons allow to select between pre defined Clickatell SMS gateway and the custom defined SMS gateways Fig II 14 SMS Settings page e Clickatell this selection allows to use a pre defined SMS gateway Selection enables the API ID text field which indicates a Clicatell s
225. e Unconditional configures the system to send ALERT PROGRESS messages with the Progress Indicator IE With this option the system will send its own progress tone e Conditional configures the system to send ALERT PROGRESS messages with Progress Indicator IE With this option the system will send its own progress tone only if there is no early media 180 183 with SDP from the called party Incoming Called Digits Size text field indicates the number of received digits in a range from 0 to 255 required to establish a call When field has 0 value system uses either timeout defined in the T302 field or the Sending Complete Information element messages to establish a call Independent on the value in this field Sending Complete Information element and pound sign always cause the call establishment The Generate Progress tone on IP checkbox selection will generate the progress tone to IP SIP If the Send ALERT Message on Call Ringing checkbox is selected the system will send ALERT messages to callers from the PSTN PBX on call ringing If not the system will send a PROGRESS message on receiving early media from the called party if the Generate Progress Tone to PSTN PBX setting is not set to None Enable CLIR Service checkbox selection enables Calling Line Identification Restriction CLIR service which displays the incoming caller ID only in case if Presentation Indication is allowed on the remote side Otherwise if CLIR service is disabled caller I
226. e caller will be disconnected from the QX IP PBX when the Loopback Timeout expires The FXS Lines Loopback Settings page shows the only table where all FXS lines of the QX IP PBX are listed On this page the loopback diagnostics may be enabled disabled and the Loopback Timeout can be adjusted for FXS lines The FXS Lines Loopback table lists all the FXS lines on the QX IP PBX along with their loopback parameters Loopback State and Loopback Timeout E1 T1 Trunk ISDN Trunk PSTN Gateways FXS Lines Loopback Settings Er Interfaces Edit Enable Disable Loopback The Edit functional link leads to the FXS Lines Loopback Settings Edit Entry page where Loopback Timeout in seconds may be ps1 configured for one or more selected FXS line s ee Line Name Loopback State Loopback Timeout The Enable Disable Loopback functional link is used to enable disable the Loopback service on the selected FXS line s Fig IT 109 Diagnostic Loopback page Hot Desking If QX IP PBX has limited number of analogue and IP phones connected and much more users wishing to make and receive calls through the QX IP PBX some of the connected phones can be announced as public Public phones have no static owners they are just connected to the analogue or IP lines Each user that accesses the public phone should first login with the previously created virtual extension and the corresponding password in order to make the phone assigned to the certain extension From
227. e end will be applied as a prefix This tag can be used in combination with other digits at the beginning or at the end as well as with wildcards e lt dialednum range gt used to apply the complete or a part of dialed number the number dialed by the caller to place a call as a prefix For example lt dialednum 1 3 gt indicates that the first 3 digits of the dialed number will be considered as a prefix lt dialednum 3 end gt indicates that the dialed number from its 3 digit and up to the end will be applied as a prefix This tag can be used in combination with other digits at the beginning or at the end as well as with wildcards The syntax aaa bbb in the Prefix field allows for two stage dialing The aaa and bbb are the numbers to call bbb can also be a series of digits to inject a comma indicates a delay of one second The syntax can be applied to include more call destination numbers separated by time intervals A two stage dialing allows successive numbers to be dialed one after another with a delay in between For example 11 11018 will call 11 wait until the call is established wait for three seconds and then dial 11018 The capability of automatically dialing successive numbers allows the caller to bypass the IVR system on the call path and establish a direct call The two stage dialing is available for PBX and ISDN destination types Suffix requires entering the symbols letters digits and any characters supported in the SIP user
228. e particular ISDN trunk from CO besides ISDN Wizard the QX In case of connection to the PBX Network interface type is selected on QX choose this option if only the PBX is connected to the ISDN trunk from the QX ISDN Gateway no se aaa other ISDN devices are connected to the particular ISDN trunk from the QX ISDN Gateway od In both cases with this selection QX sets the TEI to manually PIP pon To Po mode assigning the default value of 0 If needed that value can be changed later in the Advanced Settings page of ISDN Wizard e PTMP Point to Multi Point In case of connection to the CO User interface type is selected on the QX choose this option if there can be other devices connected to the same ISDN trunk from CO except the QX IP PBX In case of connection to PBX Network interface type is selected on the QX choose this option if there can be other devices connected to the same ISDN trunk from QX ISDN Fig II 127 ISDN Wizard ISDN Settings page Gateway except for the PBX In both cases with this selection QX sets the TEI to automatic mode Please Note Consult with your CO operator or network administrator before configuring the ISDN connection type PR Interfaces ISDN Settings Y PTMP Point To Multi Point The ISDN Wizard Page 2 content is dependent on the connection type selected on the previous page of ISDN Wizard The next page is ISDN Wizard MSN Settings page which is used to turn on the MSN config
229. e pattern are 2115 2125 2135 2145 2155 2a5 2b5 2c5 2d5 s Use an exclamation point to exclude a character or a string from a set Example The pattern is 2 11 15 a d 14 c 5 Numbers matching the pattern are 2115 2125 2135 2155 2a5 2b5 2d5 Please Note You can use the wildcard within the braces but not Thus 12 104 15 36 is a valid pattern whereas 15 36 is not Please Note The symbol cannot be used to exclude a range of symbols For example 2 15 60 23 32 or 2 15 60 23 32 are not valid patterns To valid pattern will be to 2 15 22 33 60 The same as above with the exception that character ranges can include single digit character elements only Example The pattern is 2 1 5 a c 5 Numbers matching the pattern are 215 225 235 245 255 2a5 2b5 2c5 Precedes a control symbol and to indicate that it is used as an ordinary character not a wildcard Example The pattern is 1 1 3 Numbers matching the pattern are 1 1 1 2 1 3 Please Note Patterns cannot be prefixed with the symbol The system considers the patterns starting with as feature codes and does not parse them through the Call Routing table Used to indicate the full SIP address example 20233 sip epygi com This pattern is mainly used to call back users registered on the SIP server different from the one where the called party is registered Please Note Patterns containing symbol will not be parsed among tho
230. e servers Time is used to set the local time hour minute Date is used to set the date month day year Enable Simple Network Time Protocol Server enables the SNTP Simple Network Time Protocol server on QX IP PBX thus QX IP PBX becomes the timeserver for its LAN System LAN Internet WAN Email SMTP Short Text Messaging SMS Enable Simple Network Time Protocol Client enables the O Setup l l SNTP client on the QX IP PBX thus QX IP PBX becomes a client toe A SCMNGE to an external timeserver A checkbox disables Date and Time drop down lists and enables the following parameters The SNTP Servers table lists all defined NTP Servers Y Enable Simple Network Time Protocol Server Enable Simple Network Time Protocol Client The Add functional button opens an Add SNTP Server page where a new NTP server can be defined This page offers the LAO NTP Server radio buttons that are used to choose between a ae manual and a predefined NTP server a e Manual requires the NTP servers FQDN Full AA Qualified Domain Name or its IP address e Predefined is used to select the NTP server s host address from the drop down list where the most common NTP servers are listed Fig II 11 Date and Time Settings page QX50 QX200 QX2000 SW Version 6 0 x 15 e C Pyg l QX50 QXK200 QX2000 Manual Il Administrator s Guide The Move Up and Move Down functional buttons are used to sort NTP servers in the order they ne
231. e specified SMTP server allows using both secure and unsecure connections then this selection forces to establish the secure connection Fig II 13 System Mail Settings page Enable SMTP Authentication must be selected if the specified SMTP server requires authentication In this case authentication User Name and User Password configured on the SMTP server should be defined in the corresponding text fields Attention The following symbols are not allowed for the Password field WN TN With the button Send test mail a test mail can be sent to the defined email address to verify the settings This button will be enabled if correct values have been submitted and saved on this page To configure the System Mail Enable the system mail sending by the Enable checkbox selection Update or set the SMTP host in the SMTP Host text field Update or set the e mail sender address in the Mail Sender Address text field Update or set the e mail address in the Mail Recipient Address text field Enable the secure connection TLS if the specified SMTP server requires secure connection eer QX50 QX200 QX2000 SW Version 6 0 x 16 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide Enable SMTP Authentication if it is required on the server Insert into the corresponding text fields an authentication User Name and User Password defined by your SMTP server Press the Save button to submit these settings Use th
232. e time pere a Subject lt subject gt remained until the conference will start etc Desciton lt ideeciptonl gt Participant password lt password gt All these tokens can be inserted by using the links on the right side of cai the page Please Note Changing the body of the token will disable the token functionality and will be implied as a simple text The Restore Defaults button is used to restore the default mail Fig Il 77 Conferences Mail Default Settings page templates Using this button all user defined mail templates will be lost Universal Extension Recordings The Universal Extension Recordings are to be defined by the QX IP PBX administrator and will be present instead of the default voice messages for all extensions on the QX IP PBX They will be used when no custom messages have been uploaded or recorded The following system messages can be uploaded from this page e Hold Music played to the held user The Edit link is used to select the way custom hold music will be provided e Voice Mail Regular Greeting played when a caller reaches the extension s voice mailbox Voice Mail Out of Office Greeting played when a caller reaches the extension s voice mailbox if the Out of office greeting is enabled Incoming call blocking played when a blocked user calls the extension e Outgoing call blocking played when the extension dials a blocked destination e Call Queue Welcome Message playe
233. e type of the option value It may be an IP address a boolean or integer value etc The Option Value text field is used to insert the value of an option Depending on the selected Option Value Type this field should have the corresponding value Warning messages will prevent saving if the value inserted in this field does not correspond to the requirements of the Option Value Type If an array should be inserted here the values should be separated with a comma DHCP Leases The DHCP Leases page includes a list of the leased host addresses that are part of the QX IP PBX s LAN For these hosts QX IP PBX acts as a server supplying them with a unique IP address It displays a read only table describing all the leased IP hosts and their parameters The table contains the following columns IP address host IP address assigned by QX IP PBX PPP PPTP SNMP VLAN VPN MAC address host MAC address provided by the host itself g DHCP Leases Lease Start date and time when the leased IP address has been IP Address MAC address Lease start Lease end 172 30 0 253 00 04 f2 24 c5 11 Tue Aug 05 15 36 26 2014 Tue Aug 12 15 36 26 2014 a ctivate d 172 30 0 254 00 15 65 2e 95 ad Tue Aug 05 15 35 47 2014 Tue Aug 12 15 35 47 2014 Lease End date and time when the leased IP address has been or will be deactivated Binding State indicates the state of the DHCP lease Fig IT 210 DHCP Leases page for LAN interface Hostname hostname
234. e will be sorted by the selected column Upon sorting ascending descending arrows will be displayed close to the column heading QX50 QX200 QX2000 SW Version 6 0 x 150 QX50 0X200 0X2000 Manual II Administrator s Guide e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide The Number of Records displays the current number of statistics entries in the table For successful calls Total Duration Maximum Duration Average Duration and Minimum Duration Sse nicol statistics are displayed on top of the table a a Call History Successful Calls The Call History Successful Calls Missed Calls and Unsuccessful Outgoing Calls pages consist of the general a a information on successful missed and unsuccessful calls search onimi fields and the calls table The Filter button performs searching a sus within the statistics tables The search may be done with several criteria at the same time From05 Aug 2014 Calling Phone Called Phone The following search criteria are available To 06 Aug 2014 dd Month yyyy hh mm ss e The text fields From and To are used for the search by Call Start Time The data must be entered in the format dd mm Call Start Time A Call Duration Calling Phone Called Phone Details 06 Aug 2014 12 20 19 sec PSTNGW 192 168 74 126 1 ZavZak lt 164 gt 143 Codec PCMU Quality 1 excellent yyyy hh mm ss The time criteria are optional if it is not PS
235. each extension such as SIP username or IP Line configuration should be specified by the Epygi s Bulk User Extensions Importer configuration tool and imported later from the appropriate configuration file These settings are marked with variable sign in the extensions configuration page see User Extension Settings The Epygi Bulk User Extensions Importer configuration tool is a MS Excel based form which allows a configuration file to be created based on the configured templates for Add Modify type of files When your configuration file is ready select the Extension Import Settings tab to upload the Bulk User Extensions Importer configuration file to the QX IP PBX Browse opens the file selection window to browse for a new user bulk extension configuration file The Override Existing Extension indicates whether the settings of Extension Template Management the imported file should change the settings of existing extensions if the imported file is of the Add type It can also contain the settings for extensions which already exist on the QX IP PBX When the Override Import User Extension Bute Choose File No e chosen Existing Extension is unchecked and the uploaded Add type CSV Over Existing Extension configuration file contains extensions which already exist on the QX IP PBX an error will appear and the conflicting extensions will be highlighted If the uploaded file is of the Add type and the intent is to modify existing extensions th
236. eceived during this timeout QX IP PBX will perform a request dependent default action For example if the call controlling application is configured to handle incoming calls on the QX IP PBX Once the incoming call occurs QX IP PBX is trying to transfer the call to the call controlling application If the call controlling application does not response within the mentioned timeout QX IP PBX will answer the call or perform an action configured for unanswered incoming calls This setting is dependent on the network conditions therefore consult with your network administrator before changing the default value The read only Feature Key text field indicates whether the feature key for the 3PCC Support is installed on the system The system will not accept connections from 3PCC applications if no key is found The 3PCC support is an optional feature and can be activated with a feature key from the Feature Keys page The read only WAN Port text field indicates whether there is a filtering rule specified for the Call Control Access If a third party call control application connects to the QX IP PBX from the WAN interface a filtering rule for the corresponding host should be created on the Call Control Access page to allow the application a remote access Creating a filtering rule is not required if the firewall is not setup on the QX IP PBX The field shows Opened if there is at least one enabled filtering rule for the Call Control Access RADIUS Client Sett
237. ected a group of radio buttons is activated to select the storage for the recorded conference audio files e Use Internal Storage switches the location used to store the recorded conference audio files to the system internal memory Max Recording Time requires the maximum duration in minutes of one recording to be done If the conference recording has been paused and resumed again the Max Recording Time value will indicate the actual recorded time Leave this field empty not to limit the duration of the conference recording Conference Progress Conference Recording Settings Conference ID 888 Y Enable Recording Use Internal Storage QX50 0X200 0X2000 Manual II Administrator s Guide Recorded Conferences Conference Settings General Customization Participants Schedule Send Notification Mail Max Recording Time min 15 Use External Storage Recording SIP Address e Use External Storage switches the location used to store Recording Indication the recorded conference audio files to an external destination which can be any device or application that has audio recording capabilities The SIP Address of the remote destination where the recorded conference will be stored is required to be defined for this selection Optionally the SIP address of a user can be inserted here In this case the conference will be recorded to the private mailbox of the user or will be directly played to him if he answers the incoming
238. ection enables the following information to be defined The Extension Length drop down list requires you to choose the length of the extensions on the QX IP PBX This number will apply to all existing extensions on the QX IP PBX as well as to any newly created extensions The length of the extension can be 3 4 or 5 The Extension Prefix text field is used to define a prefix with which all existing extensions on the QX IP PBX as well as to any newly created extensions should start The prefix cannot start with the digits O or 9 otherwise an error message appears Please Note By saving the settings on the Choose Extensions Length page all existing extensions will lose the custom voice messages and voice mails in the voice mailbox The device will be rebooted You will not be automatically redirected to the login page so you need to access it manually again when reboot ends After the reboot the Choose Extensions Length page will disappear and the Extensions Management page will be displayed The Choose Extensions Length page will not appear again unless the default configuration settings are restored on the device Two types of user extensions active and inactive can be created on the QX IP PBX Active extensions are those that are attached to a line can place and receive calls and use available telephony services Inactive extensions are those that are not attached to the line They can use some available telephony services but they cannot place an
239. ection of the loudest participant in the conference the current speaker and switching the video on all of the video conferencing phones in automatic mode to the video from that participant Initially when the user joins a conference with Automatic Speaker Detection checkbox enabled his video phone works in automatic mode Dialing 900 or 900 feature codes will switch the phone to manual mode displaying the video of the next or previous participant correspondingly When the phone is in manual mode it will not switch automatically to display the loudest participant but it will show the video of the same participant until next time when 900 or 900 is being pressed Entering the 00 feature code will switch the phone back to automatic mode For making the video source switching decision in automatic mode the video conferencing uses the values of the following parameters e Calculate the voice energy for the last sec e Calculate the voice energy every sec e Switch to new Video Source if energy difference is more than dB For example if the values of the parameters are 3 1 and 6 default values correspondingly the Conference Server will calculate every one second the average voice energy of each participant during the last three seconds Then the largest calculated value will be compared to the average voice energy of the participant providing currently the video for all phones in automatic mode If the difference between energies is
240. ed by a Service an Action a Restriction to certain IP address es or IP groups and if adding a rule for Incoming Traffic Port Forwarding the destination IP address for Forwarding The page to add arule for Incoming Traffic Port Forwarding offers the following input options Service includes a list of possible services to be configured All custom services also will be displayed in this list Action includes possible actions to setup the rule Forward to IP requires the destination IP address where traffic should be transferred to if it comes from the restricted host The IP address defined in this field will be ignored for blocked action of the Incoming Traffic Port Forwarding rule QX50 QX200 QX2000 SW Version 6 0 x 122 QX50 0X200 0X2000 Manual II Administrator s Guide Please Note It is not allowed to forward incoming packets when the NAT service is disabled on the QX IP PBX Port Translation text field is available for Allowed action only and optionally requires the port number that will stand instead of the original port nu mb er wh en inco ming packet is being omi ri eccess Call Control Access SIP Access Blocked IPs Allowed IPs forwarded If this field is left empty the original port number will be used when forwarding the packet Restriction radio buttons Add Filtering Rules Incoming Traffic Port Forwarding O Go Back In order to prevent malconfiguration only one rule per service is allowed You ca
241. ed due to selected Failure Reasons the call routing table will be parsed for the next matching pattern and if found the call will be routed to the specified destination Busy available for PBX SIP SIP Tunnel and IP PSTN destination types and indicates cases when the dialed destination is busy Wrong Number available for PBX SIP SIP Tunnel and IP PSTN destination types and indicates cases when the dialed number is wrong Network Failure available for SIP SIP Tunnel and IP PSTN destination types and indicates cases when system overload network failure or timeout expiration occurred System Failure available for SIP SIP Tunnel and IP PSTN destination types and indicates cases indicated in Network Failure and Other fail reasons Cannot Establish Connection available for FXO ISDN and E1 T1 destination types and indicates cases when connection cannot be established Other available for SIP SIP Tunnel and IP PSTN destination types and indicates cases when authorization negotiation not supported or request rejected or other unknown errors occur e Any stands for all failure reasons mentioned in the Failover Reason s group The Custom Profile text field is present if the PBX Voicemail destination type has been selected on the first page of the Call Routing Wizard This field requires the Voice Mail Profile name to activate the custom voice mail settings see Manual III Extension User s Guide on the extension when the cor
242. ed in to the public phone with this extension forgot to log out after using it With this option enabled once the expiration time arrives the extension will automatically log out from the public phone The following options are available e Never the extension will never expire and will remain logged in to the public phone e After the defined period of time requires the period after which the extension will automatically log out from the public phone e Atthe certain moment requires the moment hour and minute when the extension will automatically log out from the public phone 5 Call Queue Settings This group is used to configure the Call Queue service that allows multiple incoming calls to be kept in the queue when being on the line and enables the calls to be answered in the order they have been received This feature can be also used within Receptionist Management see below for more details The Enable checkbox activates the Call Queue functionality on the extension The Call Queue Size text field requires the length of the call queue This is the maximum number of calls that will be accepted into the queue and kept on hold while the extension user is on a call If a E maximum number of calls are already held in the call queue the Call Queue Settings 103 next incoming call will be routed to the extension s Voice Mail if Esos enabled or will be disconnected Redirection on Timeout Please Note By configuring Call Qu
243. ed to be accessed If the Overvie Security Lice s NTP server in the first position of the SNTP Servers table does System LAN Internet WAN MCCAIN Emai ism Short Text Messaging sms not answer NTP server in the next position will try to be T Add SNTP Server reached NTP server Please Note You can add another NTP server to the list if the defined NTP servers are not functional for example QX IP PBX s date time is not being updated automatically ie a Polling Interval indicates the time interval for the periodical synchronization between the timeserver and QX IP PBX It counts in hours Fig II 12 Add SNTP Server page Attention Date and Time Settings will be reset if QX IP PBX has lost power System Mail Settings Email SMTP The Email SMTP page allows you to send warnings automatically about the board status or problems to the administrator System events that require email notification are selected on the Events page System mail must be enabled and the SMTP server needs to be configured for voice message transmission to the extension user s mailing account QX IP PBX may automatically generate emails to the administrator e If events specified in the Events list occur e Ifvoice mails are set from the Voice Mail Settings see Manual III Extension User s Guide to be sent as e mail With the Enable checkbox system mail sending and voice messages transmission to the extension user s mailbox could be enabled
244. ed to restrict the delivery of the SIP message if any of the selected headers cannot be hidden or replaced depending on the configuration of the SIP server before being sent to the destination For E1 T1 destination type the Port ID drop down list contains available E1 T1 trunks The available Timeslots TS should be selected on the next page For FXO destination types a group of Port ID radio buttons allows you to select whether a specific or any available FXO line will be used to route the call The AnyOAny selection indicates that the call will be routed through the first available FXO line The Specific Ports selection is used to select a group of routing settings for shared FXO lines Each Shared Gateway Ports radio buttons group is dedicated to one shared FXO device and is used to configure shared FXO lines usage when using the corresponding routing entry None selection means no shared FXO lines will be used for the call routing of the specific routing rule Any Port ipaddress where ipaddress is the IP address of the FXO gateway that shares its FXO lines selection means the call will be routed through the first available shared FXO line FXO ipaddress port checkboxes are used to select those which shared FXO ports will be used for the corresponding rule routing In case if multiple shared FXO ports are selected here the first available port will be used The FXO Lines Load Balancing drop down list is used to enable load balancing mechanism
245. el Participant Video Dial oa Number Type Allowed Out 99 Participant Confirmation Nested Mail Add Indication Type Conference a sa Duplication John 11369 sip epygi loc Speaker Yes Yes Yes Yes Star Yes Yes Smith John Smith gmail com Alice 5986744 Speaker Yes Yes Star No Yes alice dawson epygi com Dawson Fig II 300 Conference Settings Participants page Please Note By default no participant is able to make video calls Administrator should set one of the following checkboxes to enable the video capability of the participant e Allow Video checkbox from the Participants Add Entry GUI page see Fig II 298 e New Participant Can Make Video Call checkbox from the New Participants Configuration GUI page see Fig II 303 e Allow Video checkbox from the Handset Added Participants Confi Add opens an Add Entry page where new participants can be added to the conference The following parameters are needed to configure participant settings Participant Name requires optional information first name last name nickname etc about the participant SIP Address Tel number requires the contact phone number SIP address or Routing Number of the participant This number automatically will be dialed by the system when the participant is configured to be a Dial Out see below or when a corresponding Conference Code is used see Conference Codes The participant s SIP address should be a combination of userna
246. elled the language pack update procedure on the following steps The next page displayed will show verification of the language pack being uploaded and asks for confirmation to overwrite the existing custom language pack if applicable After final confirmation the system will upload the selected custom Language Pack and it will reboot Update Languages for IP Phones The Update Languages for IP Phones page is used to upload a custom language pack to the IP phone This page only contains those IP phones that support custom language pack uploading from the QX IP PBX To upload the custom language pack go to your IP phone related page and Choose File the custom language pack file Save the changes to upload the custom language pack to the IP phone Attention Pressing the Save button will stop some vital processes on the IP Phone therefore you will need to reboot your phone manually even if you have cancelled the language pack update procedure on the following steps setup Update languages for IP phones Aastra Snom Grandstream Aastra Language Pack file to substitute txt Choose File No file chosen Save Polycom Fig I1 21 Update Languages for IP Phones page The next page displayed will show verification of the language pack being uploaded and asks for confirmation to overwrite the existing custom language pack if applicable After final confirmation QX IP PBX will upload the selected custom Language Pack
247. elpful to support as many codecs as possible In this case all codecs that the system offers should be enabled in the Codecs table On the other hand some codecs require quite a high transfer rate of up to 64 kBit s If you definitely do not want to use these codecs make sure they are disabled in the Codecs table QX50 QX200 QX2000 SW Version 6 0 x 49 QX50 0X200 0X2000 Manual II Administrator s Guide The Codecs table lists the voice and video codecs supported by the QX IP PBX Each table entry is assigned a checkbox that is used to manipulate the entry for example to disable to move it up or down Se etc Add Extension Add Mutiple Extensions Bulk Import The table entries in bold type indicate codecs enabled for the selected E extensos Extension 103 Codecs extension attendant conference The enabled codecs participate in Seema codec negotiation at the call setup The order of the enabled codecs is very important Each codec in the table has a higher priority than the codecs below it and a lower priority than the codecs above it A era raae ona ine TT codec placed at the top of the table is used as the preferred codec 6725 16 ADPCM spec h G 726 24 ADPCM speech coding at 24 kbit s rate Disabled h h Enable Disable Move Up Move Down Make preferred Audio Codecs State G 711u PCM audio coding standard 8 kHz sample rate 8 bits 64 kbit s data rate preferred Enabled G 711a PCM audio coding standard 8 kHz sample
248. em will route the call to the destination The Incoming Interdigit Service page lists a table with existing E1 T1 dial plan entries and allows you to manage them By default the table on the Incoming Interdigit Service page lists Fig II 124 Incoming Interdigit Service page the locale specific selected from the System Configuration Wizard E1 T1 dial plan settings For some countries this table may however be empty E1 T1 Trunk ISDN Trunk PSTN Gateways 012 3 QX50 QX200 QX2000 SW Version 6 0 x 84 QX50 0X200 0X2000 Manual II Administrator s Guide Add functional button leads to the Add Entry page where a new SA PVE E1 T1 dial plan entry can be configured The Add Entry page consists of the following fields The Incoming DNIS Prefix text field requires the prefix of the a ene incoming dialed number are used to define a range or ace cae a quantity of prefixes For example 2 5 9 means that the prefix of the dialed number may be 25 26 27 28 or 29 3 4 7 0 means that the prefix of the dialed number may be 34 37 or 30 Only one range of prefixes can be defined in the Incoming DNIS Prefix text field Trunk 1 192 168 74 127 5060 E1 Signaling Type CCS B Channels Edit Entry Timeslot 2 Fig IT 125 Incoming Interdigit Service Add Entry page The Incoming DNIS Size text field requires the total length of the dialed number including the prefix digits The number defined in this field should
249. en a file has been previously uploaded and is used to remove the uploaded information file Participants This page allows to configure participants of the conference as well as to adjust settings of the participants dialed out during the conference or independently connected to the conference The New Participants Configuration moves to the page where the settings of participants independently dialed in to the conference can be configured Once the new participant connects the conference he will automatically appear in the Conference Recorded Conferences Conference Settings w Conference General Recording Customization Schedule Send Notification Mail Progress table on this page and remain there unless disconnected from the conference The Handset Added Participants Configuration moves to the page where the settings of participants dialed out from the handset by the moderator during the active conference can be configured Once a handset added participant connects the conference he will automatically be added to the Conference Progress table on this page and remain there unless the conference is terminated The table on this page lists all preconfigured participants allows to add new participants and to modify the settings of the exiting ones Participants Conference ID 888 New Participants Configuration Handset Added Participants Configuration Registered Participants Add Edit Delete Name SIP Address T
250. en the Override Existing Extension should be enabled otherwise the file must be of the Modify type Extensions Extension Template Management Extension Import Settings Fig II 73 Extension Import Settings page When you upload the Bulk User Extensions Importer configuration file the system will check the entire file before applying the uploaded configurations If there are some incorrectly configured settings in the file the system will return a table with all uploaded configurations and highlight the parameters which have an error If the uploaded file passed and did not give any error message the system will start to Add Modify all specified extensions As a result the system will Add Modify the specified extensions In addition for any settings that need to be updated in the IP phone e g Display Name a new IP phone configuration file will be created and ready for sending to the phone the next time it is rebooted Conferences Please Note The Conference Server and the Video Conferencing are optional features and can be activated with a feature key from the Feature Keys page Conference users with video will be able to see the current speaker and either manually or automatically switch between participants This gives the user power over which person they get to view or allows the video conference server to rotate the video feed to the person currently speaking After activating Video Conferencing feature from the Setup
251. en this option is enabled or the analogue FXS lines are attached to the corresponding extension the caller gets dial tone Otherwise there will be no dial tone for FXS lines The Hot Desking Automatic Logout section is used to configure Hot Desking functionality expiration on the corresponding FXS line This may be useful when someone who logged in to the public phone with the extension attached to this line forgot to log out after using it With this option enabled once the expiration time arrives the extension will automatically log out from the public phone QX50 QX200 QX2000 SW Version 6 0 x 74 QX50 0X200 0X2000 Manual II Administrator s Guide The following options are available e Never the extension will never expire and will remain logged in to the public phone e After the defined period of time requires the period after which the extension will automatically log out from the public phone e Atthe certain moment requires the moment hour and minute when the extension will automatically log out from the public phone Information on the Caller ID system Caller ID is a service identifying the caller when performing a call or sending a voice mail and notifying the called party about the identity of the caller The Caller ID service is available only for phones with a display to show that information Two types of Caller ID notification are available on QX IP PBX FSK and DTMF FSK Standard The FSK standard supports
252. en this participant joins or leaves the conference e Nested Conference must be selected if the participant is a Conference itself and enables the correct behavior of conference termination e Allow Duplicated Participation checkbox allows multiple participants with the selected Caller ID calling address to join the corresponding conference This is applicable when different participants are using the same shared number to place a call Dial Out functional button is used call one or more inactive participant s inviting them to join the conference Delete removes the selected participants from the conference Set Speaker functional button is used to grant selected participants a speaker s permissions A participant with speaker permissions is able to speak to the conference Set Listener functional button is used to grant selected participants a listener s permissions A participant with listener permissions is not able to speak to the conference and is only a listener Lecture Mode functional button is used to grant selected participants a lecturer s permissions Both listener and speaker participants can get lecturer permissions Enabling lecture mode for a participant will allow him to speak to the conference and will mute all other participants of the conference Please Note Only one participant can act ina lecture mode at the same time Recorded Conferences Conference recording service allows you to record conferences and save them on the
253. ending It offers the following components The Recording Codec drop down list contains the existing codecs for voice mail compression Changing the Voice Mail recording codec will directly affect the allocated memory size for users Email Subject for voice field is used to when user enables Send new voice messages via e mail option from his personal Voice Mail Settings In this field you may define a flexible subject for all emails sent from the QX IP PBX and carrying the voice mails QX50 QX200 QX2000 SW Version 6 0 x 113 QX50 0X200 0X2000 Manual II Administrator s Guide Besides using static text in the subject line you may want to use the predefined tags to combine the needed subject RTP Streaming Channels Gain Control 3PCC Radius Client Timeout Call Quality Notification e Hostname the hostname of the QX IP PBX e Displayname the caller s display name This value is not displayed for PSTN callers sae Se e Username the caller s SIP username For PBX caller E mail Subject for voice this is the caller s PBX number for PSTN callers this is A Tom SYM AEPNAME SAM UARAN the caller s PSTN number Voice Mail Common Settings Recording Codec Insert Hostname Displayname Username Fullname Duration Date FAX to E mail format e Full name the caller s full SIP address SIP username TF Tag nage Fe Fora and the SIP server For PBX caller this is the caller s PBX number for PSTN callers this is the caller s PST
254. entication type Use the Expiration Date and Time checkbox to enable the expiration timeout Select the Expiration Date and Time from the corresponding drop down lists Press Save to apply these settings Allowed Characters and Wildcards The following is the set of characters and wildcards allowed in the Pattern and Source Number Pattern text fields ofthe Call Routing Wizard Characters 0 9 A Z a z _ amp Q Please Note The symbols and should be prefixed with a slash if they are used as ordinary characters otherwise the system will interpret them as wildcards Please Note The symbols and are used to define a range of characters and cannot be used as ordinary characters Wildcards i Any number of any characters Any single character QX50 QX200 QX2000 SW Version 6 0 x 101 QX50 0X200 0X2000 Manual II Administrator s Guide A character or a string from the specified set of characters and strings The following control symbols are used to specify a set e Use a comma to separate the elements of a set Please Note No spaces are allowed within braces Example The pattern is 9 1 3 11 a Numbers matching the pattern are 91 93 911 9a e Use a minus sign to specify a range of characters Each successive element of the range is obtained by increasing the previous element the element code by one Example The pattern is 2 11 15 a d 5 Numbers matching th
255. er permission to the corresponding participant Participant with the speaker permissions are able to speak to the conference The following functional buttons are present on Conference Progress page to manipulate with the participants in the conference Add functional button opens the Add Participant page where a new participant can be manually added to the conference The Conference Progress Add Participant page consists of the following components QX50 QX200 QX2000 SW Version 6 0 x 185 QX50 0X200 0X2000 Manual II Administrator s Guide Participant Name requires optional information first name last name nickname etc about the participant e SIP Address Tel number requires the contact phone number Conference Progress Recorded Conferences Conference Settings SIP address or Routing Number of the participant This 4 Conference General Recording Customization Schedule Send Notification Mail number automatically will be dialed by the system when the participant is configured to be a Dial Out see below or when a corresponding Conference Code is used see Conference Conference ID 888 Codes l Conference Progress Add Participant Participant Name Adam Scott The participants SIP address should be a combination of username hostaddress port where hostaddress can be an IP address for example 192 168 90 10 or a host name e g sip epygi com The port number is optional for the SIP address If no port is s
256. er system has problems with one of the services listed below More information about these services can be found at http www protocols com pbook ppp htm The Advanced PPP Settings page offers the following group of checkboxes Enable automatic PPP restart at checkbox is used to select the time when the PPP connection will automatically be restarted The checkbox selection enables LCP echo failures text field that indicates the number of the LCP echo failure packets received before the PPP connection will be considered as dead and will be restarted QX50 QX200 QX2000 SW Version 6 0 x 134 QX50 0X200 0X2000 Manual II Administrator s Guide Disable CCP Compression Control Protocol negotiation this option should only be selected if the peer system is not working properly For example if it is not accepting the requests from the PPPD Point to Point Daemon for CCP negotiation Advanced PPP Settings Disable magic number negotiation with this option PPPD cannot detect a looped back line This option should only be selected if the peer is not working properly Enable automatic PPP restart at Disable protocol field compression negotiation in both the receive and the transmit direction with this option no isable CCP Compression Control Protocol negotiation protocol field compression will take place sabe magic number negotiation isable protocol field compression negotiation in both the receive and
257. ere a new master device parameters needs to be provided The Add Entry page consists of the following components The Enable Registration checkbox selection is used to enable the registration to the corresponding master device The Tunnel Name text field requires the SIP tunnel name for the corresponding connection System suggests you to start the SIP tunnel name with the SIP_Tunnel_ words according to the automatic prefix used for the SIP tunnels on the QX IP PBX ene however this is not mandatory SP Tunnel 7415 Call Routing Table Call Routing Local AAA Table Global Speed Dial SIP Tunnel Class of Service SIP Tunnel Settings Tunnels to Master Devices Add Entry The User Name text field requires the authentication user name l The field in front of this text field displays the default non editable prefix for SIP tunnels SIPTunnel_ 192 168 74 12 5060 The Password text field requires the authentication password Please Note The User Name and Password should match both on master and slave QX IP PBXs for the successful SIP tunnel establishment Fig ll 153 SIP Tunnel Settings Tunnels to Master Devices Add Entry page The Master device IP text field requires the IP address of the master device The Master device port text field requires the SIP port number of the master device The Registration State field displays information whether the slave device is registered on the master or not The Registration Da
258. ered checkbox is selected incoming calls towards the corresponding extension on the QX IP PBX will be forwarded to the remote SIP phone only if it is registered Otherwise when the remote SIP phone is unregistered incoming calls will be routed to the line extension it is attached to When this checkbox is not selected all incoming calls SIP Remote Extension Settings 103 will be routed to the remote SIP phone only if it is registered Otherwise if the remote SIP phone is unregistered calls will be forwarded to the extension s voice mailbox Extensions Management Edit Entry Extensions Y Enable Hot Desking Capability Generate Password Hot Desking Automatic Logout The Symmetric RTP checkbox should be selected when the remote extension is located behind the symmetrical NAT E Enable RTP Pro W Fallback To Local Extension When Not Registered Symmetric RTP Go To Line Settings Go To Codec Settings Fig II 35 Extensions Management Edit Entry Remote Settings page The Show Hot Desking Settings and Hide Hot Desking Settings links are correspondingly used to show or hide the Hot Desking settings on this page The Enable Hot Desking Capability checkbox is used to enable the Hot Desking feature on the corresponding remote extension The Hot Desking Automatic Logout section is used to configure Hot Desking functionality expiration on the corresponding extension This may be useful when someone who logg
259. erminate available for an active conference only and used to terminate the active conference Lock available for an active conference only and used to lock the conference When a conference is locked no users can connect to it Unlock available for an active conference only and used to unlock the conference Start Resume available for an active conference only and used to start the recording of the conference or to resume the recording if it was paused Pause available for an active conference only and used to pause the recording of the conference Stop available for an active conference only and used to stop the recording of the conference Please Note Pausing and Resuming the conference recording can be used to edit the recorded conference audio file When pause resume operations are used conference is recorded in a single file leaving out the conversation during which conference recording was paused When using stop start operations new files are created each time conference recording is started All recorded conferences are listed in the Recorded Conferences page only after conference recording termination In case of pause resume the recorded file is not terminated In case of stop start recording starts in new file The table of participants on this page lists all preconfigured participants independent of the conference status as well as new participants joined the conference if still connected to the conference and those partici
260. ernate IP network of the LAN interface which stays unchanged when the device switches its mode from master to backup or vice versa The configuration and voice data synchronization daemon uses this IP address to communicate with the second QX IP PBX Redundant Device Virtual IP Address text fields require an alternate IP address of the LAN interface of the second QX IP PBX Synchronization Interval text field requires the period of time in seconds between two consecutive configuration and voice data synchronizations from master to backup device Backup Device GUI Access Port text field available only for QX50 0X200 is present on the master device only and requires the port used for accessing the GUI of the backup device through master Swap Master Device button is used for manual swapping of functionality of master and backup devices This action will result in rebooting the current master After rebooting the current master device will start running in a backup mode Switching the backup to master starts all applications on QX IP PBX and causes all IP phones to reboot The swapping takes around 1 minute however another 1 3 minutes are required in order to reboot all the IP phones connected to redundant system If backup device before swapping was in passive mode then after swapping the master will start running as backup in passive mode otherwise if it was in active mode then master will start running as backup in active mode Download system lo
261. es match a user s dial string the route with the lower metric will be chosen The Description text field requires an optional description of the routing pattern The Filter on Source Modify Caller ID checkbox selection allows limiting the functionality of the current route to be used by the defined caller s only Ifthis checkbox is enabled source caller information Source Number Pattern Source Type Source Host etc will be required later in the Call Routing Wizard This option is enabled by default The Set Date Time Period s checkbox selection allows you to define a validity period s for current routing patterns to take place and to define pattern date time rules When this checkbox is enabled the Call Routing Wizard Date Time Rules Add Entry page will be displayed The Set Overall Calling Time Limit checkbox selection allows a total call duration for all calls to be configured over a specific time frame for each Call Routing entry Once the total duration has been reached the entry can be disabled allowing calls to use the next available route QX50 QX200 QX2000 SW Version 6 0 x 94 QX50 0X200 0X2000 Manual II Administrator s Guide QX50 0X200 0X2000 Manual II Administrator s Guide If this checkbox is not selected in the Call Routing Wizard first page the overall call duration will be unlimited When this checkbox is selected Call Routing Wizard Routing Overall Call Limitation Settings page will be displayed
262. es table displays all established IP static routes with their parameters Target State for the state of the route enabled or disabled Actual State for the state of the route connection up down or erroneous Route To for the subnet where the incoming packets should be routed to and Via IP Address for the router IP address where incoming packets should be routed through Add opens the Add IP Static Route page where a new static route can be established Enable Disable is used to activate and deactivate a selected route s At least one route should be selected in order to use these functions otherwise the following error message will appear No record s selected The Add IP Static Route page offers the following components Route To requires the IP address and subnet mask for the destination the IP packet should be forwarded to Via IP Address requires the IP address of the subsequent router for IP packet forwarding to the specified destination Attention The rule with the longest subnet smallest IP range will take effect when having two or more IP Static routing rules with the coinciding subnets IP Policy Routes Network Over IP Routing DHCP DNS PPP PPTP SNMP IP Policy Routes PPTP L2TP Routes IP Static Routes Enable Disable Add Edit Delete Target State Actual State Route to Via IP Address Fig II 200 IP Static Routes table Over IP Routing DHCP DNS PPP PPTP IP Policy Routes PPTP L2TP Routes Add I
263. es the maximum number of new users allowed to connect to the conference Leave this field empty to allow unlimited number of new users W conference GH Schedule Send Notation Mal connecting the conference In one conference the maximum New Participants Configuration number of participants allowed to connect to the conference Conference ID 888 cannot exceed 95 Y New Participant Allowed To Join New Participant Type drop down list is used to select the state ee speaker or listener only of the new participants connected to om the conference New Participant Confirmation Type Password Y W New Participant Can Make Video Call 4 New Participant Can Activate The Conference Selecting the New Participant Can Make Video Call checkbox will allow participant to join the video conference Y Conference Inactive Until Moderator Login 1 9 New Participant Indication New Participant Confirmation Type drop down list is used to select whether the conference is password protected for the new users or not 1 With this option enabled participants will not be able to join the conference until the moderator has logged in New Participant Confirmation Type field should also be set to Password to enable this option Selecting the New Participant Can Activate Conference checkbox will allow new users to activate the conference When Conference Inactive Until Moderator Login option is enabled participants will not be ab
264. ess to the Third Party Call Control interface on this device Allow or block access to the SIP services on this device List of hosts whose access to any services on this device is blocked List of hosts having access to all services on this device Define the service names associated with the external ports Group IP addresses with names aliases for easier use in filtering rules Enable SIP intrusion detection system IDS to help prevent SIP attacks Fig II 177 Firewall Menu page e C Pyg l QX50 QX200 QX2000 Manual Il Administrator s Guide Firewall The Firewall Configuration page allows setting up a firewall configuring the security level and enabling the NAT and IDS services of QX IP PBX A Firewall is a security service configured by the QX IP PBX administrator based on various criteria The firewall allows or blocks traffic based on policies services and or IP addresses The firewall has several levels of security policies low medium or high The administrator may add additional service based rules Filtering rules will take effect only if the Firewall has been enabled and are independent from the selected firewall security level NAT Network Address Translation is used to allow QX IP PBX LAN members to connect to the Internet using QX IP PBX s WAN IP address The QX IP PBX NAT also handles forwarding incoming packets from the WAN to the PCs or devices on QX IP PBX s LAN The IDS Intrusion Detection System is a
265. est efforts and take all reasonable steps to safeguard the Licensed Materials to ensure that no unauthorized person shall have access thereto and that no unauthorized copy publication disclosure or distribution thereof in whole or in part in any form shall be made 7 Limited Warranty The only warranty the Licensor makes to you in connection with this license is that the media on which the Licensed Materials are recorded will be free from defects in materials and workmanship under normal use for a period of one 1 year from the date of purchase the Warranty Period If you determine within the Warranty Period that the media on which the Licensed Materials are recorded are defective the Licensor will replace the media without charge as long as the original media are returned to the Licensor with satisfactory proof of purchase and date of purchase within the Warranty Period This warranty is limited to you as the licensee and is not transferable The foregoing warranty does not extend to any Licensed Materials that have been damaged as a result of accident misuse or abuse EXCEPT FOR THE LIMITED WARRANTY DESCRIBED ABOVE THE LICENSED MATERIALS ARE PROVIDED ON AN AS IS BASIS EXCEPT AS DESCRIBED ABOVE THE LICENSOR MAKES NO REPRESENTATIONS OR WARRANTIES THAT THE LICENSED MATERIALS ARE OR WILL BE FREE FROM ERRORS DEFECTS OMISSIONS INACCURACIES FAILURES DELAYS OR INTERRUPTIONS INCLUDING WITHOUT LIMITATION TO ANY IMPLIED WARRANTIES OF MERCHA
266. eters The progress will be displayed in the area below Fig IT 280 Download Legible Configuration page The Cancel generation process button appears when the configuration generation procedure starts and it is used to stop it The Download generated configuration button becomes available when the legible configuration generation is finished It is used to download the generated file to the PC in a plain text format Necessary changes can be made in the downloaded configuration a file and then uploaded back to the system Attention Make sure the changes you have done in the Teereting throug the administrator configuration downloaded legible configuration file are valid and will not corrupt F maintenance o ee finished the system when being uploaded back to device The View generated configuration button becomes available when the legible configuration generation is finished It is used to view the generated file directly in the browser The Restart generation button becomes available when the legible configuration generation is finished It is used to cancel the generated configuration file and to start over Fig IT 281 Download Legible Configuration Configuration Summary Preview page Upload Legible Configuration The Upload Legible Configuration page is used to upload a configuration file in a text format The Choose File button in the q Ge ee EE opened page is used to browse certain leg
267. ettings checkbox is not selected on the previous page This page is used to configure advanced country settings Page consists of the following components ANI Category drop down list appears only when R2 signaling selected from Signaling Type drop down list on the CAS Signaling Wizard Page 2 is different from R2 DTMF is used to select the calling party priority depending on the call originator s location specifics ANI Request Transmit and ANI Request Receive drop down lists allow you to select the Caller ID request R2 tones for transmit and receive Seize Acknowledge Timeout text field is used to define a timeout in a range from 2 to 2000 milliseconds between incoming seize signal and the corresponding feedback CAS Signaling Wizard Answer Guard Timeout text field is used to define a wait timeout in a range from 0 to 1000 milliseconds Group B ae AR NE SNE Answer Signal and Line Answer 1 Selected Timeslots Release Guard Timeout text field is used to define an idle i timeout in a range from 0 to 120000 milliseconds between the Trunk disconnect signal receipt and call disconnection ES Dialing Delay Timeout text field is used to define a timeout in Seize Acknowledge Timeout 2 2 2000 msec a range from 0 to 2000 milliseconds before injecting dialed digits Timeout specially refers to R2 DTMF signaling Answer Guard Timeout 150 0 1000 msec Incoming DNIS Size text field indicates the number of received Release Guard
268. eue size Call Forwarding if Busy and Voice Mail telephony services will not take effect on the corresponding extension until the call queue is not filled These telephony services will affect only the calls out of the call queue Go To User Settings Go To Line Settings Go To Codec Settings The Max Calls Presented to Extension text field requires the ZeroOt Rerction maximum number of active calls on the line For example if 1 is configured in this field and the extension is in use the next incoming call will go to the call queue If 2 is configured in this field and extension is in use the next incoming call alert will be heard in the background if Call Waiting service is enabled on the corresponding extension and the extension will hold the first call Upcacrancatavevewscone message Choose Fie enterdepartment wav to answer the second one or they can be joined for a call Ulead new cal queue menage Choose Fie No file chosen conference However the next incoming call will again go to the call queue Enable Redirection Timeout checkbox is used to enable the call redirection to the other destination after some time spent in the queue This will avoid the caller to wait in the queue for too long This checkbox selection enables the following components Fig II 36 Extensions Management Edit Entry Call Queue Settings page Call Queue Message Repetition Count text field requires the number of call queue messages played
269. evice is being lost and you need to wait until the firmware a ER RUE ETE EEEE will be burned on the QX IP PBX You can do that right now by ccking the following nie Download Confizuraicn Warning Make sure the Firmware Update process is not disrupted until it is completed A power down while upgrading may cause serious damage The Download and Update functional button will automatically 6 UMD pvc anes ater sed poo erecto tiene download and update the firmware version from the FTP server a User Rights Upload Firmware Get Firmware From Server Automatic Firmware Update Firmware update Pressing the Download and Update functional button a new page with firmware download process will be displayed This page displays non editable information about the image validity Last Status shows that firmware download and updating process is running Fig II 289 Firmware Update page QX50 QX200 QX2000 SW Version 6 0 x 170 QX50 0X200 0X2000 Manual II Administrator s Guide Automatic Firmware Update The Automatic Firmware Update page allows you to configure an automatic update of the QX IP PBX s firmware software image as it becomes available on the server When this service is enabled on the configured day and time QX IP PBX will automatically check for a new available firmware on the server and will either notify the administrator or update the firmware right away depending on the configured settings The server configuration c
270. evious page of the wizard This page contains the following components ISDN L2 Timers e Excessive Ack Delay T200 configures the period in milliseconds numeric values from 500 to 9999 between the transmitted signaling packet and its acknowledgement received e Idle Timer T203 configures the period in milliseconds numeric values from 1000 to 99999 for the ISDN client idle timeout QX50 QX200 QX2000 SW Version 6 0 x 87 epygl ISDN L3 Timers E1 T1 Trunk ISDN Trunk PSTN Gateways e The T302 Timer text field requires the value for the T302 timer in milliseconds digit values from 0 to 15000 It KOEN eee indicates that the time frame system is waiting for a digit to i lela be dialed When the timer expires it initiates the call L2 amp L3 Settings Trunk 1 192 168 74 135 5060 e T309 Timer requires the value for the T309 timer in e milliseconds numeric values from 0 to 90000 It is the Tin 1209 ao 100000000 yee ci responsible for call steadin ess during link disconnection 7302 Timer 4 0 15000 msec Bearer Establishment Procedure on progress indication with in band information Y a P e i 5 T309 Timer 0 90000 msec Called Party Type of Number Unknown v within the period equal to this timer value If the value in this OA anne field S Zero 0 the T3 09 timer will be disabled Alert Guard Timeou 5 0 500 msec Called Party Numbering Plan ISDN telephony numbering plan Y Calling Party Nu
271. ext field requires the time period in seconds during which the call will be captured e Start button is used to start the active call capture To do that a checkbox beside an active call in the table should be selected and Start button should be pressed Note that only one call can be captured at the same time The Stop button appears when the call capture procedure is in progress and is used to stop the capture procedure e Download Capture and Remove Capture links appear on the page once the call is already captured The Download Capture link is used to download the captured call as an archived tar file which contains two streams receive and transmit of the corresponding call The files can be then played with an audio application The Remove Capture link is used to remove the captured audio stream The Interfaces sub page lists all available interfaces on the QX IP PBX Manipulation radio buttons allow you to select the needed line or trunk to be captured e Capture Timeout text field requires the time period in seconds during which the selected interface will be captured e Start button is used to start the capture of the selected interface The Stop button appears when the interface capture procedure is in progress and is used to stop the capture procedure e Download Capture and Remove Capture links appear on the page once the selected interface is already captured The Download Capture link is used to download the capt
272. ext fields used to insert the MAC Address of the corresponding SIP phone Use Other selection if your SIP phone is not in this list e Line Appearance text field requires a number of simultaneous calls supported by the SIP phone Username and Password are required for this selection They should match on both the QX IP PBX and the SIP phone for a successful connection The Password field is checked against its strength and you may see how strong is your inserted password right below that field To achieve the well protected strong password minimum 8 characters of letters in upper and lower case symbols and numbers should be used If you are unable to define a strong password press Generate Password to use one of system defined strong passwords s Transport drop down list is used to select the SIP protocol transport layer UDP TCP or TLS For TLS you may activate the TLS certificate update mechanism from IP Phone to obtain the latest certificate generated by the QX IP PBX Interfaces PR Interfaces IP Lines IP Line Settings IP Phone Templates IP Phones Logo FXS Gateways IP Lines Available IP Lines Attached Extension Type Details SIP UserName locext103 Model Aastra 6739i 00 08 5D 13 BC 15 Template systemdefault SIP UserName locext104 Model Cisco SPA525G2 d0 d0 fd e9 65 f0 Template systemdefault SIP UserName locext105 Model Yealink SIP T19P 00 15 65 54 3e f6 Template systemdefault SIP UserName locext106
273. extension accesses his mailbox by dialing 0 the call will be redirected to the voice mailbox on the proxy server Go To Line Settings Go To Codec Settings eo If the remote Voice Mail Server acts as a standalone location of voice mails it is recommended to select Independent Mailbox Type With this selection QX IP PBX redirects the recorded voice mails to the defined remote Voice Mail server When extension accesses his mailbox by dialing 0 the call will be redirected to the remote voice mail server For each of these selections it is required to enter the SIP URI of the Voice Mail Server where voice mails of the corresponding extension will be collected The Transport Protocol for SIP messages radio buttons allow the transport protocol UDP or TCP for transmission of SIP messages to be selected Fig II 37 Extensions Management Edit Entry Voice Mailbox Settings page With MS Exchange Server you can keep recorded voice messages into one universal inbox o UM Auto Attendant URI text field requires the SIP URI of the MS Exchange Server When extension accesses his mailbox by dialing 0 the call will be redirected to the voice mailbox on the MS Exchange Server o UM Extension text field requires an extension number that Unified Messaging will use when voice mail is submitted to the user s MS Exchange Server mailbox Please Note When the MS Exchange Server option is selected as an external voice mail se
274. fault Send AA digits to Routing Table disabled Redirection on Timeout disabled ZeroOut disabled Welcome Message enabled Ringing Announcement disabled Welcome Message Recurring Attendant Prompt and Attendant Ringing Announcement default Registration username 00 Registration password empty SIP server empty SIP Server port 5060 SIP Server Registration disabled Same as for extensions Codecs G711u preferred G711a G726 16 G726 24 G726 32 G726 40 G729a iLBC enabled H 263 H263 and H 264 disabled Out of Band DTMF Transport enabled T 38 FAX enabled Pass Through FAX enabled Pass Through Modem disabled Force Self Codecs Preference for Inbound Calls disabled SRTP Policy Accept anything Feature is disabled by default For QX50 QX200 Percentage of System Memory 1 For QX2000 Percentage of System Memory 0 08 No entries No entries Undefined 178 Parameter Authorized Phones Database IP Lines Settings FXS On board settings FXO Settings E1 T1 Trunk Settings ISDN Trunk Settings External PSTN Gateways VoIP Carrier Call Routing Table Call Routing Local AAA Table Global Speed Dial Directory SIP Tunnel Settings Class of Service QX50 QX200 QX2000 SW Version 6 0 x QX50 0X200 0X2000 Manual II Administrator s Guide System Default Value No entries IP Lines Configuration Enable PnP for IP lines enabled Enable firmware version co
275. fault settings functional button resets all configuration settings and restores the board s factory Restore to Factory Default settings Reset default configuration By restoring the default configuration you A Maintenance will replace your current configuration lose all voice mails and reboot the device You will not be automatically redirected to the GUI start page After the successful reboot you will need to enter into the management page and login again to access the QX IP PBX s configuration A warning message will ask you to confirm your selection before restoring the default E e A C OE Ap Managemen PAGE QX50 QX200 QX2000 SW Version 6 0 x 166 e C Pyg l QX50 QXK200 QX2000 Manual II Administrator s Guide configuration Please Note Unlike the factory default settings restore procedure initialized from the Reset button on the QX IP PBX board this link will keep the following data e Call History Transfer Statistics System Events Feature Keys e Device Registration state Automatic Backup The Automatic Backup page allows you to enable the automatic backup of the system configuration and the voice data on the QX IP PBX With this service QX IP PBX will automatically backup the system configuration and the voice data and store it in the specified location Y Enable Automatic Backup This page contains the following components Maintenance O Sena vis Ems E mail Address astghik_sanasaryanQepygi
276. ference will yes be impossible Fig II 70 Extension Codecs list Move up moves the selected codec one level up increasing the codec s priority Move down moves the selected codec one level down decreasing the codec s priority Make preferred moves the selected codec to the top of the table setting its priority to the highest Clicking the Make preferred button when a disabled codec is selected will first enable the codec and then move it to the top The following settings are available for user extensions and attendants only Out of Band DTMF Transport enables the DTMF code transmission in parallel with the voice stream Destination received the DTMF code will play it locally if it supports the feature too This helps avoid DTMFs loss in case of heavy traffic The feature is valuable for all codecs but it is especially recommended for low bit rate codecs such as G 729 G 726 16 etc Enable T 38 FAX enables the T 38 codec support of FAX transmission for incoming unified FAX messages fax to mailbox and remote IP devices connected to Epygi unit via routing rules which using the target extension user settings UES Enable Pass Through FAX enables the G 711 codec support for incoming unified FAX messages fax to mailbox and IP devices connected to the attached IP line If both of the above checkboxes are enabled the T 38 codec will be used as a preferred codec for FAX transmission If it is not supported by the peer the G 711 codec wi
277. ffers the following components MAC Address Assignment manipulation radio buttons e This Device turns to the default MAC address of the QX IP PBX e User Defined requires user defined MAC Address The MTU drop down list allows you to select the maximum packet size on the Ethernet in bytes MTU is used to fragment the packets before transmitting them to the network The MTU preferred value is dependent on the Ethernet connection The default MTU size is 1500 Bytes for Ethernet and 1400 Bytes for PPPoE Needed Bandwidth for IP Calls Setup Setup System LAN Date and Time Email SMTP Short Text Messaging SMS Internet Configuration Wizard WAN IP Configuration IP configuration ofthe WAN interface Assign automatically via DHCP Assign manually IP Address 192 168 74 12 Subnet Mask 255 255 255 0 Default Gateway 192 168 74 5 Previous Fig II 9 Internet Configuration Wizard WAN IP Configuration page System LAN Date and Time Email SMTP Short Text Messaging SMS Internet Configuration Wizard WAN Interface Configuration MAC Address Assignment This device This device 40 40 00 0 97 72 User defined Maximum Transfer Unit MTU MTU 1500 Bytes Previous Fig II 10 Internet Configuration Wizard WAN Interface Configuration page The bandwidth required by an IP call depends on the codecs used and these specifications are listed in the
278. g arrows will be displayed next to the column heading Add opens the Add Entry page where a new IP range can be added QX50 QX200 QX2000 SW Version 6 0 x General SIP Parameters RTP Paramet ters STUN Parameters NAT Exclusion NAT Traversal Settings Add Edit Delete IP address Subnet Mask 192 168 0 0 255 255 0 0 Fig Il 162 NAT traversal Settings NAT Exclusion Table page 110 pya The Add Entry page includes the following text fields IP address requires the IP address that is placed behind NAT a within the local network General SIP Parameters RTP Parameters NAT Traversal Settings NAT Exclusion Table Add Entry Subnet Mask requires the subnet mask corresponding to the O Go Back specified IP address Telephony 1p address 192 168 0 0 Subnet Mask 255 1255 10 0 Save Fig II 163 NAT traversal Settings NAT Exclusion Table Add Entry page To Configure the NAT Exclusion Table 1 Press the Add button on the NAT Exclusion Table page The Add Entry page will appear in the browser window 2 Specify an IP Address and its Subnet Mask in the corresponding text fields 3 Press Save on the Add Entry page to add the selected IP range to the NAT Exclusion Table list To Delete an IP Range from the NAT Exclusion Table 1 Select the checkboxes of the corresponding IP range s that should to be deleted from the NAT Exclusion Table 2 Press the Delete button on the NAT Exclusion Table
279. g Attendant Prompt Choose File No file chosen Call To text field requires the destination number dialed in the format depending on the selected Call Type The Attendant Ringing Announcemen wildcard is supported in this field A Choose File No file chosen e 7ZeroOut this group is used to configure call redirection service on the Auto Attendant When a caller reaches the Auto Attendant he may want to accelerate the automatic redirection feature instead of using Auto Attendant features To activate ZeroOut caller should dial O digit see Feature Codes during the Auto Attendant welcome message The caller will then be automatically transferred to the destination specified in this page Fig IT 62 Extensions Management Edit Entry Attendant Scenario page Enable ZeroOut checkbox selection enables the ZeroOut feature and activates the following fields to be inserted Redirect Call Type drop down list includes the available call types O PBX local calls between QX IP PBX extensions and the Auto Attendant o SIP calls through a SIP server o PSTN calls to PSTN O Auto used for undefined call types Destination independent on whether it is a PBX number SIP address or PSTN number will be reached through Routing The Redirect Address text field requires the destination address where the caller should be automatically forwarded to if activating the ZeroOut feature Attention The routing patterns in the Call Routi
280. g appears if you have selected the eRe same signal type both for receive answer and receive Enable 87 busy recognitions e Partial Enable selection partially enables Group B eE Support with for answer recognition only This ee selection requires you to define transmit and receive Enable 813 Enable B14 signals The Transmit Answer Signal parameter is defined from the drop down list on this page When transmit signal is selected press Next on this page to access the R2 Receive Signal Settings page where donc Receive Answer Signal should be defined Use the checkboxes to select the Receive Answer Signal value Multiple values are allowed for each signal Y Enable B15 e Disable selection disables Group B Support and requires defining the Answer Signal parameter Fig IT 120 CAS Signaling Wizard Receive Signal Settings page The Trunk CCS Signaling Settings page allows configuring CCS signaling settings and gives a possibility to select timeslots for signaling data transfer receive and voice transfer The page consists of the following components QX50 QX200 QX2000 SW Version 6 0 x 81 QX50 0X200 0X2000 Manual II Administrator s Guide The Non Automat checkbox switches to non automatic Terminal Endpoint Identifier TEI searching and enables the TEI Address text field that requires a TEI number digit values from 0 to 63 for connection establishment between CO and E1 T1 client In automatic ne mode an E1
281. g area This equipment cannot be used on the telephone company provided coin service Connection to Party Line Service is subject to State Tariffs If this equipment causes harm to the telephone network the telephone company will notify you in advance that temporary discontinuance of service may be required If advance notice isn t practical the telephone company will notify the customer as soon as possible Also you will be advised of your right the file a complaint with the FCC if you believe it is necessary The telephone company may make changes in its facilities equipment operations or procedures that could affect the operation of the equipment If this happens the telephone company will provide advance notice in order for you to make the necessary modifications in order to maintain uninterrupted service If trouble is experienced with this equipment please contact EPYGI TECHNOLOGIES LTD If the trouble is causing harm to the telephone network the telephone company may request you to remove the equipment from the network until the problem is resolved Electrical Safety Advisory To reduce the risk of damaging power surges we recommend you install an AC surge arrestor in the AC outlet from which the Quadro or QX is powered Industry Canada Statement This product meets the applicable Industry Canada technical specifications Safety Information Before using the Quadro or QX please review and ensure the following safety instruction
282. g the Registration Timeout Registration Failure Timeout Transaction Duration and Session refresh timeout SIP timers in seconds SIP Aliases This page is used to create a list of QX IP PBX s hostnames register on remote DNS servers This list will be used to identify SIP packets received from remote servers where QX IP PBX is registered with different names The Host aliases for SIP page consists of a table where QX IP PBX s aliases are listed Add opens the Add Entry page where a new alias name for QX IP PBX should be defined TLS Certificates The Generate and Install New CA Root Certificate page is used to define generate and install a new CA root certificate for SIP TLS traffic All fields in this page require root certificate specific information The General Certificate and Install button is used to generate a new CA root certificate based on the defined data and to install it on the QX IP PBX QX IP PBX will get rebooted automatically once the new certificate is installed You may download the actual copy of the certificate from SIP Settings page To ensure a secure TLS connection with the QX IP PBX s defined CA root certificate both sides should have the same certificate installed If the end user is an IP phone you may activate the TLS certificate update mechanism from it to obtain the latest certificate generated by the QX IP PBX If the end user is a server or other device you may download the certificate from the QX I
283. ge 6 Voice Mailbox Settings This group is used to configure voice mailbox storage and consists of a group of manipulation radio buttons to define the location where voice mails will be collected e Disable Voice Mail disables the Voice Mail service for the corresponding extension With this selection the extension user will be unable to reach their Voice Mail Settings but will be able to access their Voice Mailbox and manage the existing voice mails e Use Internal Voice Mail enables the Voice Mail service for the corresponding extension and defines the QX IP PBX s internal storage as a location for the Voice Mails This selection also allows you to manipulate with the Voice Mailbox Settings used by the extension s user to setup personal settings the password the voice mail greeting message and the user s name for Extensions Directory from the handset By default the Voice Mailbox Settings is enabled when the QX IP PBX s is in the factory reset state It can be manually enabled from this page by pressing the Activate button When the Voice Mail is activated the extension s user is prompted to insert personal settings as he she enters his her Voice Mailbox for the first time Unless the required information is not inserted the button is changed to Deactivate and the Configuration Wizard Status becomes Activated Use Deactivate button to stop Voice Mail Configuration Wizard When the user inserted the required information the Config
284. gents configured with AG If the agent doesn t answer within Ringing Timeout the system tries to reach the next agent in the list etc Reaching the end of the list it starts from the beginning again If the call is not answered and the Common Timeout has expired the system either disconnects or redirects the call e Longest Idle the system calls to the first available agent who was longest idle after the last call If the agent doesn t answer within Ringing Timeout the system tries to reach another agent who was longest idle etc If the call is not answered within Common Timeout the system either disconnects or redirects the call Less Busy During Last Hour the system calls to the first available agent who was least busy during the last hour in average If the agent doesn t answer within Ringing Timeout the system tries to reach the next least busy agent etc If the call is not answered within Common Timeout the system either disconnects or redirects the call Random Hunting the system calls to the first available agent selected randomly from the list of agents configured with Agents Group If the agent doesn t answer within Ringing Timeout the system tries to reach another agent selected randomly from the list etc If the call is not answered within Common Timeout the system either disconnects or redirects the call e Skills the system calls to the first available agent with the highest composite skill s grade in the group
285. gh space warning message will appear Conference Settings Ss Conference General Recording Participants Schedule Send Notification Mail Customization Conference ID 888 Y Play First in Conference message Welcome Message Choose File opens the file chooser window to browse for a new welcome message file Upload new Welcome Message Choose File attwelcome wav Hold Music File The Download Welcome Message and Remove Welcome Message links appear only if a file has been uploaded previously The Download Welcome Message link is used to download the message file to the PC and opens the file chooser window where the saving location may be specified The Remove Welcome Message link is used to restore the Upload new Hold Music file Choose File No file chosen Info File Upload Info file Choose File No file chosen default welcome message Hold Music File parameters group allows updating the hold music played when you are alone in the conference downloading it to the PC or removing the custom welcome message The group offers the following components Fig II 299 Conference Settings Customization page Upload new hold music file indicates the file name used to upload a new hold music file The uploaded file needs to be in PCMU wave format otherwise the system will prevent uploading it and the Invalid audio file or format is not supported warning message will appear The system also prevents up
286. groups where he she is online Away the agent is logged in but temporarily Fig l 56 Agents Table of Group Add Entry page unavailable for a short time by some reason amp DND Do Not Disturb agent is busy by some other activity not related to conversation on the phone For example agent can be busy by updating the customer s record after the call or entering some data into database Versus to Away status the DND state of the agent changes automatically to Online when the preconfigured DND timeout expires it is now 30 seconds by default Please Note The state of the Agent can also be modified from the handset by calling the predefined Auto Attendant see Attendant Extension Settings and ACD Management Enable wrap up if enabled the current Group doesn t send new calls to the Agent within the wrap up Timeout after closing the active call Versus DND the agent s status doesn t change during Timeout period which activates automatically every time when the agent finishes the call That period is used for example by the agent for updating the customer s records after the call Move Up and Move Down buttons are used to move the selected entry one level up or down within the Agents Table The sequence of Agents is important when Round Robin call distribution is selected in the ACD Group Settings page see above Agents will be called in the order selected in the Agents table Recording Box Extension Settings
287. gs page or Auto Attendant has a proper SIP registration see Attendant Extension Settings Remote Call Back The Remote Call Back Configuration service is used by authorized callers to configure or reconfigure existing call back configuration on the QX IP PBX Remote Call Back Configuration is divided into two modes accessible from the QX IP PBX s Auto Attendant Permanent Call Back and Non Permanent Call Back Please Note Remote Call Back Configuration services are only available when the Automatically Enter Call Relay Menu checkbox is disabled in Authorized Phones Database for the trusted user Permanent Call Back service allows callers registered in the Authorized Phones Database to create a new trusted caller with Call Back enabled They can also modify the Call Back destination of existing callers in the Authorized Phones Database By calling QX IP PBX s Auto Attendant and entering the Auto QX50 QX200 QX2000 SW Version 6 0 x 65 QX50 0X200 0X2000 Manual II Administrator s Guide Attendant menu the caller can use the 00 code see Feature Codes to create a new trusted caller as well as to modify the Call Back destination for the already registered callers in the Authorized Phones Database By entering Permanent Call Back reconfiguration menu system asks caller to login by dialing the number and an appropriate password for the QX IP PBX s extension that is used as login extension in the Call Back settings After passing the login c
288. gs link is only present on backup device and is used to download system logs to the local PC as a tar archive file These logs can then be used by the Epygi Technical Support Office to determine the problem that has occurred on your QX IP PBX Language Pack The Language Pack page allows you to upload a custom language for GUI and Voice Messages of the QX IP PBX The language of voice messages can be switched to the custom Language Pack language from the GUI setting page in the System Configuration Wizard The language of GUI session can be changed to the custom Language Pack language from the radio buttons on the login page QX50 QX200 QX2000 SW Version 6 0 x 20 QX50 0X200 0X2000 Manual II Administrator s Guide QX50 0X200 0X2000 Manual II Administrator s Guide Uploading a language pack will also change the language of some supported IP phones Aastra snom v 6 x Grandstream GXP2000 After a custom Language Pack is uploaded onto the system reboot the IP phone to load a matching language onto the phone Uploading a Language Pack will cause the loss of the following data e All voice mails and custom voice messages only when embedded memory storage is used cas e Call History only when embedded memory storage is used A A Areboot e Pending Events only when embedded memory storage is Anew language pack replaces the existing one used Current language pack Italian Italy x8 Language pack file to upload Ch
289. guration file with parameters present on the RTP Settings page e The Group of web pages selection allows you to choose among the four predefined groups Internet Connection Settings LAN Configuration Settings Telephony General Settings and Extension Settings Each of these groups refer to all pages characterized by the selected criteria e g Internet Connection Settings group contains all parameters on the pages related to the networking and WAN configuration QX50 QX200 QX2000 SW Version 6 0 x 167 QX50 0X200 0X2000 Manual II Administrator s Guide The Extension drop down list allows you to limit the settings in the generated legible configuration file to one specific extension For example each of the extensions on the QX IP PBX have own SIP Ssckup Rescre Automate ccoo EMP settings or Codecs To download the settings for a particular Configuration Summary extension only you need to choose the corresponding extension from the list The drop down may also have a blank selection In that case the legible configuration file will contain the parameter of Se aa a aaa all available extensions on the QX IP PBX if the selected parameter iia applies to the extension and not to the overall system like RTP settings Single Page Call Recording Start generating a legible configuration file The Start generate a legible configuration file button start parsing the configuration structure of the device for the defined param
290. gured User Input options can be adjusted scription oa 8 yY Choose File No file chosen The Add Edit Option page offers the following components Description text field for an optional description of the option Option is used for choosing the user input for which some Go to the following menu announcement and or action should be configured The following input options are available in the list to configure the Customized Scenario Call to the extension Call to the following number Invoke Extension Directory e Digits ina range from 0 to 9 Terminate the Call Signs and 4 Announcement indicates the file name used to upload a new custom message When the caller selects the option configured in the Option drop down list this message will be played once before the Action will be activated Fig II 64 Main menu Add Option Edit Scenario page Attention The uploaded file needs to be in PCMU CCITT u law 8 kHz 8 bit Mono wave format otherwise the system will prevent uploading it and the Invalid audio file or format is not supported warning message will appear The system also prevents uploading if there is not enough memory available for the corresponding extension and the You do not have enough space warning message will appear The Download and Remove links appear only if a file has been uploaded previously The Download link is used to download the message file to the PC and opens the file c
291. gured every filter correctly Basic protection against the most common attacks port scans flooding etc is still provided with this policy 4 Enable Firewall e Medium Security Traffic originating from the LAN side may pass and traffic from the WAN side will be blocked by default This is the recomme nded security level High Saciirity Everything that is not explicitly allowed will be blocked This includes traffic from the LAN side You have to configure the filters to open up the firewall as desired Medium Securi ld Traffic originating from the LAN side may pass and traffic from the WAN side will be blocked per default This is the recommended policy e High Security Everything that is not explicitly allowed will be blocked including traffic from the LAN side Fig II 178 Firewall Settings page Advanced Firewall Settings Advanced Firewall Settings are used to deny Ping and Portscanning operations addressed towards the device With these features enabled QX IP PBX will answer with inscrutable messages to the Ping and Portscanning operations a ia id Firewall NAT Advanced IDS Lo Please Note Operations are available only when the firewall is enabled from the Firewall and NAT page Advanced Firewall Configuration Any changes here forces a restart of the firewall which might take a few seconds This page offers the following components 2 Ping Steath A Firewall Fool Portscanner The Ping Stealth checkbox select
292. handset added user should accept the conference invitation by pressing the button IAS Participant Type Speaker Selecting the Allow Video checkbox will allow participant to join the video conference Selecting the Participant Indication checkbox will enable a beep lo indication during the active conference when a handset added user AAA joins or leaves the conference The Allow Duplicated Participation checkbox selection allows several instances of callers with the same handset added number caller address to join the corresponding conference at the same time This option may be used to allow users from the same network with the same caller address like PSTN network to reach the conference W v Confirmation Type amt Allow Duplicated Participation Save Fig IT 304 Conference Settings Handset Added Participants Configuration page Schedule The Schedule page is used to configure and manage the conference scheduling rules so that a conference can be automatically activated on the date and time The Scheduling service may also be configured to send invitation emails to the participants asking them to join the conference or informing about a new conference Conference ID 888 Add Edit Delete Conference Schedule The Conference Schedule page offers a table that lists all scheduling rules configured for the corresponding conference When a scheduled conference is activated all participants with dial out opt
293. he FXS Gateway Configuration Wizard FXS Gateway Model page contains following components e The FXS Gateway Model drop down list is used to select the FXS Gateway model to be used as an FXS expansion device e The MAC Address text fields require the MAC Address of the FXS Gateway Based on the selected FXS Gateway model and the inserted MAC Address the FXS Gateway can be automatically configured by simple reset reboot e The Description text field requires the description of the FXS Gateway to be configured The next page of the wizard is FXS Gateway Configuration Wizard FXS Gateway Lines This page displays a list of FXS lines provided by the FXS Gateway and is used to assign each FXS line to an IP line on the QX IP PBX System will automatically assign the provided FXS lines to the first available IP lines on the QX IP PBX You may adjust the configuration from this page Please Note The FXS lines can be assigned only to inactive IP lines on the QX IP PBX If there are no enough free IP lines available on the QX IP PBX you should first deactivate the IP line from the IP Line Settings page to use it in the FXS Gateway Configuration Wizard The next page of the wizard is FXS Gateway Configuration Wizard Summary where the configured settings should be verified Once FXS Gateway Configuration Wizard terminates a new entry is added to the table and the corresponding FXS Gateway s configuration gets updated according to the se
294. he QX IP PBX s system configuration and the voice data will take place Backup Now button is used to perform a manually immediate backup of the system configuration and the voice data Download Legible Configuration The Legible Configuration Management page is used to manually manage the configuration on the QX IP PBX This will allow you to download a piece of configuration from the QX IP PBX in the way of legible file to make necessary changes in that file and to upload it back to the same or different QX IP PBX s With this service some pieces of configuration like extension settings NAT settings etc of one QX IP PBX can be used on another QX IP PBX This also helps to apply the same group of settings to the several instances for example to apply the same SIP settings to multiple extensions on the QX IP PBX on the same or different QX IP PBXs avoiding manual configuration of each of those instances i e extension from the web management on each of the QX IP PBXs The QX IP PBX reseller distributor ISP or carrier usually uses this service The manipulation radio buttons are used to select between particular page or a named group of pages for which the legible configuration file will be generated e The Single Page selection allows you to choose a certain page from the list of QX IP PBX s Web management pages for which the legible configuration can be manually managed For example selecting RTP Settings will generate a legible confi
295. he Voice Mail Service or the call queue of the called extension depending on the extension user s configuration If you have selected the snom 320 360 370 720 760 820 821 870 Grandstream GXP 2000 2100 2110 2120 2124 Yealink SIP T28P SIP T26P SIP T38G SIP T46G IP phones from the Phone Model drop down list the next page in the wizard will be the Receptionist Phone Configuration Wizard Hardware Modules For all other phone models this page is skipped For Grandstream GXP 2000 2100 2110 2120 2124 IP phones Expansion Modules Receptionist Phone Configuration Wizard Extensions Hardware Modules this page contains a single checkbox only Module 1 SnomVi0 The Enable Expansion Module checkbox is used to enable the Module 2 Snomv20 7 supplementary module attached to the IP phone The Expansion Mosul 3 SnomV20 Y Modules Count drop down list allows you to select how many additional expansion modules will be connected to the IP phone When the module is selected the number of programmable keys on the next page of the wizard is multiplied accordingly Previous For Aastra 67371 6739i 67551 and 67571 IP phones Receptionist Phone Configuration Wizard Hardware Modules page contains a number of drop down lists to select the types of the expansion modules and the sequence in which they are connected to the IP phone Fig II 84 Receptionist Phone Configuration Wizard Hardware Modules page QX50 QX200 QX2000
296. he administrator may access the settings in each respective category and perform actions specific to each category The following pages open when viewing a PBX extension or conference ID e Your Extension used to update the extension voicemail and user features see Manual III Extension User s Guide e Conference update conference settings for a particular conference ID The Return link is used to return to the Epygi QX50 QX200 QX2000 Management page Epygi QX200 Management Active Calls Call Start Time Call Duration Calling Phone Called Phone 26 Jun 2014 12 20 00 7415101 101 Terminate 26 Jun 2014 12 19 56 7415102 102 Terminate 26 Jun 2014 12 20 03 7415103 103 Terminate Active Calls Count 3 Firmware Version 5 3 20 Release Users currently logged in admin from 192 168 70 17 expires 15 53 admin from 192 168 70 17 expires 14 36 admin from 192 168 74 93 expires 12 40 admin from 192 168 7 27 expires 12 39 Internet connection status static IP Refresh in 593 seconds Fig II 1 Epygi QX IP PBX Management page The following links may also be displayed Renew WAN IP Address will be displayed if the Epygi QX WAN IP address is configured dynamically when the system performs as a DHCP client Pending Events link is located on the top right corner of the Epygi QX Management page and displays all system events that have occured Language selection is available only when the custom Language Pack has been
297. he following functional buttons are available on this page e Delete removes the selected record s from the system and Call History Archive table e The Clear all Records button is used to clear all statistics records QX50 QX200 QX2000 SW Version 6 0 x Successful Calls Missed Calls Unsuccessful Outgoing Calls Settings Archiving Settings Call History Archive Filter Oct El Delete Archive Record Number of Call Records External Backup Status Total 3 Total 3 Clear all Records Fig II 249 Call History Archive page 152 QX50 0X200 0X2000 Manual II Administrator s Guide Archiving Settings The Archiving Settings page is used to configure the automatic archiving of the Call History Overview System Status Events CallHistogy Successful Calls Missed Calls Unsuccessful Outgoing Calls Settings Archive Archiving Settings The Percentage of Total Memory used for Archive drop down Call History Archiving Settings list is used to select the internal memory space in percents that can be used for storing the archived Call History When the ener ere ere ee e required memory exceeds the size entered the oldest entries are eS being automatically deleted Gi Status Call Detail Records Archive Structure The Enable Call Detail Records Archive Collection checkbox Archive by records count or Count o enables automatic downloading mechanism of the Call History Please Note This service only refers to t
298. he last caller Key Function Vawe a e Linelnfo gets the IP line information from the QX IP PBX Line Key 2 Watch Ext 104 v W104 e CallBlk blocks the last caller Line Key3 Preconfigured e Record records the call in case if the manual call recording A is allowed for the call configured from LineKeys aa e Call Recording used for configuring the call recording rules a y Forsan Line Key 7 Watch Ext 302 v W302 e ACD Login Logout allows the corresponding ACD agent to login to all groups it is involved in if previously logged in to tine Key Preconigured log out from those groups For details on ACD functionality Figll 85 see ACD Management Receptionist Phone Configuration Wizard Programmable Keys Configuration page Please Note Once a new receptionist is created the Call Queue feature will be automatically enabled with the corresponding Call Queue Size and Max Call Queue Appearance settings on all extensions attached to the IP lines defined in the Attached IP Lines text field The next page of the wizard is a Receptionist Phone Configuration Wizard Summary where the configured settings for the receptionist should be verified Additionally this page Receptionist Phone Configuration Wizard contains a Reboot IP Phone now checkbox which should be 8 Extensions selected if you wish to have your IP phone rebooted once the Summary corresponding receptionist is created Reboot is needed for a hi
299. he statistics collected from the moment of enabling this service and forward any previously generated statistics will not be downloaded External Backup of Call Detail Records Archive Archive by time interval Time Interval 10 minutes Y Send archive files to external server Send and keep locally v The Call Detail Records Archive Structure is used to configure the intervals for archiving the Call History The archiving structure D se a AA allows to archive the Call History either by time intervals or per statistics record count Send to server Server Name 192 168 74 203 Server Port 21 Path on Server A20 CallStat The Call Records Count drop down list is used to select the Send Method portion size of the Call History including all types of call statistic i e successful missed and unsuccessful outgoing Call History which will be archived locally The number selected in this drop al TFTP User Name test203 down list indicates the number of entries in the single archived Call History file If there are no enough entries in the Call History table on the QX IP PBX the system will wait until the necessary number of entries will be collected and then will archive the a mums statistics file Password The Time Interval drop down list is used to select the time interval by which the Call History will be archived locally After each time interval the system will archive the Call History including all types of call statistic
300. hen this checkbox is enabled the js Tunnels to Slave Devices table needs to be configured Tunnels to Slave Devices The link Tunnels to Slave Devices moves you to the page where a list of slave devices needs to be defined Fig II 149 SIP Tunnel Settings page The Tunnels to Slave Devices page consists of a table where slave devices are listed with the corresponding authentication parameters Call Routing Table Call Routing Local AAA Table Global Speed Dial Class of Service Add functional button leads to the Add Entry page where a new SIP Tunnel Settings Tunnels to Slave Devices slave device parameters needs to be provided Add Edit Delete SIP Tunnel Name r Name Symmetric NAT Slave Device IP Port Registration Date Time The Add Entry page consists of the following components Pa es en eee N A The SIP Tunnel Name text field requires the tunnel name for the corresponding connection System suggests you to start the SIP tunnel name with the SIP_Tunnel_ words according to the automatic prefix used for the SIP tunnels on the QX IP PBX however this is not mandatory Fig II 150 SIP Tunnel Settings Tunnels to Slave Devices page The User Name text field requires the authentication user name The field in front of this text field displays the default non editable prefix for SIP tunnels SIPTunnel_ QX50 QX200 QX2000 SW Version 6 0 x 105 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s
301. her its priority oe The value of the metric Criterion 9 o The lower the metric of a pattern is the higher its priority The position in the routing table Criterion 10 The higher the position of a pattern in the routing table is the higher its priority Example The user has dialed 1231 and the following matching patterns have been found The list of patterns 1 123 11 15 3 2271 123 1 3 1 3 100 150 asd 1 12 31 1 1 3 3 0 8 1231 2 1 Step 1 The list is split into two groups separating the patterns with from those without Criterion 1 The patterns with form a group with a lower priority and are pushed back to the end of the list Criterion 1 The list split into two subgroups 2271 123 1 3 100 150 asd 1 1 1 3 3 0 8 1231 71 123 11 15 3 1 3 12 31 2 1 QX50 QX200 QX2000 SW Version 6 0 x 103 The list of patterns 41 123 Criterion 2 Matching digits WlolR PA e e RP ow 11 15 3 1 3 12 31 yA QX50 QX200 QX2000 SW Version 6 0 x OIN Bl PR Criterion 3 The list of patterns Matching digits 1 1 3 3 0 8 2 1231 4 100 150 asd 1 12 31 123 3 11 15 3 1 2d T 1 1 3 0 The prioritized list 1231 1 1 3 3 0 8 100 150 asd 1 123 2271 1 3 12 31 123 11 15 3 1 1 3 QX50 0X200 0X2000 Manual II Administrator s Guide The list of pattern
302. hooser window where the saving location may be specified The Remove link is used to restore the default welcome message Action is used to configure the action based on the caller s selection The Action radio buttons allows you to configure the action type after playing the Announcement message if configured QX50 QX200 QX2000 SW Version 6 0 x 46 QX50 0X200 0X2000 Manual II Administrator s Guide e No Action the Auto Attendant will continue to play the Recurring message if configured of the current menu e Go to the following menu will go to the specified submenu and take actions defined in that submenu The drop down list allows the selection of a previously created submenu or to create a new submenu by choosing the Create New Submenu item The New submenu name text field requires the new submenu name e Call To the following extension will call to the extension number specified in the extensions drop down list e Call to the following number will call the specified phone number via the Call Routing Table e Call to the number dialed will send the user inputs to Call Routing table and if there is a matching with any Call Routing rule the call will be made with the conditions of Call Routing rule available only in case when the Any input other than in the list above input is edited e Invoke Extensions Directory will connect the caller to Extensions Directory e Terminate the call will exit from this Customized Scenario and di
303. ible configuration file to Upload Legible Configuration be uploaded and updated into the system The configuration files Legible configuration meto unicas ANN at lt 2014721040 00 to be uploaded should be in the txt format otherwise a system error occurs Configuration file upload progress will be displayed e in the area below During legible configuration file upload QX IP F Maintenance PBX s functionality failures may occur Fig II 282 Upload Legible Configuration page Firmware Update This window allows updating the software of QX IP PBX by installing new firmware image Users registered at Epygi will receive a notice when new firmware is available and will be able to download it from the Epygi Technical Support WEB page Updating new firmware requires a working power supply QX IP PBX is provided with a battery accumulator If the battery is low or simply absent the There is no battery or voltage is low warning is displayed Please Note Installing new firmware will take about 15 minutes During this time QX IP PBX telephony and Internet access will be disabled Attention When the older firmware is installed on the QX IP PBX the system configuration will be lost and the device will be factory reset Please Note It is recommended to backup the configuration prior to upgrading the firmware You can do that by clicking the Download Configuration link which generates a backup file with all configuration settings and use
304. ic and foreign laws governing Voice over Internet Protocol VoIP calls QX50 QX200 QX2000 SW Version 6 0 x 194 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide 10 U S Government Restricted Rights The Licensed Materials are provided with RESTRICTED RIGHTS Use duplication or disclosure by the Government is subject to restrictions as set forth in subparagraphs c 1 and 2 of the Commercial Computer Software Restricted Rights clause at 48 C F R section 52 227 19 or subparagraph c 1 ii of the Rights in Technical Data and Computer Software clause at DFARS 252 227 7013 as applicable 11 Entire Agreement It is understood that this Agreement along with the Quadro or QX installation and administration manuals constitute the complete and exclusive agreement between you and the Licensor and supersede any proposal or prior agreement or license oral or written and any other communications related to the subject matter hereof If one or more of the provisions of this Agreement is found to be illegal or unenforceable this Agreement shall not be rendered inoperative but the remaining provisions shall continue in full force and effect 12 No Waiver Failure by either you or the Licensor to enforce any of the provisions of this Agreement or any rights with respect hereto shall in no way be considered to be a waiver of such provisions or rights or to in any way affect the validity of this Agreement If one or more of the
305. ically allocated IP addresses When QX IP PBX is connected to the Internet with a fixed IP address it will be set up to act as a VPN gateway QX IP PBX is then prepared to establish an IPSec connection with another VPN gateway device but also allows access to Road Warriors A notebook laptop used by a traveling employee could also be a Road Warrior Access to their company s intranet via an IPSec connection can be obtained regardless of their location QX IP PBX can also be set up to act as a Road Warrior If a home office is connected to the Internet via QX IP PBX with PPPoE Point to Point Protocol and dynamic IP addressing setting up QX IP PBX as a Road Warrior will allow an IPSec connection to the corporate network For the encryption and decryption of the data transmitted via the IPSec connection a Key is used RSA used by QX IP PBX is an asymmetric key system It has to be available on both sides of the IPSec connection and will generate a different pair of keys on each side a private key and a public key During the connection establishment some data is encrypted with the remote party s public key They can be decrypting the data with their private key and the data encrypted there with QX IP PBX s public key can be decrypted with QX IP PBX s private key Since the private key is never transmitted it stays completely unknown to everyone thus the system remains safe Even if someone gets the public key decryption cannot be possible with
306. ices Edit Extension Caller ID Based Services Granted grouplist Extension Group List Granted grouplistadd Extension Group List Add Edit Member Granted instanthotdesk Service Code for Instant HotDesking Denied Fig I1 277 User Rights Management Edit Roles page Backup Restore The Configuration Management page assists the administrator with managing the system configuration settings and voice data For example the administrator is able to backup and download the settings to a PC and then upload and restore them back to the QX IP PBX Additionally this page provides the possibility of restoring the factory default configuration settings The Backup and download current Configuration Download button generates a backup file with all configuration settings and user uploaded greeting messages It opens a file chooser window for immediate download to the users PC The Restore previously backed up Configuration Upload button opens a page that has a Choose File button which opens a file chooser to select a backed up file and a eg sae Configuration to Upload field requiring the file path to upload icra Bacup Dowicad gh Confguon Und Lt Confguton and to restore it immediately Pressing Save will restore the selected backup file and delete all current user defined greetings and replace configuration settings P E a pera Configuration Management Restore previously backed up Configuration Upload The Restore to Factory De
307. ices can be configured The Go to Call Routing Table link leads to the Call Routing Table page where the call routing rules can be assigned to a certain T class of service s The classes defined in the Class of Go To Codec Settings Services page will appear on this page to assign the PBX extensions to a certain class of service s Go To User Settings PBX Conference extensions can be attached to a several class of services at the same time Fig II 38 Extensions Management Edit Entry Class of Service Settings page 8 Licensing This page is only available if the corresponding licensing is enabled from the Feature Keys page This group allows you to configure the extension to be used by the iQall application and the Pro Basic level Desktop Communication Console DCC E tanks Extensions Management Edit Entry O Go Back The page contains the following components License Settings 103 Enable DCC Pro license checkbox which allows you to set the hanced Setings Enable DCC Pro license corresponding extension to be used by the DCC Pro level A application When the checkbox is not selected on this page the DCC will be functional with the extension only during trial period W Enable iQall Toggling license Enable DCC Basic license checkbox which allows you to set the corresponding extension to be used by the DCC Basic level application When the checkbox is not selected on this page the Go To Line Set
308. ider and can be requested only for certain SIP servers For others the field should be left empty Send Keep alive Messages to Proxy enables the SIP registration server accessibility to the verification mechanism Timeout indicates the timeout between two attempts of SIP registration server accessibility verification If a reply is not received from the primary SIP server within this timeout the secondary SIP server will be contacted When the primary SIP server recovers SIP packets will continue to be sent to the server QX50 QX200 QX2000 SW Version 6 0 x 91 QX50 0X200 0X2000 Manual II Administrator s Guide A group of Host address and Port text fields respectively require the host address IP address or the host name the port number of the Outbound Proxy Secondary SIP Server and the Outbound Proxy for the Secondary SIP Server These settings are provided by the SIP servers providers and are used by QX IP PBX to reach the selected SIP servers VoIP Carrier Wizard Page 3 contains the following VoIP Carrier access code selection components The Access code text field requires a digit combination by dialing which the corresponding VoIP Carrier will be reached The Access code radio buttons allows you to create outbound routing rules Overview VolPCarfer CallRouting Call Recording NAT Traversa i i i i VolP Carrier Wizard e By prefix text field requires entering the prefix that will eee YEA be placed in front of the routing pa
309. ides this the details of the conference will 5 General Settings 500 be displayed in the Public Directories on the snom and Aastra aar a SIP phones Leave this checkbox unselected if the conference pss of Senice Sens E Show on Publie Directory is reserved or not used Conferences Management Edit Entry O Go Back Percentage of Total Memory 1 Y e The Percentage of System Memory drop down list is used to select the memory space in percents that can be used for storing conference recordings Fig I1 75 Edit Entry General Settings page The Edit Entry SIP Settings Edit Entry SIP Advanced Settings and Edit Entry Class of Service Settings pages are used to configure the conference s SIP basic registration advanced settings and assign the defined classes to the conference extensions respectively The descriptions of the settings can be found in the User Extension Settings section QX50 QX200 QX2000 SW Version 6 0 x 54 QX50 0X200 0X2000 Manual II Administrator s Guide e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide Activate is used to activate the selected conferences Terminate is used to stop the selected conferences Add Conference Add Conference tab opens the Conferences Management Add Entry page where a new conference can be created The page consists of the Conference ID text field that requires a unique ID for the call conference Please Note The length of
310. ies will list either the MAC address corresponding to ppp Stopped each network interface on the QX IP PBX Monitor includes links to survey LAN VLAN WAN For QX50 QX200 and PPP for QX50 QX200 traffic correspondingly The selection of these links will open the LAN and WAN Interface Statistics page with a table of network traffic statistics on the following selected interfaces Fig Il 237 Status Network Status page e Received Bytes e Transmitted Bytes e Received Packets e Transmitted Packets e Received Errors Transmitted Errors e Received Drop Errors e Transmitted Drop Errors e Received Overrun Errors e Transmitted Carrier Errors e Received MultiCast Packets Transmitted Collisions When opening the corresponding interface statistics window no traffic values are displayed at first After opening the window the tables will serve as a counter and traffic statistics will be updated every minute DNS Server Alternative DNS Server and Default Gateway these display the QX IP PBX settings corresponding to what has been configured with the System Configuration Wizard Services NTP Server and Client DHCP Server and Client DNS Firewall NAT PPP statuses shows if they have stopped or if they are still running Lines Status The Status Lines Status page shows the current status of each of the FXS IP and FXO lines or shared FXO ISDN E1T1 lines including details of the attached extension Since only one line of info
311. ime Call Duration Call destinations Each column heading in the tables is a link By clicking on the column heading the table will be sorted by the selected column Upon sorting ascending or descending arrows will be displayed close to the column heading The Details column available for the administrator is only present in Successful Calls table and provides the following information e Briefinformation about the call quality voice codec used to receive and transmit packets and the close call reason The close call reason appears to provide more information about the call termination reason which can be a network problem termination by one of the call parties voice mail service activation etc Clicking on the details information will open the RTP Statistics page where all RTP parameters of established call are provided e Authenticated By information details the callers that passed an authentication on the QX IP PBX as configured in the Local AAA Table e Information about FAX statistics for the calls that have a FAX transmission handled It only appears when there was a FAX transmission during the call Clicking on the FAX details link in the Details column will move to the FAX Statistics page The Call Detail column is present only in the Unsuccessful Calls table and indicates the reason why the call was unsuccessful The Filter performs a search procedure by the selected criteria The search may be done with several criteria at the same
312. ime measured in milliseconds If Si is the RTP timestamp from packet i and Ri is the time of arrival in RTP timestamp units for packet i then variance for packet i may be expressed as following V i Ri R1 Si 1 Ri Si R1 1 Rx Maximum Delay max V i 8 RX Delay Increase Count indicates the number of times the delay in jitter buffer is increased during the call RX Delay Decrease Count indicates the number of times the delay in jitter buffer is decreased during the call QX50 QX200 QX2000 SW Version 6 0 x 154 pya Please Note RTP Statistics is logged only when at least one of the call endpoints is located on the QX IP PBX For example it will not be logged when e calls incoming from or addressed to the IP lines or remote extension e calls from an external user are routed to another external user through QX IP PBX s routing rules In the first case RTP statistics will be logged if remote extension or IP line user is calling locally to the QX IP PBX s extension or auto attendant The Configure Call Quality Event Notification link leads to the Call Quality Notification page where call quality control notification specifics can be configured The Configure System Events link leads to the Event Settings page where the methods of notification for each system event can be configured FAX Statistics The FAX statistics page is accessed from the Call History page by clicking on the FAX
313. in the Traceroute Target text field Trace routing gives feedback on the routers passed by packets on the way toward the destination and the round trip delay of packets to these routers Fig II 268 Diagnostics Traceroute page Attention No Traceroute is possible if a high priority Firewall has been enabled see chapter Firewall and NAT For the purpose of tracerouting several IP packets are sent out UDP User Datagram Protocol is used to send packets and ICMP Internet Control Message Protocol is used to receive information about the routers In their headers the TTL Time To Live value increases from 1 to 30 When the first IP frame is received by the first router its IP address will be returned in its acknowledgement To Check the Internet connection 1 Specify the destination address for the ICMP request in the Ping Target text field Press the Start Ping button to process the ICMP request 2 3 Specify the destination address to trace the route 4 Press the Start Traceroute button to process the router triggering System Logs In the System Logs page you may view the generated logs on the QX IP PBX System logs are useful to determine any king of problems on the QX IP PBX as well as to monitor the user s access and the usage of it On the left side of the page a list of main logs is displayed Clicking on the needed link will display the most recent log lines The number of log lines displayed on this page i
314. including new and played Check Recordings New recordings 10 All recordings 10 recordings a n The Check Recordings functional link refreshes the recording ee C 31 Jul 2014 17 16 27 A box with any latest recordings if any S1 Jur2014 17166 e 31 Jul 2014 16 10 47 de The Recorded Conferences table displays all the recorded 31 Jul 2014 11 24 17 e Jul 2014 19 03 amp conferences with the following parameters A a 23 Jul 2014 17 33 18 de 23 Jul 2014 17 30 15 amp 16 Jul 2014 17 27 03 amp Fig II 296 Recorded Conference page Date amp Time shows the initiation date and time of the recorded conference Duration shows the duration of the recorded conference in minutes seconds Play by clicking on the speaker sign beside every record in the table the recorded conference will be played using the available media player supported by your Operatinig System The column headings of the Recorded Conferences table are organized as links By clicking on the column heading the table will be sorted by the selected column Upon sorting ascending or descending arrows will appear next to the column heading Each row in the table of Recorded Conferences can be selected by the checkbox for deletion To Play a Conference 1 Click on the speaker sign of the corresponding recorded conference 2 Depending on you browser settings the wav file will be played directly or an application will ask you to save the wav file
315. ined Polling Interval 1 hour Keep alive interval 120 seconds NAT IP checking interval 300 seconds No entries in NAT Exclusion table Properties for all Codecs except iLBC G 722 G 722 1 TDVC Packetization 20ms Silence Suppression yes iLBC properties Packetization 30ms Silence Suppression yes G 722 G 722 1 TDVC properties undefined G 726 Standard ITU T specification RTP RTCP port range 6000 6255 RTCP Support disabled UDP and TCP Port 5060 TLS Port empty Realm epygi Session Timer disabled DNS Server for SIP default SIP timers RFC 3261 Host Aliases for SIP undefined Voice Mail Recording G729a Email Subject for voice Voice mail received from VM_DISPNAME VM_USERNAME FAX to E mail format TIFF Undefined FXS lines Transmit Gain 6 Receive Gain 0 FXO lines Transmit Gain 0 Receive Gain 6 Voice Mail Recording Gain 0 Playback Gain 0 Audio Lines Transmit Gain Line out Off Receive Gain Line in Off Voice Mail Recording Gain 0 Playback Gain 0 Secure Connection disabled Request Timeout 10 Feature Key not added WAN Port not opened RADIUS client disabled 4 seconds Disabled 180 pya Parameter Firewall for QX50 0X200 Firewall for QX2000 Filtering Rules SIP IDS Settings IP Routing Configuration DHCP Advanced Settings DNS Server Settings Dynamic DNS SNMP Settings VLAN Settings IPSec PPTP and L2TP availab
316. ings RADIUS Remote Authentication Dial In User Service specifies the RADIUS protocol used for authentication authorization and accounting to differentiate to secure and to account for the users The RADIUS Server provides the option for a caller from through QX IP PBX to pass authentication and to be able to dial a specific number QX50 QX200 QX2000 SW Version 6 0 x 115 QX50 0X200 0X2000 Manual II Administrator s Guide When a RADIUS client is enabled on the QX IP PBX and according to the configuration of AAA Required option the RADIUS server will be used to authenticate user and or to account for the call This can be accomplished by automatic detection of the caller s number or a customized login prompt where the caller is expected to enter a username and password Transactions between the client and the RADIUS server are authenticated through the use of a shared Secret Key which is never sent over the network In addition user passwords are encrypted when sent between the client and RADIUS server to eliminate the possibility of a party viewing an unsecured network where they could determine a user s password If no response from the RADIUS Server is returned after the Receive Timeout expires the request is resent numerous times as defined in the Retry Count list The client can also forward requests to an alternate server s if the primary server is down or unreachable An alternate server can be used after a number of failed tries
317. ings Telephony VoIP Carrier Common Settings VolP Carrier Advanced Settings Account Name 25527423523 Y Use RTP Proxy The Password text field requires a password for ee Authentication authentication on the defined SIP server i ia Password The Confirm Password text field requires a password SiP serve sip provdercom Timeout 50 Pm confirmation If the input is not corresponding to the one in the SIP server Port 5489 aioe Extension Password field the error message Incorrect Host Address Password confirm will appear The SIP Server text field requires an IP address or the hostname of the SIP server destination party it is registered on The SIP Server Port text field requires the port number of the SIP server destination party it is registered on Port Secondary SIP Server Host Address Port Outbound Proxy for Secondary SIP Server Host Address Port Previous Fig l 135 VoIP Carrier Wizard page 2 2 VoIP Carrier Advanced Settings The Use RTP Proxy checkbox is applicable only when a route is used for calls towards a configured VoIP Carrier from a peer located outside the QX IP PBX When this checkbox is selected the RTP streams between external users will be routed through QX IP PBX When the checkbox is not selected RTP packets will move directly between peers User Name requires an identification parameter to reach the SIP server It should have been provided by the SIP service prov
318. ings page contains a list of all configured traphosts with the referring information Add functional button is used to add a new traphost to the table and opens Add SNMP Traphost page where the new traphost might be defined Page consists of the following components Traphost text field requires an IP address or the host name of the traphost Administrating application s host address should be inserted here Community text field requires community description public private etc for the administrating application to accept the notifications about the certain events on the QX IP PBX Field may contain some kind of password which should be the same both on QX IP PBX and on the administrating application for successful SNMP management A group of radio buttons is used to select the SNMP protocol version used for events notifications delivered by the QX IP PBX to the administrating application VLAN Configuration VLAN Settings page lists all existing virtual interfaces created on the QX IP PBX and allows you to create new interfaces Enable and Disable functional buttons are used to correspondingly enable and disable the selected virtual interface s Add functional button opens an Add Entry page where a new virtual network can be defined The page consists of the following components Enable checkbox is used to select whether the corresponding virtual interface will be enabled or disabled after it is created Interface Type ma
319. ion Configure some general settings for IP phones Create custom configuration templates for uploading to IP Phones Upload a custom logo for supported IP Phones Configure an FXS Gateway to add more analog FXS ports Configure FXS ports for connecting basic analog phones or fax machines Diagnostics for incoming calls to automatically pick up on the first ring and loopback the voice to the caller Configure FXO ports connected via analog telephone lines to the Public Switched Telephone Network PSTN Configure El or T1 trunks connected to the Public Switched Telephone Network PSTN Configure ISDN BRI trunks connected to the Public Switched Telephone Network PSTN Allows trunks lines of the E1 T1 ISDN BRI or FXO Gateways to be shared with the PBX Configure the unique authorization settings for each gateway providing the PBX with shared trunks lines Fig II 96 Interfaces Menu page 67 pya IP Lines QX50 0X200 0X2000 Manual II Administrator s Guide The IP Lines page is used to configure IP lines for IP phones to be connected to the QX IP PBX QX IP PBX provides the options to connect SIP phones to its LAN side assign the corresponding IP line to an active extension and use SIP phones as a simple phone with all telephony services of the QX IP PBX for example call hold waiting transfer etc By default 16 IP lines are available on QX50 24 IP lines are available for QX200 and 200 IP lines are
320. ion BECAUSE THE FREE SOFTWARE IS LICENSED FREE OF CHARGE THERE IS ABSOLUTELY NO WARRANTY Please make sure you download the GNU license from www gnu org For a list of free software go to http www epygi com about free software list QX50 QX200 QX2000 SW Version 6 0 x 195
321. ion Distribution field indicates IP phone s subscriptions distribution among BLF Busy Lamp Field subscriptions which are used for watching extensions on IP phones and MWI Message Waiting Indication subscriptions which are used for voice mailbox status indication on the phone License Status The License Status page displays a table with all available licenses on the QX IP PBX and the corresponding settings for each license Currently only iQall and DCC Pro Basic Level license statuses are displayed This page includes the following information Status License Status Type indicates the type of the license available on the QX IP PBX E Type ul Extensions Count indicates the number of the corresponding licenses available on the 2 uone a E O Status DCC Basic License 99 QX IP PBX N iQall Toggling License 99 In Use indicates the number of used licensed from the total available licenses Extension lists the extensions that are using the corresponding license Links in this column move to the corresponding service configuration page for the extension Fig II 243 Status License Status page QX50 QX200 QX2000 SW Version 6 0 x 148 epygl Events System Events The System Events page lists information about system events that have occurred on QX IP PBX When a new event takes place a record is added to the System Event table For failure events priority 2 and 3 see below the warning Please check your pending even
322. ion enabled will be dialed Send Mail 5 Weekly Allow Participants to Join 1 mins before Conference Activation Send Mail on Behalf of Save Fig II 305 Conference Settings Schedule page Clicking the Add button takes you to the Add Entry page where new scheduling rule can be configured This page offers the following components A group of radio buttons that are used for selecting the frequency of the Confer scheduled conference WB conference Gs Schedule Add Entry e Once the calendar date month day year should be specified Conference ID 888 for this option e Daily e Weekly weekdays when scheduling out to be activates should be selected for this option Use Select All and Select None to select or deselect all weekdays Select All Select None e Monthly the calendar day should be selected for this option e Annually the calendar day and the month should be selected for this option In the Time text fields the time of the scheduled conference activation should be defined The time selected in these fields will be considered according to the Date and Time Settings The Allow Participants to join conference before Conference Activation checkbox selection allows participants to dial in to the conference before conference activation Fig IT 306 Conference Settings Schedule Add Entry page QX50 QX200 QX2000 SW Version 6 0 x 192 QX50 0X200 0X2000 Manual II Administrator
323. ion prohibits a Ping operation toward QX IP PBX from its WAN The Fool Portscanner checkbox available only for QX50 QX200 selection prohibits QX IP PBX portscanning from its WAN As a reply to a Portscanning operation network unreachable or host unreachable feedback messages will be sent Fig I1 179 Advanced Firewall Settings page IDS Log The IDS logging page available only for QX50 QX200 contains information about dropped packets and the senders responsible for those packets IDS discards dangerous packets or packets including intrusion attacks It generates a table with the IDS log report The administrator can be notified about newly logged entries in various ways mail display notification Flashing LED sms depending on the settings in the Event Settings page To make an IDS log reporting table IDS needs to be enabled on the Firewall and NAT page QX50 QX200 QX2000 SW Version 6 0 x 119 e C Pyg l QX50 QXK200 QX2000 Manual II Administrator s Guide The IDS Logs table is a list of new or read IDS entries and descriptions referring to them The table provides a status row that has the value a M New if the entry is still unread or it is empty if the entry has already Dashboard ER E been read i IDS Logs P This ae to be the first time you are viewing this log Mark All aS Read marks all IDS logged entries aS read and removes i Matkal as Red v l set the te ig eet oi AAN new entries irewa or secur
324. iously added groups that may also be added as a member to another group Member description text fields can be used to enter an optional description ofthe member Fig II 196 IP Pool Group Configuration Add Member To Add a new Group with Members 1 Click on the Add button on the IP Pool Configuration page A page where a new group may be added will appear in the browser window 2 Define a group name in the Group Name text field and fill in the Group Description if needed 3 To add a group with the given parameters press Save 4 Open the IP Pool Group Configuration page by clicking on the group name 5 Select the Add button on the IP Pool Group Configuration page A page opens where new members may be added to the group 6 Enter an IP address for the member in the IP Address text fields select a IP subnet or IP group from the User defined Group drop down list to assign it to the currently selected group 7 Enter a Member Description in the corresponding text field if needed 8 To adda member with these parameters to the selected group press Save To Delete a Member 1 Check one or more checkboxes of the corresponding members that should be deleted from the Members table 2 Press the Delete button on the IP Pool Group Configuration Members page 3 Confirm the deletion by pressing on Yes or cancel the deletion by pressing on No To Delete a Group 1 Check one or more checkboxes of the corresponding groups that shoul
325. is complete and starts calling the dialed number Only predefined values included in the ae drop down list can be used for this setting i ieee lt Voice Mail RTP Streaming Channels Gain Control 3PCC Radius Client Call Quality Notification Dial Timeout Settings The Routing Dial Timeout setting will also be applied to all the supported IP phones that are auto configured with the QX IP PBX and provide the possibility of changing this setting through the auto configuration file The modified value of the setting will take effect after rebooting the IP phones Fig II 175 Dial Plan Settings page Call Quality Notification From the Configure Call Quality Event Notification page you may configure event notification policy when the call quality is lower than the allowed level This page consists of a Notify checkbox which enables the call quality monitoring mechanism for the corresponding event notifications and a Call Quality less than drop down list where the least satisfactory call quality should be selected When a call with the quality less than the level selected here is registered on the QX IP ace PBX an event notification will appear When the Notify checkbox is wos disabled no Call Quality events will occur on the QX IP PBX Configure Call Quality Event Notification Notify when Call Quality less than satisfactory Y Please Note The ways of notification for the Call Quality events should be configured fro
326. is named Modified Destination Host if the Pattern field on the first page of this wizard contains O symbol Destination Port requires the port number of the destination or of the SIP server This field is named Modified Destination Port if the Pattern field on the first page of this wizard contains O symbol User Name and Password require the identification settings for the public SIP server or servers requiring authentication Enable Activity Timeout checkbox is used to limit time to live period of routing pattern makes sense if accept or failure feedback arrives too late from the destination Checkbox selection enables the Activity Timeout text field which is used to insert a routing pattern activity timeout in the range from 1 to 180 seconds When timeout is configured the routing pattern will be active within the defined time frame and if no response has been received from the destination during that period the pattern will be stopped and next routing rule might be optionally considered depending on the Fail Reason configuration on the corresponding pattern The Restrict the Number of Simultaneous Calls checkbox is only available for IP PSTN destination type and is used to restrict the number of simultaneous calls to the public SIP server with the same username at the same time This checkbox enables Allowed Call Count text field which requires the number of simultaneous calls allowed in a range from 1 to 64 If you leave this fie
327. ith the name phone number of the last caller Extension Status QX IP PBX provides the possibility of controlling and determining the actual state of the managers phones through the receptionist s IP phone configuration of the IP phone is done automatically by QX IP PBX through the Receptionist Phone Configuration Wizard A programmable key on the receptionist s IP phone that is assigned to the corresponding manager will blink when an incoming call to the manager s phone is currently ringing The key lamp will be ON when manager is on a call and will be OFF if the manager s phone is in the idle state The extension status can be watched viewed by the receptionist to determine the availability of managers for incoming call transfers to them Call Interception To use Call Interception service the managers phones watch option should be enabled and each manager should have a programmable key assigned on the receptionist s IP phone This is performed automatically by QX IP PBX through the Receptionist Phone Configuration Wizard When an incoming call addressed to the certain manager comes in the receptionist can see the corresponding programmable key blinking and the caller s ID on the phone s display The receptionist is able to intercept the incoming call by pressing the blinking key The caller will then be connected to the receptionist If the receptionist does not answer the call addressed to the manager and if the manager does
328. ity reasons this will N automatically Selecting Mark all as Read will also disable this warning the New status from the Status row of the IDS entries table Mark alas Read Delete tog Status A Description New Microsoft Windows Media Services nsiistog dll Extension to Internet Information Server IIS Buffer Overflow Delete Log is used to delete all entries from the IDS table A detailed log of the selected entry can be seen by clicking on the Description link of the corresponding entry in the IDS Entries table Fig II 180 IDS Log page The IDS Logs detailed page has a following preview Overiew Firewall Firing Rules The Issue Detailed Log table is a detailed list of new and read IDS IDS Logs entries The table contains a Status row that has the value New if the Microsoft Windows Media Services nsiislog dll Extension to Internet Information Server IIS Buffer Overflow entry is still unread or that is empty if the entry has already been read da A a http cve mitre org cgi bin cvename cgi name CAN 2003 0349 Status Date 8 Time a Wed Aug 6 18 36 15 2014 Proto TCP IP 192 168 74 185 Port 80 Wed Aug 6 18 36 10 2014 Proto TCP IP 192 168 74 185 Port 80 Wed Aug 6 18 36 07 2014 Proto TCP 1P 192 168 74 185 Port 80 Wed Aug 6 18 36 06 2014 Proto TCP 1P 192 168 74 185 Port 30 Wed Aug 6 18 36 05 2014 Proto TCP 1P 192 168 74 185 Port 80 Wed Aug 6 18 36 04 2014 Proto TCP 1P 192 168 74 185 Port 30 Wed Aug 6 18 36 03 2014 Pr
329. ivated only in the following two cases e Anattempt was made to call a non existent extension e Anattempt was made to call a number not matching with any Destination Number Pattern in the Call Routing table Attention If a file with the same name is uploaded for other options the previous file will be replaced QX50 QX200 QX2000 SW Version 6 0 x 47 e C Pyg l QX50 QXK200 QX2000 Manual Il Administrator s Guide The Submenus page consists of the following functional buttons Add opens the Edit Scenario Add menu page where a new Menu n E Eee Ge name may be defined 8 extensions Edit Scenario 00 Submenus O Go Back Main menu Submenus Edit opens the Edit Scenario page where a newly created submenu naa Ea Dent scenario settings might be adjusted ubl Fig II 65 Create scenario Submenus page ons Add Extension Add Multiple Extensions Bulk Import Edit Scenario 00 Add menu O Go Back Extensions Menu name Subi Fig II 66 Submenus Add Entry Edit Scenario page O The Edit Scenario link appears only if a new scenario has been created previously The Edit Scenario link opens the Edit Scenario page where a previously created scenario can be changed O The Import Export scenario link leads to the page where a new scenario file can be imported or exported The Import Export Scenario page offers the following components Import scenario is used for uploading the previously downloaded amp
330. kbox is selected the system will NOT use timeslots blocked by the carrier If the checkbox is clear the system will try to unblock the timeslots and will make outgoing calls if succeeded QX50 QX200 QX2000 SW Version 6 0 x 80 e C Pyg l QX50 QXK200 QX2000 Manual II Administrator s Guide Group B Support manipulation radio button group is present only when R2 signaling selected from Signaling Type drop down list on the previous page is different from R2 DTMF and is used to enable disable the Group B Support The Group B Support manipulation radio button group offers following selections e Enable this selection enables Group B Support both for answer and busy recognitions of transmit and receive signals This selection requires you to define transmit and receive signals The Transmit Answer CAS Signaling Wizard Signal and Transmit Busy Signal parameters are defined from the drop down lists on this page When transmit signals are selected press Next on this page E1 T1 Trunk Pe Interfaces R2 Receive Signal Settings Trunk 1 192 168 74 127 5060 to access the R2 Receive Signal Settings page where Selected Tmests 1 23 4567 8 9 10 11 12 13 1415 17 1819 20 21 222324 25262728 29 203 Receive Answer Signal and Receive Busy Signal ci n should be defined Use the checkboxes to select the Enabie Bt Receive Answer Signal and Receive Busy Signal sees values Multiple values are allowed for each signal Enable B4 Please Note Warnin
331. king the extension Confirm Password requires a password confirmation If the input is not corresponding to the one in the Extension Password field the Incorrect Password confirm error will appear GUI Login Allowed checkbox enables the current extension to be used to access the QX IP PBX via WEB interface by extension name and password With the Show on Public Directory checkbox enabled the details of the corresponding extension will be displayed in the User Settings table on the Main Page of the Extension s Web Management accessed by the extension s login see Manual III Extension User s Guide Besides this the details of the extension will be displayed in the Public Directories on the Snom and Aastra SIP phones Leave this checkbox unselected if the extension is reserved or not used or when the extension serves as an intermediate unit for call forwarding etc The Percentage of Total Memory drop down list allows you to select the space for call recordings and the uploaded custom messages of Recording Box extension The maximum value in the drop down list is equal to the maximum available space for voice messages on QX IP PBX 2 Recording Box Settings This group contains Recording Box settings and has the following confer C O mp 0 ne nts i Add Multiple Extensions Bulk Import ean Extensions Management Edit Entry Ask Password on Local Access checkbox selection enables the Go Back password protection for local
332. l 192 168 25 50 9122 Remote 192 168 74 206 6006 Quality 1 excellent Fig II 251 RTP Statistics page The Local and Remote fields indicate the two peers between which the RTP stream is transmitted The characteristics in the table below describes to the piece of RTP stream between these peers Rx Tx Codec codec for received and transmitted RTP stream respectively Rx Tx Packets number of RTP packets received and transmitted respectively Rx Tx Packet Size size of RTP packet payload received and transmitted respectively Rx Lost Packets number of lost RTP packets for received stream Rx Jitter inter arrival jitter is an estimate of the statistical variance of the RTP data packet inter arrival time measured in timestamp units The inter arrival jitter is defined to be the mean deviation smoothed absolute value of the difference D in packet spacing at the receiver compared to the sender for a pair of packets If Si is the RTP timestamp from packet i and Ri is the time of arrival in RTP timestamp units for packet i then for two packets i and j D may be expressed as D i j Rj Ri Sj Si Rj Sj Ri Si JO JG 1 J DG 1Li JGi 1 16 where J i is Rx Jitter for packet i For more details about Jitter calculations please refer to the RFC1889 Rx Maximum Delay maximum variance absolute value of actual arrival time of the RTP data packet compared to estimated arrival t
333. ld empty no limitation will apply to the number of simultaneous logons The Use RTP Proxy checkbox is available for SIP and IP PSTN destination types and is applicable when a route is used for calls through QX IP PBX between peers that are both located outside the QX IP PBX When this checkbox is selected RTP streams between external users will be routed through QX IP PBX When the checkbox is not selected RTP packets will move directly between peers The Collect Call checkbox is available only for E1 T1 destination type and is used when it is simply preferable for the called phone to pay for the call This service is applicabe only if the Collect Call checkbox is enabled on both calling and called party s IP PBXs The Single Call Duration Limit checkbox is available for SIP IP PSTN and PSTN destination types and is used to limit the duration of the call placed with the selected routing rule If this checkbox is not selected the call duration will be unlimited This checkbox selection enables the Maximum Duration text QX50 QX200 QX2000 SW Version 6 0 x 95 QX50 0X200 0X2000 Manual II Administrator s Guide field where the maximum duration of the call in seconds should be defined Once the call duration reaches the value defined here the call will be disconnected without prior notice The Play audible signal before Intercom activation checkbox is appeared only if PBX Intercom is selected as Destination Type see Manual III Extension U
334. le only for QX50 QX200 Event Settings Call History Conference History System Logs Settings Remote Logs Settings User Rights Management QX50 QX200 QX2000 SW Version 6 0 x QX50 0X200 0X2000 Manual II Administrator s Guide System Default Value Enable Firewall disabled Enable IDS enabled Enable NAT enabled Ping Stealth enabled Fool Portscanner disabled Enable Firewall disabled Ping Stealth enabled Outgoing Traffic MS File Sharing Blocked for all SIP Access Allowed for all Enable SIP IDS enabled Add the IP address into the Blocked IP list in Firewall enabled Discard SIP messages from IP address enabled No Routes DHCP Options Gateways 172 30 0 1 for QX50 QX200 192 168 0 200 for QX2000 Subnet mask 255 255 0 0 Domain name servers 172 30 0 1 for QX50 QX200 192 168 0 200 for QX2000 NBT name servers 0 0 0 0 NTP servers 172 30 0 1 for QX50 QX200 192 168 0 200 for QX2000 Domain name epygi config loc Overload tftp server name 172 30 0 1 for QX50 QX200 192 168 0 200 for QX2000 DHCP Server Statements Authoritative enabled Ping Check enabled Ping timeout 1 sec Time to live TTL 86400 seconds Mail Exchange MX undefined No aliases defined Disabled SNMP disabled Undefined No connections RSA Key Management 1024 bit key defined PPTP Server Configuration Subnet 172 31 1 0 24 Authentication MSCHAPv2 MPEE 128 bit L2T
335. le to join the conference until the moderator has logged in New Participant Confirmation Type field should also be set to Password to enable this option Fig IT 303 Conference Settings New Participants Configuration page QX50 QX200 QX2000 SW Version 6 0 x 191 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide Selecting the New Participant Indication checkbox will enable a beep indication during the active conference when a new user joins or leaves the conference Handset Added Participants Configuration This page is used to configure the settings of participants dialed out from the handset by the moderator during the active conference Once the handset added participant connects the conference he will automatically appear in the Conference Progress table and remain there unless the conference is terminated This will allow the handset dialed participant to hang up and dial in to the corresponding conference again while it is active The page consists of the following components Participant Type drop down list is used to select the state speaker or listener only of the handset added participants connected to the conference Confirmation Type drop down list is used to select whether the po Conference General Recording Customization Schedule Send Notification Mail conference is password protected for the handset added users or not Handset Added Participants Configuration When Star selection is chosen the
336. lement The B1 Channel and B2 Channel checkboxes enables disables timeslots for voice transfer Disabling the timeslot will prevent both incoming and outgoing calls The ISDN Stats are not available in shared mode External PSTN Gateways The External PSTN Gateways page allows QX IP PBX to use the PSTN lines FXO lines E1 T1 and or ISDN trunks on other QX This provides the option to call not only through local PSTN lines but also through available shared FXO E1 T1 or ISDN lines in the network of QXs When the sharing mode is enabled and one QX IP PBX is configured to use the shared PSTN lines of another QX the corresponding routing patterns will automatically be created in the Call Routing Tables see Call Routing Table on both QXs This will allow PSTN call routing between the two QXs The Use PSTN lines of the other device checkbox is used to enable QX IP PBX to use the shared PSTN lines on a remote device This selection requires you to configure the Authorization e Parameters El T1 Trunk ISDN Trunk PSTN Gateways External PSTN Gateways de Interfaces Y Use PSTN lines of the other device Fig II 131 External PSTN Gateways page Authorization Parameters Overview d es FXS XO ISDN Trunk PSTN Gateways The Authorization Parameters page is used to create accounts External PSTN Gateways Authorization Parameters for the remote QX Gateway allowing them to connect the QX and PE SIE o EI share the available PSTN lines The
337. lendar day should be selected for this option o The Renewal Amount text field requires the renewal amount in minutes to be added to the Available Calls Duration when the expiration date of the Available Calls Duration is reached o The Discard remainder before renewal option selection allows to discard the remainder of Available Calls Duration before renewal and set the Renewal Amount as an available calls duration o The Specific Date selection provides a possibility to manually define the expiration date allocated for the Available Calls Duration for the selected routing rule When the Specific Date expires the selected routing rule becomes unavailable automatically and no new call will be possible until this field is updated QX50 QX200 QX2000 SW Version 6 0 x 98 The Call Routing Wizard Tracing Debug Options page appears if the Set Tracing Debug Options on This Rule checkbox was previously enabled on Page 1 of the Local Call Routing Wizard It will require information about the tracing debug options This page offers result options of the corresponding routing rule execution when the notification event will be printed in the Events page In Case of Successful Call a notification event is printed when the successful call was established with the routing rule e In Case of Failover a notification event is printed when the call ends up on one of the failover reasons selected on the Page 2 of the Local Call
338. les View All Incoming Fonwarding Outgoing Management Access Call Control Access SIP Access Blocked IPs Allowed IPs Filtering Rules Allow or deny access from your LAN to the Internet Firewall Enable Disable Add Edit Delete Restricted State Service Action IP Description Enabled User QuadroFXS26GW Blocked Any Fig I1 184 Filtering Rules page Filtering Rules View All Incoming Forwarding Outgoing Call Control Access SIP Access Blocked IPs Allowed IPs Filtering Rules Here you can allow hosts from the Internet to reach the Web interface of this device If you are having doubts leave this empty Y Firewall Enable Disable Add Edit Delete State Service Action Restricted IP Description Enabled HTTPS Allowed Any Fig II 185 Filtering Rules page Filtering Rules View All Incoming Forwarding Outgoing Management Access Call Control Access SIP Access Blocked IPs Allowed IPs Filtering Rules Here you can allow hosts from the Internet to reach the Call Control interface on this device create and manage calls If you are having doubts leave this empty Enable Disable Add Edit Delete State Service Action Restricted IP Description Enabled CCA Blocked 192 168 74 25 Fig II 186 Filtering Rules page 121 pya SIP Access SIP Access is used to allow or deny the SIP access to or from the particular SIP servers SIP hosts or a group of them The SIP Access filtering rule
339. licable on the board An LED notification may appear depending on the notification type given in the Event Settings page when a new event occurs Numerous circumstances may cause a certain application on QX IP PBX to flag an event Event Settings The Event Settings page lists all possible events on the QX IP PBX and allows controlling notification action when an event takes place Each entry in the events table has a checkbox assigned to each row By selecting the corresponding checkboxes operations such as Edit may be done for one or more events Edit opens the Edit Event Settings page to modify the event action QX50 0X200 0X2000 SW Version 6 0 x 35009 T 3 m Y SA E z y a System Events Current System Time Sat Jan 1 02 05 14 2000 Delete Mark all as read Reset LED Status Timestamp Priority Application Event Settings Edit Application SYSTEM SYSTEM wy SYSTEM SYSTEM SYSTEM Ppp MAIL IDS O ppp Y SYSTEM PNPIPPHONES PNPIPPHONES Fig II 244 System Events list Description the device has been successfully started after reboot Default configuration has been created the rollback mechanism restored the old system configuration Could not add ip route DynDNS Event The PPP daemon got an error could not send a mail SNTP daemon corrected the system time SNTP daemon could not reach the time server possible intrusion detected password or user is wrong Sy
340. ll be used instead For virtual extensions the incoming FAX can only be stored in the extension s voice mailbox To allow FAX to be stored in the voice mailbox the extension s user should not answer the incoming calls so that they are forwarded to the voice mailbox Please Note If both of the above checkboxes are disabled no FAX transmission to the peer s voice mailbox will be possible Enable Pass Through Modem checkbox is available for the Auto Attendant and the extensions attached to the FXS lines only This checkbox enables the modem tone detection and the G 711 codec support for the data transmission from to the modem attached to the line During data transmission Silence Suppression and Echo Cancellation are automatically disabled on the line Please Note If the extension attendant is intended to accept modem connections disable the Enable T 38 FAX checkbox to allow the system to identify the modem tones correctly Otherwise the modem connection may fail Force Self Codecs Preference for Inbound Calls checkbox enables the usage of your own preferred codecs if available on both peers Secure RTP Settings are used to configure secure voice over IP communication on the QX IP PBX The SRTP Policy drop down list is used to select the secure IP connection policy For IP phones the following options are available e Make and accept only secure calls only the secure calls will be generated and accepted e Make and accept only unsecure calls
341. ll display an IP Pool Group Configuration page with the Members list for the current group Fig II 194 IP Pool configuration Add Group page QX50 QX200 QX2000 SW Version 6 0 x 124 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide The IP Pool Group Configuration page displays a list of all the added member IP addresses for the selected group It offers the following components Current Group provides read only information about the current IP Pool Group Configuration r name the members are listed for l ee Current Group EpygiGroup Add Edit Delete Add opens the Add Member page where a new member may be poa _ added eit Edit opens the Edit Members page where the service parameters can be modified This page includes the same components as the Add Member page To operate with Edit only one record may be selected otherwise the error message One row must be selected will appear Fig II 195 IP Pool Group Configuration page The Add Members page provides the following radio buttons ey cates en he ene an nee i Current Group EpygiGroup IPaddress requires the member IP address that is to be added to the group Y Firewall IP Subnet requires the subnet specified by the IP address and the Maskbits See above for more information about Maskbits URL Address requires the member hostname to be added to the group Member description Sales Group The User defined Group includes prev
342. ll particularly allow you to reach users over IP PSTN providers or to call to the peers registered on the certain SIP servers by dialing simple digit combinations For each configured VoIP carrier the wizard creates a specific IP PSTN routing rule in the Call Routing Table This entry is available to PBX users only which means only PBX users can make calls to the corresponding VoIP carrier Additionally a virtual extension automatically generated in Extensions Management will be registered on the defined VoIP Carrier s SIP server The settings of that extension will be used to make calls from QX IP PBX s users towards the created VoIP Carrier will be placed VoIP Carrier Wizard Page 1 provides a following option of describing the VoIP carrier VoIP Carrier Wizard Select VoIP Carrier When predefined carrier is selected in the VoIP Carrier drop X Telephony down list the SIP Server and Port will be already predefined in VIP Carrier Manual the next page Manual selection allows you to manually set up gt the VoIP Carrier settings The Description field allows you to insert an optional description of the VoIP Carrier Fig l 134 VoIP Carrier Wizard page 1 VoIP Carrier Wizard Page 2 is used to define VoIP Carrier Settings The page contains following components ne N 1 VoIP Carrier Common Settings The Account Name text field requires a username for authentication on the defined SIP server e VoIP Carrier Sett
343. load All Call Detail Records link is used to download the entire displayed statistics in a file that can be viewed with a simple text editor This type of conference Call History file is easy to read and can be displayed in a spreadsheet Fig II 255 Conference History Settings page The Download All Call Detail Records in CSV format link is used to download the entire displayed conference Call History in a CSV Comma Separated Values formatted file The Clear all Records button is used to clear all conference Call History records When the number of Conference Call History entries exceeds the numbers specified in the Conference History Settings page the oldest entries are being automatically deleted LAN WAN LAN and WAN Interface Statistics The LAN and WAN Interface Statistics pages display the LAN and WAN statistics LAN Interface Statistics page is not available for LAN WAN Qx2000 The table displayed here shows the number of receive LAN Interface Statistics and transmit events that occurred since the last resetting of the Started at 21 Aug 2014 14 54 05 Time difference 8 sec counters by pressing the Clear button Received Bytes Transmitted Bytes Received Packets Transmitted Packets Receive Errors Transmit Errors Receive Drop Errors Transmit Drop Errors Receive Overrun Errors Transmit Carrier Errors Receive Multicast Transmit Collisions Refresh Clear Fig IT
344. load an announcement The uploaded file needs to be in PCMU CCITT u law 8 kHz 8 bit Mono wave format otherwise the system will prevent uploading and the Invalid audio file or format is not supported warning message will appear The system also prevents uploading if there is not enough memory available for the corresponding extension This will cause the You do not have enough space warning message to appear Choose File opens the file chooser window to browse for a new announcement The Download Ringing Announcement and Remove Ringing Announcement links appear only if a file has been uploaded previously The Download Ringing Announcement link is used to download the announcement file to the PC and opens the file chooser window where the saving location may be specified The Remove Ringing Announcement link is used to restore the default ring back tones e RTP Channel selection is used to define the channel for the broadcast streaming The RTP channels are created by the system administrator Therefore if you are experiencing problems with using the RTP channels as ringing announcement or no RTP channels are available to select on this page turn to your system administrator for clarification s Audio Line In available only for QX50 QX200 selection uses the external radio broadcasting or any other audio resource as the hold music When selecting this option check with your system administrator if there is an external audio resource con
345. loading if there is not enough memory available for the corresponding conference and the You do not have enough space warning message will appear Choose File opens the file chooser window to browse for a new hold music file The Download Hold Music File and Remove Hold Music File links appear only if a file has been uploaded previously The Download Hold Music File link is used to download the hold music file to the PC and opens the file chooser window where the saving location may be specified The Remove Hold Music File link is used to restore the default hold music QX50 QX200 QX2000 SW Version 6 0 x 189 QX50 0X200 0X2000 Manual II Administrator s Guide Info File parameters group allows you to upload a text file with some conference related announcement advertisement or any other information to be displayed on the Conference Progress page The group offers the following components Upload Info file indicates the information file name The system will display the file content exactly in the way it is formatted in the file It is recommended to use a txt formatted plain text file The uploaded file should not exceed the size of 2000 bytes The system also prevents uploading if there is not enough memory available for the corresponding conference and the You do not have enough space warning message will appear Browse opens the file chooser window to browse for an information file The Remove Info File link appears only wh
346. locally to QX IP PBX i e it can use QX IP PBX s PBX features place and receive calls access voice mails etc The Enable checkbox activates the SIP Remote Extension s functionality Please Note SIP Remote Extension functionality may be enabled only for active attached to an onboard FXS or IP line extensions Identification parameters used by the remote SIP device for registration on the QX IP PBX should be defined in the Username and Password text fields They should match on both QX IP PBX and SIP phone for a successful connection The Password field is checked against its strength and you may see how strong is your inserted password right below that field To achieve the well protected strong password minimum 8 characters of letters in upper and lower case symbols and numbers should be used If you are unable to define a strong password press Generate Password to use one of system defined strong passwords Line Appearance text field requires a number of simultaneous calls supported by the SIP phone When the Enable RTP Proxy checkbox is selected incoming and outgoing RTP streams to and from the remote SIP phone will be routed through QX IP PBX When the checkbox is not selected RTP packets will be moving directly between peers QX50 QX200 QX2000 SW Version 6 0 x 28 QX50 0X200 0X2000 Manual II Administrator s Guide QX50 0X200 0X2000 Manual II Administrator s Guide When the Fallback To Local Extension When Not Regist
347. locally to the PC If you need to save the file please specify the path then run the media file from the specified location To Delete a Recorded Conference 1 Select the checkbox of the corresponding record s in the Recorded Conferences table that will be deleted 2 Select the Delete button 3 Confirm the deletion clicking Yes The selected conference then will be deleted To abort the deletion and keep the conference on the QX IP PBX select No Conference Settings General Settings The General Settings page is used to configure the basic conference settings Conference Recording Customization Participants Schedule Send Notification Mail The page contains the following components General Settings Conference ID 888 Conference ID indicates the unique ID of the conference escription Daily Conference Description indicates any descriptive information about the ia sos conference Participant Password seeeee Max Duration minute 45 Moderator Password text field requires a password for the Play Hold Music Until Moderator is Connected Automatic Speaker Detection moderator access to the conference The password inserted here should be used by the moderator to join the conference Moderator is able to use conference codes during the active call conference as well as to access conference specific GUI pages and coordinate the conference view change conference properties activate deactivate it start stop resume
348. lp you manage the settings for group of IP phones which PE Interfaces Add Edit Delete saves your time and ensures consistency Template Name Description systemdefault System default configuration template Template Office Template for Office IP phones This page allows you to adjust the IP phone s template general settings and define options for advanced configuration of the IP phones models which can be common for group of IP phones Template Marketing Template for Marketing Department IP phones The subpages for each supported IP phone model allows you to define a set of extensions mapped to keys on IP phones see Programmable Keys Configuration For Aastra models the General Settings page contains the following components Fig II 101 Manage IP Phone Templates page QX50 QX200 QX2000 SW Version 6 0 x 71 QX50 0X200 0X2000 Manual II Administrator s Guide e Local Dial Plan indicates the number and pattern of digits dialed by the user in order to reach a particular destination e Send Dial Plan Terminator is used to switch a dial plan terminator or timeout When the IP phone is configured to use a dial plan terminator such as the pound sign the phone waits for 4 or 5 seconds after the handset is picked up or a key is pressed to place a call Play a Ring Splash is used to switch a call waiting tone when there is an incoming call on the BLF Busy Lamp Field monitored extension If the host
349. ls either in the local Recording Box or upload them to the remote server From Call Recording Settings page the call recording can be configured to be started automatically once the call starts or to be started manually from Administrator s Main Page of the QX IP PBX s Web Management or by pressing the Record button on the IP phone during the call If no such button exists on IP phone the functional key can be configured from QX IP PBX to handle the recording functionality see Programmable Keys Configuration To configure Call Recording an extension of the Recording Box type should be created first The memory allocated to that extension will be used for storing the recorded calls There are two ways to access the recorded calls in the Recording Box through handset and through Web Management Through QX50 QX200 QX2000 SW Version 6 0 x 107 e C Pyg l QX50 0X200 0X2000 Manual II Administrator s Guide handset Recording Box is accessible by calling the Recording Box extension On QX IP PBX s Web Management call recordings are available from Extensions Management page by clicking on the Recording Box extension Attention Following limitations apply to the call recording on the QX IP PBX e Calls to Auto Attendant or Voicemail cannot be recorded The Call Recording Settings page is used for configuring the call recording rules It has two view modes the Basic View and Advanced View which can be switched by appropriate butto
350. ly shared ISDN trunks are available Use PSTN lines of the other device disabled Authorization Parameters undefined VoIP Carrier Manual Description Empty Call Routing table 3 entries defined for a call to the default Auto Attendant 00 for calls to PBX and SIP Route all incoming SIP calls to Call Routing disabled Local AAA Table Authentication by Caller ID enabled Undefined Enable Tunnels to Slave Devices disabled Tunnels to Slave Devices no entries Enable Tunnels to Master Devices disabled Tunnels to Master Devices no entries Disabled 179 Parameter Call Recording NAT Traversal Settings RTP Settings SIP Settings Voice Mail Common Settings RTP Streaming Channels Gain Control Settings for QX50 QX200 Gain Control Settings for QX2000 3PCC Settings RADIUS Client Settings Dial Timeout Call Quality Notification QX50 QX200 QX2000 SW Version 6 0 x QX50 0X200 0X2000 Manual II Administrator s Guide System Default Value Basic View All extensions are disabled Advanced View Call Type Auto Address empty Recording Type Always start automatically Max Recording Duration 1 hour Recording To same as extension number Description empty NAT Traversal for SIP Automatic SIP and RTP Parameters Use STUN SIP TCP Port 5060 STUN Parameters Primary STUN Server stun epygi com Primary STUN Port 3478 Secondary STUN Server undefined Secondary STUN Port undef
351. m the Action list that is used in the rule If the list has a default value do not change the default values Enter the IP address in the Forward to IP field if an Incoming Traffic Rule is to be added Choose the restriction type by selecting Any Single IP IP Mask or Single URL and enter the required information in the text fields or select a group Insert a Description if needed To add arule with these parameters press Save To Delete Filtering Rules 1 La 3 4 Select the corresponding Filtering Rule Incoming Traffic Port Forwarding Outgoing Traffic Management Access Call Control Access SIP Access Blocked IP List or Allowed IP List Check one or more checkboxes of the corresponding rules that should be deleted from the rules table Press the Delete button on the Filtering Rules page Confirm the deletion by clicking on Yes or cancel by clicking on No Custom Services Service Pool Configuration The Service Pool table is a list of all created services and their parameters It is used to add new services with the appropriate settings protocol type and port range New services can be used to add a restriction or permission by defining a new filtering rule with the following l a Service Pool Configuration Add opens the Add New Service page where new services may be Add Edit Delete added Edit opens the Edit Service page where the service parameters except for the service name can be modified This
352. m the Events page Fig II 176 Configure Call Quality Event Notification page QX50 QX200 QX2000 SW Version 6 0 x 117 epy8l Firewall Menu The Firewall menu allows you to configure the following settings e Firewall Firewall and NAT Advanced Firewall Settings e Filtering Rules ini View All Filtering Rules Incoming Traffic Port Forwarding Outgoing Traffic Management Access Call Control Access SIP Access Blocked IPs Allowed IPs e Custom Services Service Pool Configuration e IP Groups IP Pool Configuration e SIP IDS Settings QX50 QX200 QX2000 SW Version 6 0 x Overview Firewal Overview Firewall Advanced IDS Log Filtering Rules View All Incoming Forwardin Outgoing Call Control Access SIP Access Blocked IPs Allowed IPs Custom Services Custom Services IP Groups IP Groups SIP IDS QX50 0X200 0X2000 Manual II Administrator s Guide Filtering Rules Custom Services IP Groups SIP IDS Enable NAT and firewall choose the protection level Enable device to respond to ping requests Intrusion Detection System IDS logs Monitor for suspicious network activity on the WAN port List of all defined firewall rules Forward external service or port number to internal IP address and port Allow or deny outgoing traffic from LAN to Internet Allow management access from specific hosts Create the list of hosts having acc
353. mation about events on the QX IP PBX QX50 QX200 QX2000 SW Version 6 0 x 163 QX50 0X200 0X2000 Manual II Administrator s Guide The Log Lines to Show drop down list is used to choose the maximum number of log lines to display on the System Logs page The Mark all Logs button is used to set a line marker in the logs If you need to follow a certain piece of log push this button to set a starting mark in all logs and then perform the needed actions over the QX IP PBX When the actions are done push this button again to set an ending mark in all logs This way you shall clearly see a piece of log between the staring and ending marks generated during the certain actions taken over the QX IP PBX The Comment text field is used to insert some text information which will be displayed next to the marks inserted in the logs This comment may describe the problem captured in the following logs and may be useful for the Technical Support The Download all Logs button is used to download all logs to the local PC as a tar archive file These logs can then be used by the Epygi Technical Support Office to determine the problem that has occurred on your QX IP PBX Remote Logs Settings The Remote Logs Settings page is used to adjust the system logging settings and contains the following components The Enable Remote Logging checkbox is used to enable remote monitoring of QX IP PBX s logs When this option is selected remote administra
354. may prevent or allow incoming or outgoing SIP calls to or from specified SIP server s or host s Blocked IPs When Blocked IP List is used traffic from specific hosts may be blocked no matter what services are opened in the other filters NO traffic will be allowed to the specified hosts The Blocked IP List service has a higher priority if the same host is also listed in the Allowed IP List table Allowed IPs Allowed IP List allows trusted hosts to reach your network and vice versa It is an exception to other rules and only all services may be allowed for a single host QX50 0X200 0X2000 Manual II Administrator s Guide Filtering Rules View All Incoming Forwarding Outgoing Management Access Call Control Access Blocked IPs Filtering Rules Allow or deny access to or from SIP servers and other SIP devices in the Internet A Firewall Enable Disable Add Edit Delete 2 State Service Action Restricted IP Description Enabled SIP Allowed Any Filtering Rules Custom Services IP Groups SIP IDS View All Incoming Forwarding Outgoing Management Access Call Control Access SIP Access Blocked IPs Filtering Rules This blocks access for special hosts Traffic to or from these hosts will be blocked in any case Y Firewall Enable Disable Add Edit Delete Allowed IPs Fig II 187 Filtering Rules page Allowed IPs State Service Action Restricted IP Description Disabled All Blocked 172 30 4 254 Blo
355. mber of Discarded Symbols and Prefix values Press the Next button If IP has been selected on the previous step in the Source Type drop down list then Source Host should be inserted in the current page If FXO ISDN or E1 T1 has been selected in the Source Type drop down list then the ISDN E1 T1 trunk or the FXO line number should be selected here If the Set Date Time Period s checkbox has been selected on the first page pressing Next will open the Date Time Rules page where route validity should be defined If the Set Overall Calling Time Limit checkbox has been selected on the first page pressing Next will open the Routing Overall Calls Limitation Settings page where the total call duration for all calls can be configured over a specific time frame for each Call Routing Entry If the Set Tracing Debug Options on This Rule checkbox has been selected on the first page pressing Next will open the Tracing Debug Options page where the tracing debug options should be defined If the Class of Service feature is enabled assign the defined classes to the selected routing rule Press the Finish button to establish a local route with the inserted settings To create a local AAA entry Serer r gt Click on the Local AAA Table tab on the Call Routing page Press the Add button on the Local AAA Table page Choose the Authentication type Enter the Phone Number Username and Password or the Authentication by PIN depending on the selected Auth
356. mbering Plan ISDN telephony numbering plan v e T310 Timer requires the value for the T310 timer in aii errno nen ae milliseconds numeric values from 1000 to 120000 It is eo TN responsible for the outgoing call steadiness when CALL n Generate Progress Tone to P PROCEEDING is already received from the destination but call confirmation ALERT CONNECT DISC or PROGRESS has not yet arrived Switch Type basic_dss1 v Enable CLIR Service 4 Alternative Disconnection Mode P Asserted Identity e Alert Guard Timeout requires the value for the Alert Guard Timer in milliseconds numeric values from 0 to 500 between CALL PROC and ALERT messages Alert Guard Timer itis used when QX is connected to a slow ISDN PBX Send Caling Party Subsddress Recommended values are eee fast connection Oms a em normal 150ms default slow ISDN PBX 350ms very slow ISDN PBX 500ms Override CLID with P Asserted identity Use Redirecting Number Info Element with P Fig IT 130 ISDN Wizard 12 amp L3 Settings The Coding Type drop down list allows you to select between a law and mu law coding types The Switch Type is another configuration parameter that depends on the Service Provider The Passive Mode checkbox is used to leave the ISDN Layer1 connection in the Slave mode When this checkbox is selected Layer1 remains idle when calls are not available When this checkbox is not selected QX keeps its Layer1 always a
357. me hostaddress port where hostaddress can be an IP address for example 192 168 90 10 or a host name e g sip epygi com The port number is optional for the SIP address If no port is specified 5060 will be used The range of valid ports is between 1024 and 65536 Please Note A direct call will be placed toward a participant s SIP address if the corresponding conference is registered on a different SIP server than the participant is registered on or if the participant is not registered on any SIP server E Conference Conference Settings General Recording Customization Schedule Send Notification Mail Participants Add Entry Conference ID 888 Participant Name John Smith SIP Address Tel Number 113690sip epygi loc Mail Address John Smith gmail com Participant Type Speaker Confirmation Type Star bd Y Allow Video Y Dial Out Y Activate On Dial In Y Participant Indication Y Nested Conference Y Allow Duplicated Participation Fig II 301 Conference Settings Participants Add Entry page The value will be implied as a Routing Number and will be parsed through the Call Routing table if it does not match the SIP URI syntax Email Address requires the email address of the participant Conference related notifications configured from the Schedule page or using the Send Notification Mail option will be sent automatically to this address This field is not available on this page when it is re
358. me to the IPSec Connection Press Next to go to the next page of the IPSec Connection wizard 3 Enter the remote side IP parameters check subnets gateways for the connection select the NAT traversal option if needed and the desired keying type Press Next to go to the next page of the IPSec Connection wizard 4 Ifthe Automatic Keying type has been selected enter the automatic keying parameters and select the PFS and IPSec compression options if needed If the Manual Keying type has been selected enter the encryption and authentication keys and SPI s 5 To specify an IPSec connection with these parameters press Finish To Manage an RSA key for the IPSec Connection 1 Press the RSA Key Management button on the IPSec Connection Settings page The IPSec Connection RSA Key will appear in the browser window 2 Select the RSA key length and press Generate to generate a new RSA public key This may take several seconds 3 Enter a destination e mail address in the Email this key to peer text field then press Send to send the new RSA public key To Delete Stop Start a PPTP L2TP Connection Select one or more checkboxes of the corresponding connections that should to be deleted stopped started from the Connections tables Click on the Delete Stop Start button from the table s menu to perform the corresponding operation for the selected PPTP L2TP connection s If deleting confirm it with pressing on Yes The PPTP L2TP connection will
359. month month when the call recording started e date_day day when the call recording started e extension recording box extension e hostname QX hostname Any combination of above variables can be used in the Naming Scheme text field along with the manually text inserted The following syntax applies Example MyQX caller_dispname duration time_hour time_min _business In case if the caller s display name was Andrew the call lasted 15 seconds and it took place on 14 10 the files stored on the FTP server for this Recording Box extension will have the name MyQX Andrew 15 sec 14 10 business wav Attention Make sure Naming Scheme text field contains symbols that your FTP server allows For example symbols lt gt are not allowed by the MS Windows Operation System running servers Retry Count text field indicates the number of retries to access the server in case of networking problems Retry Timeout text field timeout between retries to access the server The Go to Recording Box link moves to the recording box of the corresponding extension s Recording Box where all recorded calls are locally stored The Recording Box is also accessible from Extensions Management table by clicking on the corresponding Recording Box extension Recording Box Recorded calls on the QX IP PBX can either be stored locally in the Recording Box or be uploaded to the remote FTP server The Recording Box is used to locally st
360. n Call Recording Settings X ER User Extension Recording Parameters The Basic View displays the table with the list of all active ia Always start automatically Max Duration ur Recording To 400 extensions recording states of those extensions and recording parameters Always start automatically Max Duration ur Recording To 400 Always start automatically Max Duration ur Recording To 400 The Advanced View displays the table with all existing call recording rules Click on the recording box extension number in the Recorded To column will move to the i Always start automatica Max Duration 1 hour Recording To 400 m corresponding Recording Box aa Always start automatically Max Duration wr Recording To 400 p9o9o99 9 9 99 oo 8 86 8 8 0 8 Always start automatically Max Duration wr Recording To 400 Note there are some ca cannot be shown in Basic View Use Advanced View to see or modify them Switch to Advanced View gt gt gt Fig II 155 Call Recording Basic View Settings page The Call Recording Settings table offers the following functions Enable and Disable functional buttons are used to activate and deactivate the selected call recording rule s At least one rule should be selected in order to use these functions otherwise the following error message will appear No record s selected Add functional button opens the Add Entry page where a new call recording rule is being configured The Add
361. n 30 sec 0 sec 1 sec 302 400 Recording Box 800 Recording Box 850 ACD Group 900 ACD Group 950 ACD Group System memory Ss 0 sec 1 min 32 sec Free Space 1 day 3 hour 31 min 38 sec 5 hour 30 min 20 sec 1 day 3 hour 31 min 25 sec 1 day 3 hour 31 min 38 sec 1 hour 4 min 34 sec 1 hour 6 min 4 sec 1 hour 5 min 57 sec 1 hour 6 min 4 sec 1 hour 6 min 4 sec 1 hour 6 min 4 sec 1 hour 6 min 4 sec 1 hour 6 min 4 sec 2 hour 12 min 8 sec 33 min 2 sec 33 min 2 sec 2 hour 12 min 8 sec 2 hour 12 min 8 sec 16 hour 30 min 59 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 13 day 18 hour 51 min 5 sec Total Space 1 day 3 hour 31 min 38 sec 5 hour 30 min 20 sec 1 day 3 hour 31 min 38 sec 1 day 3 hour 31 min 38 sec 1 hour 6 min 4 sec 1 hour 6 min 4 sec 1 hour 6 min 4 sec 1 hour 6 min 4 sec 1 hour 6 min 4 sec 1 hour 6 min 4 sec 1 hour 6 min 4 sec 1 hour 6 min 4 sec 2 hour 12 min 8 sec 33 min 2 sec 33 min 2 sec 2 hour12 min 8 sec 2 hour12 min 8 sec 16 hour 30 min 59 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 13 day 18 hour 52 min 55 sec Call History 135 record s Call History Archive Total 15484 KB Used 800 KB Conference Memory Status Conference ID y Recorded Conferences 0 sec 0 sec System Message
362. n Type text field is used for a special parameter required by the DynDNS provider TZO The DHS Cloak Title text field is used for a special parameter required by the DynDNS provider DHS The Mail Exchange text field requires the address of the e mail server where the DynDNS service provider will relay your e mails QX50 QX200 QX2000 SW Version 6 0 x 133 QX50 0X200 0X2000 Manual II Administrator s Guide Attention If this service is used ensure that there is port forwarding configured for SMTP port 25 to the internal e mail server The easyDNS Partner text field is used for a special parameter required by the DynDNS provider easyDNS Selecting the Create Custom HTTP GET Request radio button will switch to the custom settings of the DynDNS service Normally the DynDNS provider uses HTTP get requests to map dynamic IP addresses to host names If the HTTP receive request is known to you choose the Create Custom HTTP GET Request radio button and enter the appropriate value into the URL text field The selection enables the following optional settings The URL text field requires the complete request to be sent to the DynDNS server Normally it has the following format http www server domain port scriptpath scriptname param1 value1 amp param2 value2 The request modifies the nameserver database so that the hostname will be resolved to the new IP address The Basic Authentication checkbox enables the encoding of the username and passwor
363. n enabled on Page 1 of the Local Call Routing Wizard It will require information about the pattern validity period s This page provides selection between Typical and Custom date time rule definitions The Typical selection contains the following group of radio buttons that are used to select the frequency of the corresponding routing pattern that is to take place e Daily e Weekly the preferred weekday s should be selected for this option e Monthly the calendar day should be selected for this option e Annually the calendar day and month should be selected for this option In the Available Time Period drop down lists the time range of the pattern validation should be defined Any time selected in this field will be considered corresponding to the QX IP PBX s Date and Time Settings The Custom selection provides the option to manually define the validity period s Use the following format to insert pattern date time rule s Month Month Month Day Day Day hh mm hh mm The Call Routing Wizard Routing Overall Call Limitation Settings Edit Entry page appears if theSet Calling Time Limit checkbox previously had been enabled on Page 1 and allows to define the available duration of the calls with the selected routing rule as well as to specify the Expiration Renewal Date for the available calls duration The Routing Overall Call Limitation Settings Edit Entry page consists of the following components
364. n requires you to enter a routing pattern to the corresponding field Based on the registered PSTN users the caller will be able to reach the destination according to configurations in Call Routing Table PSTN Number Fig II 111 FXO Line Settings page By choosing a destination the QX IP PBX administrator virtually assigns a default number that will start ringing when a call is initiated to the QX IP PBX s PSTN number The PSTN Number text field allows you to enter the PSTN number that the current FXO line is attached to The field value is optional and used as an identification parameter for FXO lines The field value can be left empty QX50 QX200 QX2000 SW Version 6 0 x 76 QX50 0X200 0X2000 Manual II Administrator s Guide e C Pyg l QX50 QX200 QX2000 Manual Il Administrator s Guide Alternative AC Termination Mode appears if the local country Germany Israel France etc selected for QX IP PBX has two COs that use different types of AC termination Contact your CO to learn about your AC termination mode Selecting the checkbox may help if the voice quality over FXO is poor or an echo is noticed To modify the FXO Settings 1 Select the FXO line number from the FXO Settings table The FXO Settings FXO will appear where the line settings may be modified 2 Enable the FXO line to receive calls from the PSTN To reject calls from to the PSTN deselect the Enable FXO checkbox 3 IfFXO has been enabled select the
365. n this case the blocking rule will take effect only in 3 minutes View All Filtering Rules View All displays all configured filters specified by their State enabled or disabled the selected Service the set Action allowed oa na 8 ana or blocked the IP addresses the filters apply to if Restricted and eee Ge ee Ge A ea Gee the destination of port forwarding Since it is read only no Filtering Rules modifications are allowed and no functional buttons are available View all configured rules No changes are allowed in this mode Filter A State Service Action Restricted IP Forward to IP Description SIP Access Enabled SIP Allowed Any None Management Access Enabled HTTPS Allowed Any None Incoming Traffic Port Forwarding Enabled User QuadroFXS26GW Allowed Group MyPC 172 30 4 253 80 Blocked IP List Disabled All locked 172 30 4 254 None Blocked by SIP UA Reason No Such Line Configured Date 05 Aug 2014 19 37 13 Blocked IP List Disabled All 172 30 4 52 Blocked by SIP UA Reason No Such Line Configured Date 05 Aug 2014 15 42 00 Allowed IP List Enabled All Allowed 192 168 0 0 16 Fig II 182 Filtering Rules page QX50 QX200 QX2000 SW Version 6 0 x 120 QX50 0X200 0X2000 Manual II Administrator s Guide Incoming Traffic Port Forwarding The Incoming Traffic Port Forwarding filter is for incoming traffic The rules here allow or deny systems on the Internet to reach the services of QX IP PBX s LA
366. n use IP groups to include several IP addresses for this rule You should only create rules that are D Firewall exceptions to the policy e Selecting Any blocks or allows all host IP addresses This Filter selection is not present for the Management Access Blocked and Allowed IP List rules e Selecting Single IP will require the IP address of the allowed or blocked host e Selecting IP Mask will require the subnet to be allowed or blocked specified by an IP address and the Maskbits The following are Maskbit examples e Single URL requires the hostname of the allowed or blocked host e Group indicates the user defined groups that include IP addresses that should to be allowed or blocked The Description field is used to insert an optional description of the filtering rule 255 0 0 0 8 255 255 0 0 16 255 255 255 0 24 255 255 255 255 32 Fig IT 190 Filtering Rules Page to add a rule for Incoming Traffic To Add a Filtering Rule 1 rte Se ale Select the Filtering Rule Incoming Traffic Port Forwarding Outgoing Traffic Management Access Call Control Access SIP Access Blocked IP List or Allowed IP List to add a rule for it The corresponding Filter table will appear in the same window Click Add on the corresponding filtering rules page Select a service name from the Service list to configure a rule for it If the list has a default value do not change the default values Select an action fro
367. nal button is used to include the selected a extension s to the Pickup Group of the corresponding pickup extension The extensions in the Pickup Group can be monitored by the pickup extension The calls addressed to the extensions in Enabled Attached the Pickup Group can be answered by the pickup extension AE Disabled Attached Enabled Attached Disabled Attached The Disable functional button is used to exclude the selected Enabled Attached extension s from the Pickup Group of the corresponding pickup exte nsio n Disabled Attached Disabled Attached Disabled Attached Disabled Attached Disabled Attached Disabled Attached Fig IT 41 Pickup Group of Extension page The Access List of Extension page lists all users or a group of users if a wildcard is used and the appropriate permissions to pickup the calls ringing on the extensions from the Pickup Group Extensions Add Extension Add Multiple Extensions Bulk Import Access List of Extension 320 a Extensions O Go Back Add Edit Delete Auto 104 105 Fig IT 42 Access List of Extension page for Pickup Group QX50 QX200 QX2000 SW Version 6 0 x 33 QX50 0X200 0X2000 Manual II Administrator s Guide The Add functional button opens an Add Entry page where a new user with corresponding permissions might be created This page consists of the following components Call Type lists the available call types e PBX local call
368. name that will be placed in the end of the routing pattern For example if the routing Pattern is 12345 the Number of Discarded Symbols is two and the Prefix is 909 and Suffix is Oa the final phone number will be 9093450a Destination Type gives you the option to select the destination type The following destination types are available e PBX local calls to QX IP PBX s extensions e PBX Voicemail calls directly to the voice mailbox of the local PBX extension e PBX Intercom local calls to PBX extensions with the request of Intercom service see Manual III Extension Users Guide e SIP calls through a SIP server e SIP_Tunnel calls through a SIP tunnels established see SIP Tunnel Settings e IP PSTN calls through the IP PSTN provider to the remote PSTN global telephone network e FXO calls to a PSTN global telephone network Calls to the FXO global telephone network through shared FXO lines are also present if available e ISDN calls to the PSTN global telephone network through shared ISDN trunk this option is only present when there are shared ISDN trunks available on the QX IP PBX e E1 T1 calls to the PSTN global telephone network through shared E1 T1trunk this option is only present when there are shared E1 T1 trunks available on the QX IP PBX Metric allows entering a rating for the selected route in a range from 0 to 20 If a value is not inserted into this field 10 will be used as the default If two route entri
369. nce extensions and call routing rules in the Call Routing Table In order to configure CoS feature follow the steps below e At first assign the specified CoS s to a certain routing rule s e Assign the specified CoS s to the PBX Conference extension s If there is no CoS assigned to the call routing rule that rule will be generally available for any PBX extension whether it is attached to a CoS or not Please Note If the Enable Class of Service option is disabled call routing rule s that are assigned to a certain CoS s will be available for any PBX extension if there are no any other filtering limitations The Class of Service page offers the following components Enable Class of Service checkbox is used to enable the Class of Service functionality on the QX IP PBX and consists of the following components Class of Services Go to Extensions Management Add opens the Class of Services Add Entry page where a PH f Go to Call Routing Table new class of service can be created t Enable Class of Service Add Edit Delete Edit opens the Class of Services Edit Entry page where the selected class of service s settings can be modified This page includes the same components as the Class of Services Add Entry page does Fig II 154 Class of Services page The Go to Extensions Management link leads to the Extensions Management page where the extensions can be assigned to use certain class of service from the Ex
370. ndefined No entries No entries Display Name undefined User Password Protection disabled both for incoming and outgoing calls User s Name for Extensions Directory default Custom Voice Messages default No answer timeout 20 sec Call Waiting Service enabled Autoredial Interval 10 sec Autoredial Period 15 min 182 QX50 0X200 0X2000 Manual II Administrator s Guide Parameter System Default Value Send Hold Music to remote IP party enabled Basic Services Hold Music Hold Music Own Music Music file default Disabled Timeout 30 min Basic Services Do Not Disturb Send Message to Caller enabled Basic Services Hot Line Disabled available only for QX50 QX200 No entries in the table For Any Callers all services are disabled Caller ID Services Call Blocking message files default Intercom Allow Activation on Request Activation Signal Ring Only if Requested QX50 QX200 QX2000 SW Version 6 0 x 183 e C Pyg l QX50 0X200 0QX2000 Manual Il Administrator s Guide Appendix Moderator s Menus This Appendix explains all menus that can be accessed and configured by conference moderators Applicable if the Conference Server feature is activated on the system Conference Moderator s Main Page The Moderator s Main Page can be accessed by clicking on the conference ID link on the Conferences Management page or by logging as
371. ne display when a call is made or a voice mail is sent O Go Back Extensions Management Edit Entry General Settings Password requires a password for the new extension General Settings 320 SIP Settings SIP Advanced Settings Display Name Subject Pickup320 The extension password may only contain digits If non numeric symbols are entered an Incorrect Password no symbol characters allowed error message will prevent making the extension Generate Password Show on Public Directory Go To Codec Settings Edit Pickup Group If you are unable to define a strong password press Generate Edit Access List Password to use one of system defined strong passwords The Password field is checked against its strength and you may see how strong is your inserted password right below that field Confirm Password requires a password confirmation If the input is not corresponding to the one in the Extension Password field the Incorrect Password confirm error message will appear Fig II 40 Extensions Management Edit Entry General Settings for pickup extension page The Edit Pickup Group link leads to the page where a list of monitored extensions can be defined The Pickup Group of Extension page lists all available regular z and virtual extensions on the QX IP PBX and allows you to manage TRE CE R the Pickup Group Pickup Group of Extension 320 a Extensions Go Back The Enable functio
372. nected to the QX IP PBX The Edit functional button provides a possibility of editing multiple extensions at the same time In this case fields that cannot be edited for multiple records have Multiple values in the Edit Entry page en Extensions Management Edit Entry When editing user and attendant extensions together the Edit Entry ae aes page displays only those fields that are for both user extension and attendant settings Additionally for the fields that need to be s iea ne ae eee modified a Select to modify fields checkbox alongside the A aS a modify corresponding field needs to be selected to submit changes otherwise vote Malos settings fslas Display Name Subject the fields will not be updated Class of Service Settings ga i cad Generate Password Delete removes the selected extensions If no records are selected an pe error message occurs Deleting an extension from the Extensions a P ssh Acs oe Table will automatically remove the name attached to the deleted D pennone uia extension in Extensions Directory EA Save Fig II 69 Extensions Management Edit Entry page for multiple edit operation Extension Codecs To establish an IP voice communication call participants have to use the same codec When establishing a communication line this codec is negotiated If the caller does not find an appropriate codec the communication does not take place To allow communication with all IP callers it is h
373. ng Table starting with digit O will not work for incoming calls to attendant if both the ZeroOut and Send AA Digits to Routing Table options are enabled The ZeroOut feature has a higher priority If it is enabled and used the system will forward all incoming calls to attendant to the specified redirect address As a result calls prefixed with 0 will never reach call routing QX50 QX200 QX2000 SW Version 6 0 x 44 QX50 0X200 0X2000 Manual II Administrator s Guide eo Attendant Welcome Message this group allows updating the active Auto Attendant welcome message played only once when entering Auto Attendant downloading it to the PC or restoring the default one The group offers the following components Enable Welcome Message checkbox is used to enable disable the Auto Attendant welcome message the default one or the custom one uploaded from this page or recorded from the handset see Feature Codes being played when callers enter QX IP PBX s Auto Attendant Upload new welcome message indicates the file name used to upload a new welcome message The uploaded file needs to be in PCMU CCITT u law 8 kHz 8 bit Mono wave format otherwise the system will prevent uploading it and the Invalid audio file or format is not supported warning message will appear The system also prevents uploading if there is not enough memory available for the corresponding extension and the You do not have enough space warning message will appea
374. ng all other participants muted in the conference This service is available for listener participants only Cancel the Request to Speak With this key combination listener cancels his request to speak and a notification hand up icon disappears from the Conference Progress table This service is available for listener participants only QX50 QX200 QX2000 SW Version 6 0 x Keys 900 9009 900 9009 900 9009 900 900 9009 900 9009 908 900 9009 174 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide Mute Unmute With this key combination any participants in the conference may mute and unmute themselves during the conference Please Note You may accelerate dial out by a pound sign at the end of your dialed number Call Codes available in the Auto Attendant For external IP calls addressed to the Auto Attendant following key combination is available to access and manipulate within Auto Attendant services Incoming Call to Auto Attendant Key Combination Conferences Menu used to access conferences Conference ID should be dialed here already in QX50 QX200 QX2000 SW Version 6 0 x 175 Administrator Settings Parameter Admin Settings Host Name Domain Name LAN IP Address DHCP Server Regional Settings and Preferences Emergency and PSTN access codes WAN Interface Protocol WAN Interface Bandwidth WAN IP Configuration
375. ng software and specifically disclaims its quality performance merchantability or fitness for any particular purpose Return Policy If the product proves to be defective during this warranty period please contact the establishment where the unit was purchased The Integrator will provide guidance on how to return the unit in accordance with its established procedures Epygi will provide the Return Merchandise Authorization Number to your retailer Please provide a copy of your original proof of purchase Upon receiving the defective unit Epygi or its service center will use commercially reasonable efforts to ship the repaired or a replacement unit within ten business days after receipt of the returned product Actual delivery times may vary depending on customer location The Distributor is responsible for shipping and handling charges when shipping to Epygi European Limited Warranty The European Limited Warranty is the same as the Limited Warranty above except the warranty period is for two years from the date of purchase Extended Warranty Extended Warranty Option Epygi offers an extended warranty program available for purchase by end users This option is available at the time of purchase extending the users original warranty for an additional three 3 years Combined with the original warranty the extended warranty would offer a total of five 5 years protection for European end users and four 4 years protection for non European
376. ng the default one The group offers the following components Play Announcement When Starting Recording checkbox is used to enable disable the announcement played during the call saying that the call recording starts When this checkbox is not selected the call recording will start silently without any notification Upload new recording announcement message indicates the file name used to upload a new recording announcement message The uploaded file needs to be in PCMU CCITT u law 8 kHz 8 bit Mono wave format otherwise the system will prevent uploading it and the Invalid audio file or format is not supported warning message will appear The system also prevents uploading if there is not enough memory available for the corresponding extension and the You do not have enough space warning message will appear Choose File opens the file chooser window to browse for a new recording announcement message file The Download Recording Announcement Message and Remove Recording Announcement Message links appear only if a file has been uploaded previously The Download Recording Announcement Message link is used to download the message file to the PC and opens the file chooser window where the saving location may be specified The Remove Recording Announcement Message link is used to restore the default recording announcement message QX50 QX200 QX2000 SW Version 6 0 x 41 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Gui
377. ngs SIP Aliases DNS Hostnames to recognize when receiving SIP messages by hostname instead of IP TLS Certificates Generate and install new TLS Certificate or download current one e RTP Settings Advanced Voice Mail Define the voice mail and fax storage method and email notification settings e SIP Settings SIP Settin S RTP Streaming Channels Assign channel names to RTP audio streams emitted by the Epygi Media Streamer application Gain Control Control transmit receive levels of audio interface ports and recording playback level of voice mails SIP Aliases 3PCC Adjust Third Party Call Controlling 3PCC settings Controlling applications to remotely initiate and handle calls and subscribe to event notifications E Radius Client External RADIUS server connection settings TLS Certificates j Dial Timeout Define timeout before sending dialed digits for call processing Call Quality Notification Notify the user when the call quality falls below the specified threshold e Advanced Settings Voice Mail Common Settings Fig II 133 Telephony Menu page RTP Streaming Channels Gain Control 3PCC Settings RADIUS Client Settings Dial Timeout Call Quality Notification QX50 QX200 QX2000 SW Version 6 0 x 90 QX50 0X200 0X2000 Manual II Administrator s Guide VoIP Carrier Wizard The VoIP Carrier Wizard is used to define access codes for available VoIP Carrier accounts which wi
378. nipulation radio buttons available only for QX50 QX200 selection allows to choose whether the virtual interface will be LAN or WAN VLAN ID text field requires the virtual network ID Numeric value in a range from 0 to 4094 is allowed in this field Priority drop down list is used to select the priority of packets in the corresponding interface Packets with the lower priority 0 will be delivered first IP Address text field requires the IP address of the virtual interface Subnet Mask text field requires the subnet of the virtual interface QX50 QX200 QX2000 SW Version 6 0 x Network Network Network QX50 0X200 0X2000 Manual II Administrator s Guide PPP PPTP SNMP Global SNMP SNMP Trap Settings Add Delete Edit Traphost SNMP Version 192 168 70 26 SNMP v2 Traps Community Global SNMP Add SNMP Traphost Traphost 192 168 70 26 Community public SNMP v1 Traps SNMP v2 Traps SNMP v2 Inform Traps Save PPP PPTP SNMP VLAN VLAN Settings Enable Disable Add Edit Delete Interface Vian ID Priority IP Address LAN 10 10 100 5 Fig II 219 SNMP Trap Settings page Fig 1 220 Add SNMP Traphost page Subnet Mask 255 255 255 0 VLAN Settings Add Entry O Go Back W Enable Interface Type 0 4094 Priority 37 IP Address 10 10 100 5 Subnet Mask 255 255 255 0 Fig II 221 VLAN Settings page Fig II 222 VLAN Settings Add Ent
379. not answer it either the call will be directed to the manager s voice mailbox if itis enabled If the manager s voice mailbox is not enabled the call will be disconnected Kickback QX IP PBX allows the receptionist to forward the incoming calls to the manager s extension and if there is no answer or if the called extension is busy on another call the call is returned to the receptionist s phone instead of getting into Voice Mail Service or being disconnected To use this service receptionist should simply transfer the incoming call to the local extension In case of no answer or busy the call will automatically get back to the receptionist Voicemail Transfer QX IP PBX allows the receptionist or extension user to forward incoming calls directly to the voice mail of the other attached extension To do so an appropriate routing pattern should be added to the Call Routing table Hence when transferring a call to the assigned extension incoming call will directly go to the extension s voice mailbox Multi Company Receptionist QX IP PBX provides the possibility to use a single IP phone to manage the receptionist s features for multiple companies at the same time To do so the incoming line appearance for the phone should be created attached to the IP line of the IP phone and be labeled to the corresponding company name Being busy with a call related to one company the receptionist is able to also receive the calls related to other com
380. ns only those extensions that have FAX support enabled FAX support can be enabled from the Extension Codecs page Please Note FAX forwarding is applicable only for incoming calls from PSTN and IP networks It is not valid for PBX calls With the Show on Public Directory checkbox enabled the details of the corresponding auto attendant extension will be displayed in the User Settings table on the Main Page of the Extension s Web Management accessed by the extension s login see Manual III Extension User s Guide Besides this the details of the extension will be displayed in the Public Directories on the Snom and Aastra SIP phones Leave this checkbox unselected if this auto attendant extension is reserved or not used The Percentage of System Memory drop down list is used to define the space for the Auto Attendant s system messages The maximum value in the drop down list is equal to the maximum available space for voice messages on QX IP PBX 2 Attendant Scenario This group is used to select between default and custom attendant functionality scenarios The Default manipulation radio button selection enables the following components e The Send AA Digits to Routing Table checkbox selection switches the Auto Attendant to the routing mode Any inserted digits on the Auto Attendant prompt will be parsed through the Routing Table on the QX IP PBX Overview Conferences Add Extension Add Multiple Extensions Bulk Import Extensions Management
381. nt includes the contract for using QX IP PBX s hardware and software QX50 QX200 QX2000 SW Version 6 0 x 8 QX50 0X200 0X2000 Manual II Administrator s Guide QX IP PBX s Graphical Interface Dashboard Administrator s Main Page If you are logged in as an administrator users admin or localadmin the Epygi QX Management page is displayed with a table of Active Calls which includes information about the originating and terminating party call duration and start time The Terminate button next to each active call is used to terminate the corresponding call The Start Recording button next to each active call except for calls to the Auto Attendant is used to manually start the recording of the corresponding call Once the call recording has started the button changes to Stop now and is used to manually stop the call recording The call recording can be started again if needed The list of administrators currently logged into the system is seen in the lower right corner of the Epygi QX Management page The IP address of the user the time until the next automatic logout and the current version of the Epygi QX s firmware are shown The idle session timeout is set at 20 minutes If no action is performed within 20 minutes the user will be automatically logged out The following main menus are available on the Epygi QX50 QX200 QX2000 Setup Extensions Interfaces Telephony Firewall Network Status and Maintenance By clicking on menus t
382. ntenance Overview Diagnostics Overview Diagnostics Traceroute System Logs System Logs User Rights Backup Restore Download Legible Configuration Firmware Upload Firmware Reboot QX50 0X200 0X2000 Manual II Administrator s Guide Start diagnostics on the WAN Ethernet port ISDN or FXO ports or download the system logs Perform a security audit of the system Capture an active call or select a specific interface to provide a DSP trace for analysis Ping to an IP address or DNS name Perform a traceroute to see the path and response time for each hop to the destination node View system logs Configure general settings of the system logs Choose the logs to be streamed to a remote telnet client Enable disable localadmin set the admin and localadmin passwords Assign permissions to access the GUI pages for localadmin or extensions Backup or restore system configuration and voice data Enable and configure the automatic backup of the system configuration and voice data Generate legible configuration and download to PC or view directly in browser Upload a configuration file in text format Upload firmware image from your computer and install it Get and install a firmware image located on the remote server Perform automatic notification or update the system when new firmware images become available from Epygi Reboot the device Fig lI 262 Maintenance Menu page e C Pyg
383. ntrol enabled Configure IP phones from WAN for QX50 QX200 Phones Default Template systemdefault IP Phone Templates no custom templates IP Phone Logo disabled no custom logos uploaded FXS Gateway Management undefined For QX50 IP Lines 1 16 enabled IP Lines 17 48 disabled 1 16 IP Lines attached to 103 118 extensions All IP lines are in inactive mode For QX200 IP Lines 1 24 enabled IP Lines 25 200 disabled 1 24 IP Lines attached to 103 126 extensions All IP lines are in inactive mode For QX2000 IP Lines 1 200 enabled 1 200 IP Lines attached to 1001 1200 extensions All IP lines are in inactive mode Disabled IP lines displayed FXS Lines Loopback Settings Loopback is disabled for all FXS lines Loopback timeout is 30 Onboard Lines Configuration CallerID Standard 2 FSK for all lines Ringer type Type A for all lines Busy Tone and Power Disconnect indications disabled for all lines Off hook caller ID disabled for all lines Hot Desking Capability disabled for all lines For QX50 2 FXO lines all lines enabled incoming and outgoing calls allowed and routed to 00 Attendant on all lines For QX200 4 FXO lines all lines enabled incoming and outgoing calls allowed and routed to 00 Attendant on all lines For QX2000 Hardware does not support FXO Only shared FXO lines are available Hardware does not support E1 T1 Only shared E1 T1 trunks are available Hardware does not support ISDN On
384. number of PBX services available on the QX IP PBX and accessible with the key combination see QX IP PBX s Feature Codes When configured from this page EE bm the key combinations become transparent for the IP phones too Line Key 2 AutReDl v Redial Fig IT 100 Programmable Keys Configuration page the preview is individual for different IP phone model Vmail accesses the voice mailbox of the extension to which the receptionist IP line is attached to DND enables the Do Not Disturb service on the extension to which the receptionist IP line is attached to CallFwd accessed Forwarding Management of the extension to which the receptionist IP line is attached to AutoReDI auto redials the last dialed call CallBack calls back to the last caller LineInfo gets the IP line information from the QX IP PBX CallBIk blocks the last caller Record records the call in case if the manual call recording is allowed for the call configured from Call Recording Settings ACD Login Logout allows the corresponding ACD agent to login to all groups it is involved in if previously logged in to log out from those groups For details on ACD functionality see ACD Management Please Note When saving changes on this page the system asks for a confirmation to remotely reboot the IP phone It is recommended to reboot the IP phone after configuration changes on this page in order to make the new configuration effective on the IP
385. o select the source type PBX SIP ISDN FXO E1 T1 SIP Tunnel Any used by the source caller to reach the QX IP PBX The settings in the Caller ID Modification group allow Caller IDs of source calls to be modified The Number of Discarded Symbols NDS text field requires the number of digits that should be discarded from the beginning of the Source Number Pattern The field should be empty if digits do not need to be discarded Only numeric values are allowed for this field otherwise the error message Error Number of Discarded Symbols is incorrect digits allowed only will appear The Prefix text field requires entering the symbols alphanumerics Tam AA E and any characters supported in the SIP username that will be placed Call Routing Wizard in front of the Source Number Pattern instead of the discarded digits O Go sak For example if the routing pattern is 12345 the Number of Discarded Source Fiter Modify Caller ID Add Entry Symbols is two and the prefix digits are 909 the final phone number a will be 909345 Wildcards are allowed here see chapter Allowed Source Number Patern Characters and Wildcards dias The two stage dialing is available for PBX ISDN and E1 T1 e destination types The Discard Non Numeric Symbols checkbox is used to discard any non numeric symbols from the Source Number Pattern Number of Discarded Symbols 2 revious Fig IT 140 Call Routing Wizard p
386. o the caller For Paging Group extensions Extensions Management Edit Entry page consists of General Settings SIP Settings and Advanced SIP Settings pages The SIP Settings and Advanced SIP Settings pages are the same as for the regular extensions see User Extension Settings while General Settings page has a different content 1 General Settings for paging group extension This group requires personal extension information and has the following components Display Name is an optional parameter used to recognize the caller Usually the display name appears on the called party s Extensions phone display whenever a call is performed Extensions Management Edit Entry O Go Back General Settings General Settings 350 Password requires a password for the new extension SP Settings SIP Advanced Settings Display Name Subject Paging350 The extension password may only contain digits If non numeric a a symbols are entered an Incorrect Password no symbol characters allowed error will prevent making the extension Go To Codec AAA Settings Edit Paging Group Edit Access List If you are unable to define a strong password press Choose Generated Password to use one of system defined strong passwords The Password field is checked against its strength and you may see how strong is your inserted password right below that field Confirm Password requires a password confirmation If the input is not c
387. oad a custom language for GUL voice messages and supported IP phones Feature Keys Free Trial Activation Redundancy Fig II 2 Setup Menu page Language Pack QX50 QX200 QX2000 SW Version 6 0 x 10 epygl Basic Setup System LAN System Configuration Wizard QX50 0X200 0X2000 Manual II Administrator s Guide The System Configuration Wizard allows the administrator to define the QX IP PBX s Local Area Network settings and to specify regional configuration settings to make QX IP PBX operational in its LAN The System Configuration Wizard MUST be run upon QX IP PBX s first startup to make sure that it works properly in its network environment The Wizard allows navigating through the following basic configuration parameters and settings e System Configuration see below DHCP Settings for the LAN Interface e Regional Settings and Preferences see below e Emergency Codes and PSTN Access Code Settings see below DHCP Settings for the LAN are described in the chapters below The LAN configuration and regional settings will be described later in this chapter Please Note It is strongly recommended to leave the factory default settings if their meanings are not fully clear to the administrator The System Configuration page contains the host name IP address and Subnet Mask information about the QX IP PBX LAN interface These settings make QX IP PBX available to the internal network The System
388. oc Save Fig II 46 Extensions Management Edit Entry Retrieve Access List for call park extension Retrieve Access Settings Add Entry Call Type SIP v 11369 sip epygi loc Address wildcard supported Fig II 47 Extensions Management Edit Entry Retrieve Access List for call park extension The Address text field is used to define the address to be included in the Retrieve Access List table The value in this field is strictly dependent on the Call Type defined in the same named drop down list If the PBX call type is selected the QX IP PBX extension number should be defined in this field For the SIP call type the SIP address should be defined for the PSTN call type the PSTN user number should be defined here The wildcard is supported in this field Wildcard is available for this field QX50 QX200 QX2000 SW Version 6 0 x 35 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide Paging Group Extension Settings Paging Group amp Access List The Paging Group service is used to page a group of extensions by forcing extensions to go off hook and opening one way communication The service is particularly used for announcements addressed to a group of extensions Service allows to page multiple extensions by dialing the Paging Group extension Please Note The Paging Group service requires called extensions to use SIP or analog phones which are able to automatically go off hook F
389. of E1 or T1 enables the Line Code Frame mode Line Build Out Coding Type LoopBackMode Coding Type and Clock Mode settings These settings are configured to match the A E1 T1 settings from the service provider Frame Mode Clock Mode Attention See the Call Routing Table chapter to ensure that modifications to the E1 T1 trunk settings do not lead to broken routes in the Call Routing Table Fig I1 113 E1 T1 Settings Edit Entry page QX50 QX200 QX2000 SW Version 6 0 x 77 pya The Trunk CAS Signaling Settings page lists the available timeslots of the trunk with CAS signaling and their settings The Incoming Interdigit Service link leads to the page where the dial plan for incoming E1 T1 calls from CO PBX to the QX IP PBX can be configured Incoming Digits Timeout text field requires a value between 0 and 20000 in milliseconds and is used to define the timeout during which incoming digits from the destination party calling QX IP PBX will be collected before being applied as an incoming called number Signaling Standard drop down list is available only in E1 mode and is used to select the connection signaling standard Force Update functional button is used to apply immediately the new settings on the selected timeslot s This will force the timeslot s to be restarted and any active connection on the selected timeslot s will be interrupted Enable Disable functional buttons are used to enable disable the selec
390. of IVR is follows via short questions to calling customer determine the set of skills required from the agent for best serving the customer On IVR s questions the customer answers by phone keystrokes DTMF digits each keystroke corresponding to some required skill After finishing the quiz IVR routs the call to AG along with information about the required skills set To calculate the agent s composite skill grade AG sums up the grades of those skills of the agent that are included into the required skill set received from IVR The grades of the non required skills are not considered The composite skill grade of AG is the sum of composite grades of the online agents of that group Interactive Voice Response system ACD IVR is a custom Auto Attendant see Attendant Extension Settings configured on QX IP PBX with VoXML script and voice prompts designed for quizzing the customers determining the set of required skills as described above and routing the call to the agent group having the maximum current value of the composite skill grade for required set Since the general skill set is configured by ACD administrator and is application specific call center specific the VoXML script and voice prompts of IVR should be built taking into account the skill set configured by administrator ACD IVR is needed mainly in case if there are Agent Groups that are configured to do skills based call distribution between agents In such circumstances the IVR i
391. of the entries in the Extensions Directory is important if several records match the same spelled name The Extensions Extensions Directory Settings Add Entry Directory table is parsed from the top down and the matched amp Extensions O Go Back entries will be played according to their position in the table Add opens the Add Entry page where a new name may be assigned to the extension An error message appears and prevents adding a new entry to the Extensions Directory if no extensions are available in the Extensions Management table Fig II 81 Extensions Directory Add Entry page The Add Entry page offers the following components Name requires the name of the extension owner Several extensions can have the same name and a single extension may have several names User s Name is the identification parameter being searched within the Extensions Directory You should use uppercases letters in this field otherwise the name will automatically be changed to uppercase when saving it to the Extensions Directory table Call to drop down list contains all extensions on the QX IP PBX that should ring when selecting the specified Name Description can be used for any optional information requiring entry in the Extensions Directory Please Note The entries in the Extensions Directory can automatically be deleted if the extensions assigned to the entries are removed from the Extensions Management table Receptionist Management The
392. ol tip about conference specific voice Codecs a p wus EZ Conference codec s can be accessed and modified by clicking on m Conf0 725001521680 209506 the link of the corresponding conference s Codecs The Link moves to the Conference Codecs page Add Edit Activate Terminate Delete Fig II 74 Conferences Management page Clicking on the corresponding conference ID will move to the Moderator s page where call general settings can be configured The page Conference consists of the following functional buttons Add opens the Conferences Management Add Entry page where a new conference can be created Edit opens the Conferences Mangement Edit Entry page where the settings of a newly created conference might be adjusted The system provides the possibility of editing multiple conferences at the same time The Edit Entry page consists of two frames In the left frame settings groups are listed Clicking on the corresponding settings group displays their configuration options in the right frame Please Note Save changes before moving among settings groups The Edit Entry General Settings page allows the administrator to edit the following conference settings e Display Name is any optional information about the subject of the conference e The Show on Public Directory checkbox is selected the details of the selected conference will be displayed in the User na Settings table on the Main Page of the Extension s Web ai Management Bes
393. olume upon playing voice mails or system messages For Audio Lines Playback Gain Transmit Gain Line Out defines the level of voice transmitted from QX IP PBX to the Audio Line Out port Audio Lines Receive Gain Line In defines the volume of voice received Receive Gain line in off by QX IP PBX from the Audio Line In port The Restore Default Gains button restores the default values Fig II 172 Gain Control page 3PCC Settings The 3PCC Settings page is used to adjust the third party call controlling settings 3PCC service on the QX IP PBX allows call l Voice Mail RTP Streaming Channels Gain Control Ea A Gee controlling applications to remotely initiate and handle calls on SPCC Settings the QX IP PBX and to subscribe for certain event notifications ASR 2 Secure Connection from the QX IP PBX phony Request Timeout sec 10 Feature Key Added WAN Port Not Opened This page consists of the following components a The Secure Connection checkbox is used enable a secure encrypted connection between the call controlling application and the QX IP PBX Fig II 173 3PCC Settings page Please Note For successful connection this option should be set up in the same way on both sides enabled or disabled on both sides The Request Timeout text field requires the timeout in seconds during which the QX IP PBX should receive a response to the request from the call controlling application If the response is not r
394. om and To are used for the search by ConfID Activation Time ConfID requires the unique ID of the conference For From and To fields the data must be entered in the format dd mm yyyy hh mm ss The time criteria are optional if it is not needed leave the text fields empty The From field must indicate an earlier date and time from that which is indicated in the To field Otherwise the error message Minimal date should be less than maximal date prevents filtering and searching e The text fields From and To drop down lists offer a search by the Call Start Time The data must be entered in the format dd mm yyyy hh mm ss The time criteria are optional if it is not needed leave the text fields empty The From field must indicate an earlier date and time from that which is indicated in the To field Otherwise the error message Minimal date should be less than maximal date prevents filtering and searching e The From and To drop down lists offer a search by the Call Duration specified by the list of values The field From must indicate a shorter duration than the field To Otherwise the error message Minimal duration should be less than maximal duration prevents statistics filtering e The text fields Calling Phone and Called Phone require the calling and called conference party s SIP address extension number or PSTN number as search criterion Wildcard symbols are allowed here The Records per page are used to select the number of displayed sta
395. on the PSTN lines The None selection in this list means that no load balancing will be applied and the call will be routed through the first available PSTN line among the selected ones The Round Robin selection means that according to an internally gained statistics of most used PSTN lines the call will be routed to the less used and currently available PSTN line among the selected ones For ISDN destination type the Port ID drop down list contains the following options e Any Port User OAny any shared ISDN trunks running in User mode e Any Port Network Any any shared ISDN trunks running in Network mode e ISDN Trunk ipaddress shared ISDN trunks on the selected gateway where ipaddress is the IP address of the ISDN gateway that shares its ISDN trunks e Any Port User ipaddress any shared ISDN trunks from the selected gateway running in User mode e Any Port Network Cipaddress any shared ISDN trunks from the selected gateway running in Network mode The Call Routing Wizard Page 3 appears if the Filter on Source Modify Caller ID checkbox had been enabled on Page 1 of the Call Routing Wizard It will require information about the source caller The Source Number Pattern field requires the caller address for which the current route will be applied The complete list of characters and wildcards is allowed in this text field see chapter Allowed Characters and Wildcards The Source Type drop down list gives you the option t
396. onents Call Type drop down list includes possible incoming call types PBX PSTN SIP or Auto QX50 QX200 QX2000 SW Version 6 0 x 34 QX50 0X200 0X2000 Manual II Administrator s Guide e PBX selection means that the call will be push back to the local extension e SIP selection means that the call will be push back to the SIP destination correspondingly e PSTN selection means that the call will be push back to the PSTN destination e Auto selection is used for undefined call types destination independent on whether it is a PBX number SIP address or PSTN number will be reached through Routing Call To text field requires the push back number dialed in the format depending on the selected Call Type The Wildcard is supported in this field 2 Park Access List This page is used to define a list of extensions that are allowed to park the call to the corresponding call park extension Wildcard is supported in the Address field to add a group of extensions with one entry If the extension is not in the Park Access List for the corresponding call park extension it will not be able to park a call to this call park extension By default this table contains a entry which allows any PBX users to park the call to this extension Attention If you modify the Park Access List by adding new extensions do not forget to remove the default entry from the list for the new configuration to take effect 3 Retriev
397. onnection between QX IP PBX and ADSL modem A fixed IP address configuration is needed in this case o Ethernet turns on the Ethernet connection For Protocols PPTP available only for QX50 QX200 Setup WAN IP Configuration see below PPP PPTP Settings WAN Interface Configuration see below DNS Settings Overvie Security System LAN Internet WAN Date and Time Internet Configuration Wizard Uplink Configuration WAN Interface Protocol PPPoE PPTP Upstream 100000 Ethernet Downstream 100000 Vian Min Data Rate 0 For Protocols Ethernet e WAN IP Configuration e WAN Interface Configuration see below e DNS Settings Email SMTP Short Text Messaging SMS kbit s max 100000 kbit s max 100000 kbit s Previous Fig II 8 Internet Configuration Wizard Uplink Configuration page The WAN Interface Bandwidth settings allow the specification of the upstream and downstream speeds in kbit s helping to assure the quality of IP calls An IP call looses the voice quality if there is no available bandwidth When approaching the limits of bandwidth capacity another IP call will be declined The bandwidth provided by the ISP has to be specified in the text fields Upstream Speed and Downstream Speed The default entry in both fields is 100000 the maximum bandwidth of a 100 Mb Ethernet You may see the required bandwidth in the chapter Needed Bandwidth for IP Calls The
398. onnection can be created The Password text field requires the password for the connection establishment Please Note These authentication settings should be identically Fig II 231 PPTP L2TP Connection Wizard for PPTP connection Page 2 configured on both peers for the successful connection QX50 QX200 QX2000 SW Version 6 0 x 141 pya establishment The manipulation radio buttons selection on this page allows you to choose whether the new connection will be a client or a server For the Client radio button selection no further details need to be provided For the Server radio button selection the following information needs to be provided For PPTP connection the PPTP Server text field requires an IP address or a host name of the PPTP server For L2TP connection the L2TP Server text fields require an IP address of the L2TP server The Authentication manipulation radio buttons are only present if the Connection Type selected on the previous page is PPTP They are used to select the corresponding authentication protocol by which the client communicates with the server The MSCHAPv2 selection enables the Encryption drop down list where the encryption method can be selected Network QX50 0X200 0X2000 Manual II Administrator s Guide Oven DHCP DNS PPP PPTP SNMP VLAN VPN PPTP L2TP Connection Wizard O Go Back L2TP Connection Properties to74554Li Peer Name Quadro4Li Password je
399. oose File No file chosen e Transfer Statistics Remove current language pack Please Note Only one custom Language Pack can be uploaded at IP phones language packs can be ational customized her Upload IP Phones LP the time Uploading a Language Pack will remove the existing one if applicable and will reboot the QX IP PBX ATTENTION After pressing Save you ll have to reboot the device manually even if you don t install the language pack Save Fig IT 20 Language Pack page The Current Language Pack field displays read only information about the custom language pack uploaded When no custom language pack is uploaded the field indicates No Language Pack installed Below there is a Language Pack File to Upload text field that displays the selected image filename The Choose File button is used to browse the custom language pack to be uploaded The Remove Current Language Pack link is only seen when a custom language pack is uploaded and is used to remove it from the system The Custom languages for IP phones link is only seen when a custom language pack is uploaded and is used to move to the Update Languages for IP Phones page where a custom language pack may me uploaded to the IP phone Pressing Save will start uploading the custom language pack to the board Attention Pressing the Save button will stop some vital processes on the QX IP PBX therefore you will need to reboot your device manually even if you have canc
400. or Paging service supported on IP phones refer to the Epygi IP PBX Features on Epygi Supported IP phones document on the Epygi s Web portal The Paging Group list is used to define the extensions that will be paged They will automatically go off hook when the paging call comes in The Access List is used to define PBX SIP or PSTN users that are explicitly allowed forbidden to activate the call paging using the corresponding extension When calling to the Paging Group extension the call will be forwarded to the extensions listed in the Paging Group table The phones of the called extensions will automatically go off hook the phone speaker automatically becomes activated and the caller will be able to make his announcement Since the paging call opens one way communication the called extensions will not be able to give an answer to the caller To terminate the paging call caller should simply hang up Attention Call paging will not work if the called extension is in call When caller not listed in the Access List calls the Paging Group extension password authorization using the password of the Paging Group extension will be required to start the call paging When a denied user tries to call the Paging Group extension Party does not accept your call message will be played to the caller When caller dials the Paging Group extension with empty Paging Group table Number dialed temporarily unavailable message will be played t
401. or digit to be dialed and when timer expires it initiates the call Timer is not applicable for DMS 100 switch types e The T309 Timer text field requires the value for the T309 timer in milliseconds digit values from 0 to 90000 responsible for call steadiness during link disconnection within the period equal to this timer value If the value in this field is 0 T309 timer will be disabled e The T310 Timer text field requires the value for the T310 timer in milliseconds digit values from 1000 to 120000 responsible for the outgoing call steadiness when CALL PROCEEDING is already received from the destination but call confirmation ALERT CONNECT DISC or PROGRESS is not yet arrived e The No Answer Disconnect Timer text field requires the value for the No Answer Disconnect Timer digit values from 0 to 200000 which is used in certain types of PBXs The value 0 indicates that the timer is disabled When time expires QX will play a busy tone towards the PBX if the call has been disconnected by the peer The D Channel Timeslot For Transmit Receive drop down list contains the timeslots to be selected for signaling data transmit receive QX50 QX200 QX2000 SW Version 6 0 x 82 QX50 0X200 0X2000 Manual II Administrator s Guide The B Channel link leads to the Signaling Type CCS B Channel Settings page where available timeslots may be enabled disabled for E1 T1 Trunk ISDN Trunk PSTN Gateways the voice transfer and echo cancellation featu
402. or may go to the Call Routing table so any unprotected 5 routing rule can be misused That is why it is strongly recommended to secure the rules in the Call Routing table by setting the filtering or authentication options Fig II 145 Call Routing page 99 e C Pyg l QX50 QX200 QX2000 Manual Il Administrator s Guide Attention Regardless of whether the Route all incoming SIP calls to Call Routing checkbox is selected or not SIP calls from external callers will or may go to the Call Routing table so any unprotected routing rule can be misused That is why it is strongly recommended to secure the rules in the Call Routing table by setting the filtering or authentication options Local AAA Table The Local AAA Table page allows you to manage local authentication and the authorization database Callers dialing the routes which have an AAA Authentication Authorization and Accounting option enabled will pass the authorization on the Local AAA Table by using a phone number or username password depending on the corresponding entry configuration on this page The caller passes authorization automatically if the detected phone number of the caller dialing a route has the AAA option enabled and is e registered in the Local AAA Table If the caller ID service is disabled or e en a the caller s phone number is not registered the caller is asked to enter a de registration user name and password ee Add Edit Delete Caller I
403. ore the recorded calls The Recording Box can be accessible online from Web Management or from handset by calling the corresponding Recording Box extension With both options the user can play and delete the recorded calls located in the Recording Box Please Note When using Call Recording on the QX50 QX200 it is advisable to use an SD memory card to expand the system memory QX50 QX200 QX2000 SW Version 6 0 x 42 QX50 0X200 0X2000 Manual II Administrator s Guide When accessing the Recording Box through the handset all recording box functionality settings such as enabling the welcome message adjusting the maximal call recording duration recording box access security etc are configurable from Recording Box Extension Settings page Instructions on accessing and navigating within the Recording Box via the phone handset are described in the Feature Codes Please Note When playing a new call recording via a phone handset or with the use of the Play button in this page will deprive the New state of the recorded call The Recording Box can hold New not yet played and Old already played call recordings The Status column in the Recording Box table indicates the current state of the call recordings All new recordings in the table are displayed in bold font Playing a call recording cancels both the New status and bold font Call recording can be selected to be played or deleted The following information is available on this page
404. ork availability or equipment failure is beyond Epygi s control and Epygi shall have no responsibility for losses arising from such interruption Music on Hold Copyright The default Music on Hold on the Quadro or QX is a 22 second fragment from Chopin s Nocturne Op 9 2 performed by Marina Vardanyan and kindly provided to Epygi Technologies LTD The recording is royalty free Compliance with Laws You may not use the Epygi Materials for any illegal purpose or in any manner that violates applicable domestic or foreign law You are responsible for compliance with all domestic and foreign laws governing Voice over Internet Protocol VoIP calls QX50 QX200 QX2000 SW Version 6 0 x 3 QX50 0X200 0X2000 Manual II Administrator s Guide Table of Contents Manual I see Installation Guide Step by step guide to install and configure QX IP PBX basically Manual II Administrator s Guide About TEMES Administrator s QUIE AN A AE EAN 8 OXIPPBAS Graphical Interna aaa AAA A AR 9 Dashboard AGMIMIStrator Main Page ree 9 Adminstrator S MEMU Sii dd ANNAN 10 SOU MCI APN o E O E E AA A 10 AAPP o o o daidi d a aaarnas dieis 11 py Stel LAN Sy stent COMPTON Wi dai 11 Internet WAN Internet Conde tlration WIZaARO mooiii 12 Needed Bandido Cal Roca 14 Date eC Tie 5c UNS aia ita 15 System all Sectas Emal SMTP aaa iaa airada 16 SMS Settiias Or rexe Mes sac aia 17 SISTE ECU ada 18 LICENSE FEAtUTES E E 18 Ss a eee AP
405. orresponding to the one in the Extension Password field the error will appear Incorrect Password confirm Fig IT 48 Extensions Management Edit Entry General Settings for paging extension page The Edit Paging Group link leads to the page where a list of extensions to be paged can be selected The Paging Group of Extension page lists all available regular and virtual extensions on the QX IP PBX and allows you to manage Add Multiple Extensions Bulk Import the Paging Group 8 txensions Paging Group of Extension 350 The Enable functional button is used to include the selected a extension s to the Paging Group of the corresponding extension ney Once the call to the paging group comes in all the extensions in 101 Disabled Attache that group will be paged i e will automatically go off hook by ps eamhan automatic activation of the phone s speaker Enabled Attached Enabled Attached The Disable functional button is used to exclude the selected extension s from the Paging Group of the corresponding Enabled Attached Enabled Attached exte MALO n Enabled Attached Enabled Attached Enabled Attached Enabled Attached Disabled Attached Fig IT 49 Paging Group of Extension page QX50 QX200 QX2000 SW Version 6 0 x 36 QX50 0X200 0X2000 Manual II Administrator s Guide The Edit Access List link leads to the page where permissions for users to use the Paging Group service can be
406. oto TCP 1P 192 168 74 185 Port 80 Wed Aug 6 18 36 03 2014 Proto TCP 1P 192 168 74 185 Port 30 Fig II 181 IDS issue detailed preview Filtering Rules The Filtering Rules page allows you to configure the filters for incoming and outgoing traffic To prevent inaccurate configuration only one rule per service is allowed The user may use IP groups to include several IP addresses for this rule Since the filtering rules specify the operation mode of the firewall they only take effect if the firewall has been enabled additionally NAT should be enabled to use the Port Forwarding function in the Incoming Traffic Port Forwarding filtering rules The filtering rules are independent from the security level so they will work if enabled no matter what security level has been selected Please Note Applying firewall rules will prevent the establishment of new connections that violate the rules Applying rules does not kill existing connections that violate the rule Attention The newly created blocking filtering rules will take effect immediately if there is no any active connection matching to that rule Otherwise if there is an active connection matching to the created blocking rule please restart the QX IP PBX to make the newly created blocking rule effective immediately However if you are unable to restart the QX IP PBX you may need to stop an existing active connection to make the newly created blocking rule effective Please note that i
407. ound Autodialer Support No key found Conference Server No key found Video Conferencing No key found Call Recording No key found Disabled Default English Current Language Pack none Extension Length 3 once applied extensions 00 101 150 appear for QX50 00 101 302 appear for QX200 176 Parameter Extensions Management for QX2000 Extension Settings General for QX50 QX200 Extension Settings General for QX2000 Extension Settings SIP Extension Settings SIP Advanced Extension Settings Remote Extension Settings Call Queue Extension Settings Voice Mailbox Extension Settings Codecs for QX50 QX200 QX50 QX200 QX2000 SW Version 6 0 x QX50 0X200 0X2000 Manual II Administrator s Guide System Default Value Extension Length 4 once applied extensions 00 1001 1200 appear for QX2000 Display name none Password empty 101 and 102 extensions are attached to the FXS lines 1 and 2 correspondingly 103 118 extensions attached to the IP lines 1 16 for QX50 103 126 extensions attached to the IP lines 1 24 for QX200 Kickback disabled Call Relay disabled Login Allowed disabled 3pcc Click2Dial Access Allowed disabled Audio Line out disabled Show on Public Directory disabled Percentage of Total Memory for extensions 101 102 5 Percentage of Total Memory for extensions 103 118 0 4 for QX50 Percentage of Total Memory for extensions 103 126 0 4 for
408. ound at Epygi Technical Support e iQall Toggling Support this feature enables users to alternate the call from their mobile device iPhone running iQall to their desk phone without the call being dropped QX50 QX200 QX2000 SW Version 6 0 x 18 QX50 0X200 0X2000 Manual II Administrator s Guide e IP Phone Support enables additional IP phones support on the Epygi QX50 0X200 0QX2000 This feature key allows activating up to 8 16 or 32 additional IP lines for QX50 up to 8 16 32 64 or 128 additional IP lines for QX200 and up to 8 16 32 64 or 128 additional IP lines for QX2000 which will bring to a maximum 2000 total IP lines for QX2000 e Autodialer Support allows run with QX IP PBX the Autodialer application the application description can be found at Epygi Technical Support e Conference Server activates the conferencing feature allowing the system to act as a standalone conference server This allows up to 16 person conference calls for QX50 up to 32 person conference calls for QX200 and up to 288 conference calls for QX2000 to be set up and offers a bundle of helpful features to easily manage the conferences e Call Recording activates the Call Recording feature which is used to record PBX SIP or PSTN calls on the QX IP PBX and save the recordings into the local recording box or upload to the remote server Please Note When using Call Recording on the QX50 QX200 it is advisable to use an SD memory card to expand the
409. out QX IP PBX s Field may contain some kind of password which should be matching both on QX IP PBX and on the administrating application for successful SNMP management Enable SNMP v1 2c Read Write Access checkbox additionally enables a read write access on the QX IP PBX for the SNMP monitoring application With this checkbox enabled administrator will be able to remotely configure the QX IP PBX via SNMP administrating program Fig IT 218 Global SNMP Settings page SNMP v1 v2c Read Write Community text field is used to insert the community description public private etc for the read write management like gathering information events statistics etc about QX IP PBX s and remotely changing QX IP PBX s configuration Field may contain some kind of password which should be matching both on QX IP PBX and on the administrating application for successful SNMP management The Service Restart button restarts the SNMP sub system on the QX IP PBX Restarting the SNMP sub system is recommended if it does not respond to a SNMP manager s requests QX50 QX200 QX2000 SW Version 6 0 x 135 SNMP Trap Settings SNMP Trap Settings page is used to define the traphosts that should be informed when certain events occur on the QX IP PBX For the listed traphosts to be informed about the events on the QX IP PBX Send SNMP Trap action should be configured for the corresponding event s from the Events page SNMP Trap Sett
410. out the private key QX IP PBX generates such a pair of keys automatically when it is set up The user cannot see the private key but must know the public key because their IPSec connection partner will need it Please Note A pair of keys will always be generated a public one and a private one The previously generated pair of keys will become invalid as well as all existing IPSec connections that use RSA keying The IPSec Configuration link refers to the page where IPSec connections can be created and managed The IPSec Configuration page consists of two sub pages Connection and RSA Key Management Connection The Connection sub page provides an overview of all existing IPSec connections characterized by their Connection Name the Remote Gateway the IP address or the hostname of the IPSec connection partner the State of the IPSec connection Stopped Connecting Activated Waiting or Connected and the dedicated Keying Type the encryption type The content of the table can be sorted in ascending or descending order by clicking on the header of the respective column There is a checkbox for every IPSec connection to select it for further editing Start activates the connection establishment of the selected IPSec connection The State of the IPSec connection will change into Connected or Activated depending on the IPSec connection eeen type If no record is selected the error message One Record IPSec Configuration
411. over NAT 150 Mapped Port requires the port number on the mapped host for the SIP UDP traffic over NAT Fig IT 159 NAT traversal Settings SIP Parameters page TCP TLS Parameters Mapped TCP Host requires the IP address of the mapped host for SIP TCP traffic over NAT Mapped TCP Port requires the port number on the mapped host for the SIP TCP traffic over NAT Mapped TLS Host requires the IP address of the mapped host for SIP TLS traffic over NAT Mapped TLS Port requires the port number on the mapped host for the SIP TLS traffic over NAT RTP Parameters The RTP Parameters page is used to choose between the STUN and Manual NAT traversal connection for the RTP traffic and to define the RTP RTCP ports for the connection over NAT Manipulation radio buttons allow you to select the type of connection over NAT QX50 QX200 QX2000 SW Version 6 0 x 109 QX50 0X200 0X2000 Manual II Administrator s Guide Selecting Use STUN will switch to automatic discovery of Mapped settings for the RTP UDP traffic over NAT STUN settings are configured on the STUN Parameters page see below Selecting Use Manual NAT Traversal allows you to manually define the RTP RTCP port ranges for the RTP traffic over NAT e The Mapped Host text fields require the Mapped Host for RTP traffic over NAT e Mapped RTP RTCP Port Range e Min minimal port has to be higher than 1024 and lower than the maximal port range Only even numbers are allo
412. owed error will prevent making the extension Confirm Password requires a password confirmation If the input is not corresponding to the one in the Extension Password field the Incorrect Password confirm error will appear With the Show on Public Directory checkbox enabled the details of the corresponding extension will be displayed in the User Settings table on the Main Page of the Extension s Web Management accessed by the extension s login see Manual III Extension User s Guide Besides this the details of the extension will be displayed in the Public Directories on the Snom and Aastra SIP phones Leave this checkbox unselected if the extension is reserved or not used or when the extension serves as an intermediate unit for call forwarding etc The Percentage of Total Memory drop down list allows you to select the space for the uploaded custom messages The maximum value in the drop down list is equal to the maximum available space for voice messages on QX IP PBX QX50 QX200 QX2000 SW Version 6 0 x 37 e C Pyg l QX50 QXK200 QX2000 Manual Il Administrator s Guide 2 ACD Group Settings This group is used to adjust the ACD group settings and has the following components Max Queue Size defines the maximum number of calls waiting in the queue If all positions of the queue are busy and a new call arrives it will be rejected by the Agents Group Agent Ring Timeout defines the maximum ringing time of the aa EA
413. own list allows you to select the Maintena Emergency Codes prefix code for accessing the PSTN line in the routing mode PR Dialing the digits inserted in this text field will provide the PSTN dial tone when dialed from the handset Previous Fig II 6 System Configuration Wizard Emergency Codes and PSTN Access Code Settings page Internet WAN Internet Configuration Wizard The Internet Configuration Wizard Uplink Configuration Wizard in case of QX2000 allows the administrator to configure the WAN interface settings and to adjust QX IP PBX s connectivity with an external network The Internet Configuration Wizard MUST be run for QX IP PBX to be connected to the Internet QX50 QX200 QX2000 SW Version 6 0 x 12 All the settings of the Internet Configuration Wizard are described in the chapters below except those for the IP settings which will be described in this chapter Attention It is strongly recommended not to change the factory default settings if their meanings are not fully clear to an administrator Setup QX50 0X200 0X2000 Manual II Administrator s Guide Overvien Security System LAN Date and Time Internet Configuration Wizard Getting Started This wizard guides you through Uplink Selection and based on which WAN Interface Protocol you select there through for Protocols PPPoE ll for Protocol PPTP PPP PPTP Configuration WAN Interface Configuration DNS Settings
414. page includes the same components as the Add New Service page To operate with Edit only one record may be selected otherwise the error QuadroFXS26GW Y Firewall Fig II 191 Service Pool Configuration page message One row must be selected will appear QX50 QX200 QX2000 SW Version 6 0 x 123 e C Pyg l QX50 QX200 QX2000 Manual Il Administrator s Guide The Add page is used to add new services and includes the following text fields and buttons Service Pool Configuration Add Service Name requires a name for the service that should be O Go sack added Service Name GameZone 7 Protocol TCP X Protocol includes a list of possible protocols to be selected TO pease in 2550 Port Range requires a port range for the defined service x 2590 Fig IT 192 Service Pool Configuration Add Service page To Delete a Service 1 Check one or more checkboxes of the corresponding services that should be deleted from the Service Pool Configuration table 2 Click on the Delete button on the Service Pool Configuration page 3 Confirm the deletion by clicking on Yes or cancel by clicking on No IP Groups IP Pool Configuration The IP Pool table is the list of all added groups and the members assigned to these groups If a group is empty EMPTY will be indicated in the Members column If hidden group members will still remain active but HIDDEN will be displayed in the Members column IP Pool Configur
415. panies While calls are ringing in the background the receptionist can switch between the incoming calls If the receptionist does not answer the incoming calls and if the Call Queue service is enabled on the extensions the incoming calls will be stored in the queue specific for each company line QX50 QX200 QX2000 SW Version 6 0 x 58 QX50 0X200 0X2000 Manual II Administrator s Guide The Receptionist Management page allows you to configure IP phones to be used as a receptionist on the QX IP PBX This page contains the list of configured receptionists with information about the attached IP lines and watched extensions Receptionist Management amp Extensions Add Edit Delete Receptionist Attached IP Lines Watched Extensions Epygi Receptionist Fig II 82 Receptionist Management page Add opens the Receptionist Phone Configuration Wizard where the new receptionist phone can be created and configured The wizard consists of several pages The Receptionist Phone Configuration Wizard IP Phone Model page has the following components The Description text field requires the description of the receptionist to be configured Receptionist Phone Configuration Wizard Extensions The Phone Model drop down list is used to select the IP phone IP Phone Modes model to be used by the receptionist Description Epygi Receptionist The MAC Address text fields require the MAC Address of the An A corresponding IP phone e MAC Ad
416. pants added from the handset or GUI unless the conference is terminated For the active conference the table also displays participants added manually from GUI or from the handset and those participants that called in to the conference The Conference Progress table contains the following information for each participant Name this information is specific to manually added participants only see below SIP Address indicates the SIP address of the participant Participant Type indicates whether the participant is a speaker or a listener only Participant Indication indicates whether or not a beep indication during the call conference is configured for this participant to be played when he joins or leaves the conference Participant Status this column is only present for active conferences and indicates the state of the participant active for participants currently in the conference not active for participants not in the conference and joining for participants currently joining but not yet connected to the conference Nested Conference indicates if the participant acts as a nested conference or not Request to Speak this column is only present for active conferences and indicates whether a listener participant has requested to speak by dialing 9 from the handset see Feature Codes When a listener participant requests to speak a hand up icon appears in this column Clicking on the hand icon in this column will grant the speak
417. pecific parameter obtained from the server and should match on both sides e Custom this selection allows to use a custom SMS gateway Selection requires following parameters to be inserted Resource text field requires the HTTP resource name on the SMS gateway for example http sms cgi Parameters text field requires the parameters to be submitted to the resource address The value of this field represents a string with tokens separated by percent symbols inside Each token indicates a value of the certain field on this page The value is dependent on the SMS gateway requirements For example user usernameN password password to to from from text text The tokens are the strings that have the following dependencies from the field in this page username indicates the username defined in the field Username password indicates the password defined in the field Password to indicates the password defined in the field SMS Recipient Address from indicates the password defined in the field SMS Sender Address text indicates the SMS text generated by QX IP PBX voice mail notification event notification etc Server text field requires the IP address or the host name ofthe SMS gateway Port text field requires the port number of the SMS gateway Use Secure HTTP checkbox enables access to SMS server via HTTPS Checkbox selection enables a Secure Port text field that requires the port number for HTTPS traffic
418. pecified 5060 will be used The range of valid ports is between 1024 and 65536 SIP Address Tel Number 2347852 sip epygi loc Participant Type Speaker v Confirmation Type Star Y Y Allow Video Dial Out Y Participant Indication Please Note A direct call will be placed toward a participant s Men SIP address if the corresponding conference is registered on a a Atte different SIP server than the participant is registered on or if the participant is not registered on any SIP server The value will be implied as a Routing Number and will be parsed through the Call Routing table if it does not match the SIP URI syntax Participant Type list is used to select the type speaker or aa i Fig II 295 Conference Progress Add Participant page listener of participant in the conference Confirmation Type list is used to set the password protection for the participant joining the active conference Star selection allows the participant to accept the conference invitation by pressing the button Only participants connected to the conference with the moderator password will be provided with permissions to manipulate the conference A group of checkboxes on this page allow configuration of participant specific settings e When the Dial Out checkbox is selected the participant will be automatically dialed out when the conference is activated e Participant Indication enables the beep indication during the conference wh
419. ple road warrior connections with the Shared Secret automatic keying selected For multiple road warriors to be started at the same time it is recommended to use RSA keying with Local ID and Remote ID fields configured e RSA requires the public RSA key of your IPSec Connection partner Please Note System prevents to start a connection with Shared Secret automatic keying selected if there is already a connection with RSA automatic keying started and vice versa The Local ID requires an IP address QX IP PBX FQDN Fully Qualified Domain Name that is resolved to an IP address or any ed string that is used in the same way Remote ID also requires an IP address the IPSec Connection partner s FQDN Fully Qualified Domain Name that is resolved to an IP address or any ed string that is used in the same way The Local ID and Remote ID text fields may have the values in As one of the formats presented below IPSec Configuration Wizard e IP address example 10 1 19 32 Locos Automatic keying to7440M32x e Host name example vpn epygi com This form requires additional resources to resolve the host name therefore it is not recommended to use this format Remote RSA public key e FQDN example vpn epygi ccom This form is considered as a string and is not being resolved It is recommended to use this form for most applications Local ID 1 e user FQDN example qx vpn epygi com This form is also considered as
420. provisions contained in this Agreement are found to be invalid or unenforceable in any respect the validity and enforceability of the remaining provisions shall not be affected 13 Governing Law This Agreement shall be governed by and construed in accordance with the laws of the state of Texas without regard to choice of law provisions that would cause the application of the law of another jurisdiction 14 Attorneys Fees In the event of any litigation or other dispute arising as a result of or by reason of this Agreement the prevailing party in any such litigation or other dispute shall be entitled to in addition to any other damages assessed its reasonable attorneys fees and all other costs and expenses incurred in connection with settling or resolving such dispute Ifyou have any questions about this Agreement please write to Epygi at 1400 Preston Road Suite 300 Plano Texas 75093 or call Epygi at 972 692 1166 15 Free Software Certain software utilized in the Epygi products is free software in its original form or in its modified form Both types of free software are available to you free of charge for redistribution or modification under certain conditions Permission is granted to copy distribute and or modify any free software you wish to download whether in its original or modified forms under the GNU General Public License or Free Documentation License Version 1 1 or any later version published by the Free Software Foundat
421. public phone e After the defined period of time requires the period after which the extension will automatically log out from the public phone e Atthe certain moment requires the moment hour and minute when the extension will automatically log out from the public phone By pressing the Web link in the Details column for each configured SIP phone will lead you to the Web configuration page of the corresponding SIP phone Please Note This link only works from the LAN side of the QX IP PBX i e when the QX IP PBX s GUI is accessed from a PC located in the QX IP PBX s LAN If you wish to connect the SIP phone s GUI through the WAN an appropriate Incoming Traffic Port Forwarding filtering rule should be added on the QX IP PBX The Advanced link in the Details column takes you to the Programmable Keys Configuration page where programmable keys for the corresponding IP phone can be configured The Reboot link in the Details column appears for supported IP phones and is used to remotely initiate a reboot of an IP phone attached to the line IP Line Settings Enable PnP to IP lines checkbox is used to setup the SIP phones connected to the QX IP PBX via Plug and Play automatic configuration service To use this service this checkbox needs to be selected The SIP phone should be reset then After a a S E1 T1 Trunk ISDN Trunk PSTN Gateways clean boot up of the SIP phone QX IP PBX will detect the SIP _ IP Line Settings phone and all
422. r Browse opens the file chooser window to browse for a new welcome message file The Download Welcome Message and Remove Welcome Message links appear only if a file has been uploaded previously The Download Welcome Message link is used to download the message file to the PC and opens the file chooser window where the saving location may be specified The Remove Welcome Message link is used to restore the default welcome message Recurring Attendant Prompt this group allows updating the active recurring Auto Attendant message played after the Attendant Welcome Message and then periodically repeated while being in the Auto Attendant downloading it to the PC or restoring the default one The group offers the following components Upload new Recurring Attendant Prompt indicates the file name used to upload a new recurring auto attendant prompt The uploaded file needs to be in PCMU CCITT u law 8 kHz 8 bit Mono wave format otherwise the system will prevent uploading and the Invalid audio file or format is not supported warning message will appear The system also prevents uploading if there is not enough memory available for the corresponding extension This will cause the You do not have enough space warning message to appear Browse opens the file chooser window to browse for a new Recurring Attendant Prompt file The Download Recurring Attendant Prompt and Remove Recurring Attendant Prompt links appear only if a file has been
423. r Gi 45 sed di G sea d Q sed d 1 sed Q G sea d Q seo Q 4 min 4 sec GG seo Q 1 min 13 sed Qi 56 sed Qi 17 sea d 59 sed d 53 sed Q 1 min 8 sec Q 1 min 25 sec Q 1 min 53 sec d 26 sed Q 1 min 27 sec Q 1 min 24 sec d 12 sed d 28 sed 32 min 28 sec Fig II 60 Extension s Recording Box Message indicates call recording duration in minutes seconds and a speaker sign used to play using any available media player supported by your Operation System the recording or to download the audio file to the PC The column headings of the voice mail tables are created as a link By clicking on the column heading the table will be sorted by the selected column Upon sorting ascending descending arrows will be displayed next to the column heading Each row in the Voice Mailbox tables can be selected by a checkbox for editing deleting or marking To Play a Call Recording 1 Click on the speaker icon of the corresponding recorded call 2 Depending on you browser s settings the wav file will be played directly or an application will ask you to save the wav file on the local PC In the second option please specify the path and run the media file from the specified location to play it To Delete a Call Recording 1 Select the checkbox of the corresponding record s in the Recording Box table that should to be deleted 2 Select the Delete button 3 Confirm
424. r s phone number Please Note DTMF supports only parameters consisting of digits If any letter symbol has been used in the external caller user name DTMF will not display caller ID To Configure the Line Settings 1 Select the line number that should to be configured from the Active Lines column in the Lines table on the Line Settings page 2 Press on the line number link in the Line Settings table The Line Settings Line page will appear in the browser window 3 Use the Caller ID drop down list to select the caller ID detection system mode corresponding to the phone type 4 Enable the Dialing Prefix With Caller ID checkbox if needed 5 Configure the Remote Party Disconnect Indication parameters by selecting the corresponding checkboxes 6 Define a Ringer Type from the corresponding drop down list 7 Enable Off hook Caller ID if needed 8 Press the Save button on the Line Settings Line page to save the caller ID system and other line specific configuration settings Diagnostic Loopback The FXS Lines Loopback Settings page is used to configure the lines for voice loopback diagnostics When loopback is enabled on the line any incoming calls to the corresponding line will automatically pick up on the first ring and any voice towards the line will automatically be sent back to the caller the caller will hear themselves in the handset Loopback Timeout provides the option of limiting the voice loopback diagnostics duration i e th
425. r The Alternative Nameserver text field requires the IP address of the secondary name server The Alternative Nameserver is used if the main name server cannot be accessed Fig II 212 DNS Settings page DNS Server Settings The DNS Server on the QX IP PBX provides the services to the hosts in the QX IP PBX s LAN With this service QX IP PBX returns the correct IP address to the requested domain name so that any device in the LAN can be accessed by its hostname or alternative alias name The DNS Server Settings page is used to configure DNS server settings on the QX IP PBX and to define a list of aliases for the devices in the QX IP PBX s LAN This page contains the following components QX50 QX200 QX2000 SW Version 6 0 x 132 QX50 0X200 0X2000 Manual II Administrator s Guide Zone field displays the QX IP PBX s host domain name as it is oue DHO configured in the System Configuartion Wizard ons CED oromi ons Time to live TTL text field indicates the time in seconds during PEES which the DNS server will keep the resolved names in its cache moman 7 ana During this time the same address will be resolved from the cache of Mai exchange 000 the DNS server When this timeout expires the requested address Add at Delt will be resolved newly _ cm na Mail Exchange MX text field indicates the mail server s hostname When resolving the email address the reference will go to the mail server defined in this
426. r PSTN users that are allowed or forbidden to intercept calls ringing on extensions in the Pickup Group If a user dials the pickup extension when several extensions of the pickup group are ringing the first oldest in time call will be picked up When the user dials the pickup extension and no extensions of the pickup group are ringing the No call is available to pickup message will be played to the user When QX50 QX200 QX2000 SW Version 6 0 x 32 QX50 0X200 0X2000 Manual II Administrator s Guide the user that is not listed in the Access List dials the pickup extension password authorization of the pickup extension will be required to answer the call When a denied user dials the pickup extension the Party does not accept your call message will be played to the user For Pickup Group extensions the Extensions Management Edit Entry page consists of General Settings SIP Settings and Advanced SIP Settings pages The SIP Settings and Advanced SIP Settings pages are the same as for regular extensions see User Extension Settings described above The General Settings page has a different content as follows 1 General Settings for pickup group extension This group requires personal extension information and has the following components Display Name is an optional parameter used to recognize the EEI Ace tension Ada Multiple Bdensions Bulk Import caller Usually the display name appears on the called party s ss pho
427. r and might be used by a billing program for grouping the calls having the same Identity Code Attention It is highly recommended to secure PSTN and IP PSTN routing rules by selecting AAA Required options Unsecured routing rules may cause unexpected expenses The Check with 3PCC checkbox is used to request a 3PCC approval before placing a call with the specific routing rule When this checkbox is selected and the corresponding routing rule is used to place a call QX IP PBX sends a request to the call controlling application for the managing person to accept or reject the specific call it can be a popup window or any other type of dialog box depending on the call controlling application If the request is accepted the call will be placed Otherwise if the request is rejected the call will be skipped In case of no feedback from the call controlling application the call will be accepted after a timeout defined in the configuration of the call controlling application The Failover Reason s radio buttons indicate whether the system should use the next matching pattern if call setup with the current routing rule fails and allows choosing the reasons to be considered as a failover e None indicates that matching patterns should not be used regardless of the failover reason e Failover Reason s indicates possible failure reasons Failure reasons vary depending on the destination type selected on the previous page If the call cannot be establish
428. r is required for SIP the SIP address is requires and for PSTN a PSTN number is required Auto is used for undefined call types destination independent on whether it is a PBX number SIP address or PSTN number will be reached through Call Routing Table If this field is left empty the callers address will be implied as a callback destination The Callback Response Delay text field requires the delay in seconds after which the call back will be performed To Add an Authorized phone to the database Enter the desired Auto Attendant Settings page Select Edit Authorized Phones Database to enter the Authorized Phones Database page Press the Add button on the Authorized Phones Database page The Add Entry page will appear in the browser window Choose the call type and enter a caller address in the corresponding text field Select a Login Extension and the Automatically Enter Call Relay Menu checkbox if required Enable Call Back service if required and define a Call Back Destination in the same named field Fill in an optional Description in the appropriate field if required Press Save to submit the settings oe ee To Delete an Authorized phone from the database 1 Enter the desired Auto Attendant Settings page 2 Select Edit Authorized Phones Database to enter the Authorized Phones Database page 3 To remove an authorized phone s select one or more checkboxes of the corresponding records that should be deleted from the Authori
429. r not it is registered for each i Status SIP Registration Status extension and the registration date and time By clicking on the row heading the table will be sorted by the selected column When sorting A ascending or descending arrows will be displayed next to the column i Beray colmo Smo En ee h di j 741500 192 168 0 209 Yes 06 Aug 2014 11 19 48 ea Ing lll Status 101 7415101 192 168 0 209 Yes 06 Aug 2014 11 19 48 i F k a E i 2 7415102 192 168 0 209 Yes 06 Aug 2014 11 19 48 The links inside the table will link you to the Extensions Management page 5 O RE O VRT where the SIP registration settings may be altered 4 TSIO 1921680209 Ves 06 Aug 2014 11 1948 666 192 168 0 209 Yes 06 Aug 2014 11 19 48 The Detected Connection Type field displays the connection type QX IP An ha ll cc 7415702 192 168 0 209 Yes 06 Aug 2014 11 19 48 PBX currently is acting in direct connection or behind NAT If QX IP PBX is acting behind NAT the NAT machine IP address is also displayed Fig II 241 Status SIP Registration Status page The SIP Tunnels to Slave Devices and SIP Tunnels to Master Devices tables list the SIP tunnels between local and the remote QX IP PBX s see SIP Tunnel Settings The SIP Tunnels to Slave Devices table lists those tunnels where local QX IP PBX acts as a master The SIP Tunnels to Master Devices table lists those tunnels where local QX IP PBX acts as a slave IP Lines Registration Status The IP Lines Registration Status
430. r uploaded greeting messages It opens a file chooser window for immediate download to the users PC Please Note If you consider the Call History entries in the displayed tables to be important it is recommended to download them from the corresponding page prior to starting the Firmware Update e All pending events QX50 QX200 QX2000 SW Version 6 0 x 168 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide e User specific GUI states The following main processes will be stopped during the firmware update and will be restarted after the installation is completed Voice Software 0 Network Time Protocol Daemon 0 Network Interface Statistic Daemon o Dynamic DNS Daemon To update firmware manually select one of the following pages Upload Firmware or Get Firmware From Server For automatic firmware update select the Automatic Firmware Update tab Upload Firmware The Upload Firmware procedure is created in 3 pages In the first page of Upload Firmware the image file should be Overview Diagnostics System Lo selected ashboa Get Firmware From Server Automatic Firmware Update Firmware update Specify Image text field displays the selected image filename It is recommended to backup the configuration prior to upgrading the firmware You can do that right now by clicking the following link Download Configuration Ch oose File butto n use d to b rowse th e ima ge fil e Warning Make sure the Firmware Update proce
431. rd Dialing service for the selected timeslot s CAS Signaling Wizard Page 3 allows to set the destination for incoming calls to be routed to and to enable Cut Through and Automat Ringing Down services for signaling different from R2 all types Route Incoming Call to drop down appears when Both incoming and outgoing calls or Incoming calls only is selected from the Allowed Call Type list and allows selecting the destination where incoming calls should be routed The list contains all extensions of the QX Attendant and Routing agent The routing agent gives two kinds of call routing possibilities in user mode and one in network mode Choosing the Routing selection available in User mode only will request the caller to pass the authentication if enabled and will invite the caller to dial the destination number to connect the user within the QX Network Choosing the Routing with inbound destination number selection will automatically use the initially dialed number to connect the destination without any additional dialing When DID service is enabled in User mode only incoming calls can be only routed to the Routing agent with simple Routing and Routing with inbound destination number call routing possibilities PE Interfaces PE Interfaces QX50 0X200 0X2000 Manual II Administrator s Guide El T1 Trunk ISDN Trunk PSTN Gateways CAS Signaling Wizard DID Service Settings runk 1 192 168 74 127 5060 Selected Timeslots 1 Y
432. re may be configured aa Te Ibe TE ARESE Type CCS Eanes PR Interfaces Channel Selection preferred Y The Force Update option can be optionally used to apply new m A aes settings immediately The Restart option is used to bring timeslot s Eat ForeeUpdate Resta to the initial idle state on the both sides When applying one of these e AREA Echo Cancellation options any active traffic on the timeslot s will be terminated o a Timeslot 2 Timeslot 3 Channel Selection drop down list is used to select between the Preferred and Exclusive B channel selection methods For er Preferred channel selection the CO answers to the call request by ma the first available timeslot while for Exclusive channel selection CO Timestot 9 should feedback only by the timeslot used for the call request Times lot 11 Timeslot 12 Channel Selection Ordering drop down list is used to choose the B Timesiot 13 channels selection Ascending or Descending When Ascending selection is configured B channels will be defined starting from B1 to Timesiot 17 B23 B30 For Descending selection B channels will be defined from lt B23 30 to B1 If your CO PBX has Ascending B channels selection en configured it is recommended to use Descending B channels ae selection and vice versa Timeslot 23 Timeslot 24 Timeslot 25 Timeslot 26 Timeslot 27 Timeslot 28 Timeslot 29 Timeslot 30 Timeslot 31 Fig IT 122 Trunk CCS Signaling Se
433. receptionist feature on the QX IP PBX offers a variety of services to manipulate with multiple calls to keep the calls in the queue with the perspective to be answered by the receptionist and finally to be forwarded to the corresponding destination if needed The Receptionist service requires called extensions to use one of the following SIP Phones e Aastra 6730i e snom 320 e Aastra 6731i e snom 360 e Aastra 6735i e snom370 e Aastra67371 e snom720 e Aastra 6739i e snom 760 e Aastra 6755i 55i e snom 820 e Aastra 6757iCT 57iCT e snom 821 e Aastra 67571 57i e snom 870 e Aastra 91331 e Grandstream GXP 2000 e Aastra 9143i 33i Grandstream GXP 2100 e Aastra 94801 35i 0 Grandstream GXP 2110 e Aastra 9480iCT 35iCT 0 Grandstream GXP 2120 e Aastra 480i e Grandstream GXP 2124 e Aastra 480iCT e Grandstream GXP 2160 QX50 QX200 QX2000 SW Version 6 0 x 57 QX50 0X200 0X2000 Manual II Administrator s Guide e Polycom SoundPoint IP 650 e Grandstream GXP 2200 e Polycom SoundPoint IP 650 Pre 3 3 0 e Yealink SIP T 26P e Polycom SoundPoint IP 670 e Yealink SIP T 28P e Polycom SoundPoint IP 670 Pre 3 3 0 e Yealink SIP T 38G e snom 190 e Yealink SIP T 46G 0 snom 200 e Alcatel Temporis IP800 The following services are available to the receptionist e Call Queue e Extension Status e Call Interception e Voicemail Transfer e Multi Company Receptionist Call Queue This feature allows keeping multiple incoming
434. responding routing rule will be used Please Note If an extension does not have a profile specified here or the specified profile name is incorrect the default Voice Mail Settings of the extension will be used The Transport Protocol for SIP messages manipulation radio buttons group is available for SIP SIP Tunnel or IP PSTN destination types only and allows you to select the transport UDP TCP or TLS to transmit the SIP messages through The SIP Privacy manipulation radio buttons group is only available for the SIP and SIP Tunnel destination types and allows you to select the security of the SIP route by means of hiding or replacing depending on the configuration of the SIP server the key headers of the SIP messages used to establish the call e Default Privacy with this selection QX IP PBX specific SIP privacy will not be applied and all privacy will rely on the configuration of the SIP Server e Disable Privacy with this selection SIP call security will not be disabled and all headers of the SIP message will be transparently visible to the destination QX50 QX200 QX2000 SW Version 6 0 x 96 e C Pyg l QX50 QX200 QX2000 Manual Il Administrator s Guide e Enable Privacy with this selection SIP privacy will be specified for the corresponding route This selection enables a group of checkboxes in order to choose the key headers that are to be fully or partly hidden or replaced The Require Privacy checkbox selection is us
435. revious Finish Fig I1 106 FXS Gateway Configuration Wizard FXS Gateway Summary page 73 e C Pyg l QX50 0X200 0X2000 Manual II Administrator s Guide FXS Lines FXS On board Line Settings The FXS On board Line Settings page is used to configure QX lines and to define the caller ID detection type configure remote party disconnect indication and select the ringer type on each of them Additionally this page provides an option to enable Loopback diagnostics on the lines The Onboard Line Settings page shows the table Available Lines where all active lines of QX IP PBX are listed with their Attached Extension If the line is attached to an extension the corresponding extension number is displayed in this column otherwise none is E1 T1 Trunk ISDN Trunk PSTN Gateways Overview 5 FXS On board Diagnostic Loopback displayed if the extension is not attached to the line By clicking on FXS Lines the extension number the Extensions Management General Settings fh Interfaces Available Lines Caller 1D las Tee page will appear where the line attached to the extension can be arenai reconfigured Additionally the table provides information about the eS selected Ringer Type and Caller ID detection method that is configured for the selected line The caller ID detection method is different for various types of phones and can be found in the phone manual Fig II 107 FXS Lines Page When pressing on the line n
436. rmation can be displayed at a time the Line IP Line and FXO line ISDN or E1 T1 Trunk functional buttons are used to navigate through the information regarding other lines The Lines Status table displayed for FXS and IP lines includes a group of static and dynamic parameters Static parameters are always displayed Dynamic parameters only appear when an event takes place on the extension QX50 QX200 QX2000 SW Version 6 0 x 145 epygl Static Parameters Extension shows the extension number of the selected telephone line i eer mk n Status Lines Status Display Name shows the corresponding name Phone State may have the value On Hook or Off Hook For IP Line Status this field may additionally have Not Configured and Temporary Offline values Number of Active Calls shows the number of calls that are currently present on the phone Dynamic Parameters Call State shows the current state of the extension in voice mail in call waiting busy call out ring in etc Caller Party appears when a call is received and indicates the caller extension and the IP address or a phone number depending on type of call Called Party appears when a call is placed and indicates the destination extension and the IP address or a phone number depending on type of call Call Type shows whether the call is Internal or Extern
437. rn in the highest position of the rearranged list will be considered as the preferred one The second and subsequent matching patterns will be used if the destination refused the call due to the configured Fail Reason The Enable Disable functional buttons are used to enable disable the selected route s Disabled routes will have no effect Enabled routes will be parsed when initiating routing calls The State column in the Call Routing Table displays the current state of the routes enabled disabled Add starts the Call Routing Wizard where a new routing pattern may be defined The Call Routing Wizard is divided into several pages Page 1 displays the following components The Enable Record checkbox is used to enable the newly created routing rule By default this checkbox is selected so the newly created routing rule will be enabled But if you wish to create a routing rule for a later use disable it from this page The new routing rule will be added to the Call Routing Table but will be disabled and will not be considered when placing calls through the call routing unless it is enabled again The Destination Number Pattern text field specifies calls to which the rule should be applied If a call either inbound or outbound has a destination number that matches the specified pattern it will be completed according to the current rule A routing pattern may contain wildcards For the list of characters and wildcards allowed in this text field see ch
438. rn needs to be defined Enable ZeroOut checkbox enables the ZeroOut feature When this feature is enabled callers that have reached the ACD Group extension may accelerate the automatic redirection instead of holding in the extension s queue To activate this feature caller should dial O digit see Feature Codes while in the QX50 QX200 QX2000 SW Version 6 0 x 38 QX50 0X200 0X2000 Manual II Administrator s Guide queue of ACD Group extension The caller will then be automatically transferred to the destination specified in this page This selection activates the following fields to be inserted Redirect Call Type drop down list includes the available call types e PBX local calls between QX IP PBX extensions and the Auto Attendant e SIP calls through a SIP server e PSTN calls to PSTN e Auto used for undefined call types Destination independent on whether it is a PBX number SIP address or PSTN number will be reached through Routing The Redirect Address text field requires the destination address where the caller should be automatically forwarded to if activating the ZeroOut feature Upload new call queue welcome message allows updating the active call queue welcome message for the agents group played when a caller joins the agents group call queue downloading it to the PC or restoring the default one The Remove call queue welcome message functional link appears only when the custom call queue welcome message is already
439. rom and NAT Traversal Settings to the QX IP PBX will be routed through the NAT router Telephony NAT Traversal for SIP e Automatic with this selection system will analyze the QX IP PBX s WAN IP address and if it is in the IP range specified for local networks according to RFC the SIP traffic will be routed through NAT Otherwise if QX IP PBX s WAN IP address is outside the specified IP range no SIP traffic will be routed through NAT router e Force with this selection all the SIP traffic will be routed through the NAT router e Disable with this selection no SIP traffic will be routed Figli 158 General NAT traversal Settings page through the NAT router SIP Parameters The SIP Parameters page is used to configure NAT specific settings for SIP and offers two independent groups of settings UDP Parameters Manipulation radio buttons allow you to select the type of mm connection over NAT NAT Traversal Settings Selecting Use STUN will switch to automatic discovery of Mapped settings for the SIP UDP traffic over NAT STUN settings pjasi ene Use STUN are configured on the STUN parameters page see below iia Selecting Use Manual NAT Traversal allows you to manually es usada 212 158 54 157 define the mapped settings for the SIP UDP traffic over NAT Mapped TLS Host Mapped Port 210 65 14 60 1589 Mapped Host requires the IP address of the mapped host for id SIP UDP traffic
440. rt number will appear if this condition is not met The port range must consist of digits only otherwise the error Incorrect Port Range only Integer values allowed will appear The difference between Max and Min RTP ports should be 100 ports or less according to the system s capabilities otherwise the corresponding warning appears RTP RTCP Port ranges cannot include the defined SIP UDP ports see SIP Settings otherwise an error message will appear QX50 QX200 QX2000 SW Version 6 0 x 111 QX50 0X200 0X2000 Manual II Administrator s Guide pya Enable RTCP Support enables Real Time Control Protocol support and allows for the RTCP packets transmission RTCP protocol is used for monitoring the RTP streams and changing RTP characteristics depending on Network conditions The RTP Settings Edit Entry page offers a drop down list and a checkbox 1 i VW RTP Setti Edit Ent Packetization Interval contains possible values in milliseconds A A to be configured for the selected codec G 711u PCM audio coding standard 8 kHz sample rate 8 bits 64 kbit s data rate The Enable Silence Suppression checkbox selection enables voice ENERO activity detection for the selected codec Fig II 165 RTP Settings Edit Entry To Edit Codec Parameters 1 Select the codec from the Codecs Table that is to be edited 2 Press the Edit button on the RTP Settings page The Edit Entry page will appear in the browser window
441. rt procedure has been successfully completed The first IPSec Connection Wizard page Add IPSec Connection has the Connection Name text field that requires a new mandatory IPSec connection name If the text field is not filled in the error message otherwise an error will occur Error Incorrect connection name will appear Please Note The input in the Connection Name field should only be in Latin characters otherwise an error occurs and IPSec connection cannot be created The Peer type drop down list is used to choose the remote machine type for the IPSec Connection to be established If the list does not include the required type of machine choose Other IP Routing Overview PPTP L2TP The VPN Network Topology drop down list allows you to select dasi li aid iii the location of the peers participating to the VPN connection The following options are present in the list en aa nen Connection Name to7440M32x e This device lt gt Peer direct connection between QX IP PBX nen and a peer VPN Network Topology Quadro lt gt Peer e This device lt gt Internet lt gt Peer connection between QX IP PBX and peer over Internet e This device lt gt NAT lt gt Internet lt gt Peer connection between QX IP PBX and peer over Internet through QX IP PBX provider s NAT e This device lt gt Internet lt gt NAT lt gt Peer connection between QX IP PBX and peer over Internet through peer provider s NAT
442. rver the transport protocol TCP is automatically used regardless of the Transport Protocol for SIP messages radio button selection Attention By choosing the Use External Voice Mail option some internal voice mailbox services may become unavailable Instead the services of the external voice mail server will become available to the user Please consult with the external voice mail server administrator before enabling this option QX50 QX200 QX2000 SW Version 6 0 x 31 e C Pyg l QX50 QXK200 QX2000 Manual Il Administrator s Guide 7 Class of Service Settings The Class of Service Settings page is used to assign the defined classes to a PBX extensions To use Class of Service feature it should be enabled from the Class of Service page Class of Service feature allows to specify which PBX Conference E Extensions extensions can use which routing rules to make a call For example n if an extension is not assigned to a certain class of service and an ae a tee SENNOR 103 attempt is made to place a call from that extension using routing danced Settings Goto Class of service rule with the Class of Service feature enabled then Number en se ee dialed does not exist message will be played to the caller none Extensions Add Extension Add Multiple Extensions Bulk Import Extensions Management Edit Entry 4 Special 6 Special 7 The Go to Class of service link leads to the Class of Service page where the class of serv
443. ry page 136 QX50 0X200 0X2000 Manual II Administrator s Guide VPN Configuration A VPN Virtual Private Network is established to connect two local networks intranets securely over the Internet securely The VPN routers manage authentication between servers and clients and handle data encryption for the connection Only authorized users may access the network and the data exchange cannot be intercepted The VPN Configuration page is not available for QX2000 VPN connections are in many ways like every Internet connection they are based on IP addresses which means the concerned VPN gateways must authenticate the IP addresses of their respective partner s VPN gateways Each time a specific VPN is to be established usually the same IP addresses are expected This will not create problems if both VPN partners have fixed WAN IP addresses There may be circumstances reasons to prefer dynamically allocated IP addresses To enable devices that use a variable IP address as part of a VPN they are turned into Road Warriors For example at this point they are able to reach their corporate network via authentication at the company s VPN gateway device This VPN gateway device must have a fixed IP address for Internet access Every VPN needs at least one VPN gateway with a fixed IP address The partner devices of a VPN must have different WAN IP addresses and if they are connected to local area networks these LAN s must have different IP
444. s 5 hour 30 min 20 sec Free Space Total Space 5 hour 30 min 20 sec 0 sec 0 sec 5 hour 30 min 20 sec 5 hour 30 min 20 sec 11 hour 0 min 40 sec 11 hour 0 min 40 sec Fig II 239 Status Memory Status page Hardware Status The Hardware Status table displays a list of the hardware devices and parts present and currently available on the QX IP PBX The hardware device version number and additional comments about its state are indicated here QX50 QX200 QX2000 SW Version 6 0 x fil Status System Status General Network Lines Memory SIP Registration IP Lines Registration Status Hardware Status LAN Ethernet 10 100 Mbps Link is up 100Mb s full duplex WAN Ethernet 10 100 1000 Mbps Link is up 100Mb s full duplex SD Card Mounted Model SU04G Capacity 3 7 GB FXS 2 Ports Available FXO 4 Ports Available RAM memory 483 83 MB Available License Fig II 240 Status Hardware Status page 147 QX50 0X200 0X2000 Manual II Administrator s Guide SIP Registration Status The SIP Registration Status is a table displaying the SIP registration information of the QX IP PBX extensions The table contains a list of all the registered extensions of the QX IP PBX System Status Events SIP registration name for each extension addresses of SIP servers where m Ga E E Ge Pines Regtaon cen they are registered if applicable whether o
445. s Feature Keys This page lists all features that may be activated by a software key characterized by a Feature Description and provided with its Status e No Key Found the feature is currently not available e Reboot Needed the feature key has been entered and QX IP PBX needs to be rebooted e Activated the feature is now available on the QX IP PBX System Security Licensed Features Redundancy Language Pack Following features may be activated via the software key Fee ors ter Features e Debug enables SSH connection towards the QX IP PBX een ere ee for debugging purposes i had Upgrade Name Additional Features e 3PCC Support enables Third Party Call Control feature Debug En Activated on the QX IP PBX The feature allows the call controlling applications running on a user PC to remotely initiate and 3pcc Support ACD Support Support for Third Party Call Control Support for Automatic Call Distribution Free trial Expiration Date 2014 08 06 23 00 Free trial Expiration Date 2014 08 06 23 00 handle calls on the QX IP PBX and to subscribe for certain event notifications from the QX IP PBX Free trial Expiration Date 2014 08 06 23 00 Barge In Support for Barge In DCC Pro Support Full support for Pro level Desktop Communication Console Free trial Expiration Date 2014 08 06 23 00 DCC Basic Support Full support for Basic level Desktop Communication Console Free trial Expiration Date 2014 08 06 23 00 e AC
446. s 1 1 3 3 0 8 1231 100 150 asd 1 3 12 31 123 11 15 3 2 1 1 1 3 The list of patterns Matching digits 1231 4 1 1 3 3 0 8 2 100 150 asd 1 1 3 12 31 123 3 11 15 3 1 2 1 A 1 1 1 3 0 Matching digits ainia pA PrP rN wool gt Oo Step 2 The two groups of patterns are arranged separately from each other by the total number of matching digits inside and outside the braces brackets in the descending order Criterion 2 The patterns that contain the same number of matching digits are grouped into sub lists Step 3 The new sub lists are arranged separately from each other by the number of matching digits outside the braces brackets Criterion 3 The patterns that contain the same number of matching digits are grouped into sub lists The Best Matching Algorithm will stop after executing step 3 as no new sub lists are formed The resultant list of prioritized patterns will be the following 104 QX50 0X200 0X2000 Manual II Administrator s Guide Entering SIP Addresses Correctly Calls over IP are implemented based on Session Initiating Protocol SIP on the QX IP PBX When making a call to a destination that is somewhere on the Internet a SIP address must be provided The display name and the port number are optional parameters in the SIP address If a port is not specified 5060 will be set up as the default one The range of valid ports is be
447. s a VPN Gateway en Dynamic IP Roadwarrior Static IP Remote Gateway Remote Gateway 192 168 74 40 Y Quadro lt gt Remote Gateway Y Local Subnet lt gt Remote Subnet If Dynamic IP RoadWarrior is selected the Remote Gateway IP Address text field will automatically generate the value any to allow access independent from the sending IP address tecaswnee mz po Remote SubnetIP 172 30 0 Selecting Static IP Remote Gateway requires entering the IP address or the hostname of the remote QX IP PBX or another VPN gateway device in the Remote Gateway text field Previous Please Note The Static IP Remote Gateway selection is not possible if this Gateway is positioned behind NAT since the IP address of the remote gateway is not reachable directly in this case Fig II 227 IPSec Connection Wizard IPSec Connection Properties page QX50 QX200 QX2000 SW Version 6 0 x 139 QX50 0X200 0X2000 Manual II Administrator s Guide This device lt gt Remote Gateway allows access from the local QX IP PBX to the remote VPN gateway local subnet and remote subnet are not included This includes management access The checkbox is disabled when This device lt gt NAT lt gt Internet lt gt Peer or This device lt gt Internet lt gt NAT lt gt Peer the is selected from the VPN Network Topology drop down list on the first page of the IPSec Connection Wizard Local Subnet lt gt Remot
448. s added to the Park Access List while the phone at the remote location is added to the Park Access List of that call park extension While being on a call with user A the QX IP PBX user dials the appropriate calling code As a reply QX IP PBX will play the extension 77 and SIP username 892220 to the QX IP PBX user The user A goes on hold The QX IP PBX user moves to a remote location and makes a call to the call park extension The QX IP PBX user enters call park extension s password and resumes the conversation with user A To make a Directed Call Park To make a Directed Call Park the QX IP PBX user which has been previously added to the Park Access List for at least one of the available Call Park extension on the QX IP PBX should place the current call on hold and then dial the Call Park extension number within the five second timeout see Feature Codes in Manual III Extension User s Guide Attention If the five second timeout is exceeded then the QX IP PBX will consider it as an attempt for retrieving the parked call The Call Park extensions can be mapped directly to IP phones or simply announced via paging through the IP phones or analog paging system Calls can be easily parked by placing the current call on hold and then pressing the park button followed by the desired extension This can be further simplified if the desired Call Park extension is already mapped to the phone then the user will just press that specific park key and the c
449. s are adhered to e To prevent fire or shock hazard do not expose your Quadro or QX to rain or moisture e To avoid electrical shock do not open the Quadro or QX Refer servicing to qualified personnel only e Never install wiring during a lightning storm e Never install telephone jacks in wet locations unless the jack is specified for wet locations e Never touch non insulated telephone wire or terminals unless the telephone line has been disconnected at the network interface e Use caution when installing or modifying cable or telephone lines e Avoid using your Quadro or QX during an electrical storm e Do not use your Quadro QX or telephone to report a gas leak in the vicinity of the leak e Anelectrical outlet should be as close as possible to the unit and easily accessible Emergency Services The use of VoIP telephony is made available through IP networks such as the Internet and is dependent upon a constant source of electricity network availability and proper operation of the equipment If a power outage network disruption or equipment failure occurs the VoIP telephony service could be disabled User understands that in any of those events the Quadro or QX may not be able to support 911 emergency services and further such services may only be available via the user s regular telephone line or mobile lines that are not connected to the Quadro or QX User further acknowledges that any interruption in the supply or delivery of electricity netw
450. s from QX IP PBX s extensions A ee e SIP calls through a SIP server calType SPY N 11369 sip epygi loc Address e PSTN calls from global telephone network wildcard supported e Auto used for undefined call types The destination independent on whether it is a PBX number SIP address or PSTN number will be parsed through the Call Routing Table Save Fig IT 43 Access List of Extension Add Entry page for Pickup group The Address text field is used to define the address to be included in the Access List table The value in this field is strictly dependent on the Call Type defined in the same named drop down list If the PBX call type is selected the QX IP PBX extension number should be defined in this field For the SIP call type the SIP address should be defined for the PSTN call type the PSTN user number should be defined here The Action drop down list is used to select the defined user s permissions allow or deny to use the pickup service for the extensions included in the Pickup Group Call Park Extension Settings For Call Park extensions the Extensions Management Edit Entry page consists of General Settings SIP Settings Advanced SIP Settings Park Access List and Retrieve Access List pages The SIP Settings and Advanced SIP Settings pages are the same as for the regular extensions see User Extension Settings 1 General Settings for call park extension This group requires personal extension informa
451. s option is enabled and the call is held the caller gets dial tone Otherwise there will be no dial tone after pressing Hold Do not Disturb this selection allows you to manipulate with the IP phone DND service When the 72 is selected from this list the DND service of the IP Phone and the DND service of the QX IP PBX for the corresponding extension will be activated when enabling the DND service from IP Phone This option is recommended When keyeventF_DND is selected only DND service of the phone will be activated when enabling the DND Record Missed Calls when this option is selected the information about the missed calls will be displayed on the IP Phone Any parameters not listed above or parameters defined in this page for other IP phone models can be found in the user s manual of the corresponding IP phone Please Note Save changes before moving among the configuration pages IP Phones Logo The IP Phones Logo page is used to upload a custom logo for the IP Phones This page contains only those IP phones for which QX IP PBX supports the custom logo upload The uploaded custom logo will be visible on the display of the IP phone The Enable checkbox is used to enable the custom logo for the selected IP phone model s The Choose File button opens the file chooser to select the custom logo file QX50 QX200 QX2000 SW Version 6 0 x IP Lines IP Line Settings IP Phones Logo ft interfaces Z Enable Logo E1 T1 Trunk
452. s quizzing the calling customer to determine the set of required skills and when handing over the call to ACD module it passes the set of skills required by calling customer Having that set the ACD module calculated the composite skill grade of each AG in the system and sends the call to AG having the highest value of composite skill grade The call in AG is handled according to call distribution type configured with that AG QX50 QX200 QX2000 SW Version 6 0 x 61 QX50 0X200 0X2000 Manual II Administrator s Guide For example if the call distribution type of AG is skills based then AG will try to connect the call to the agent having the highest composite skill grade and if it is not answered within timeout the AG will try to connect to the next agent with the highest grade etc If the call distribution type is something else then AG will distribute the calls according to that distribution type don t taking into account the skill grades of the agents In case if the call is received on agent group bypassing ACD s IVR and the skills based call distribution is selected for that agent group the agent group will consider the full set of skills when making decision on which agent to make a call first In other words since there is no required set of skills received from IVR then the agent group will consider the full set of skills summing up all skill grades of agent To simplest way to build the VoXML script for IVR is using the text of the
453. s set on the System Logs Settings page The text field on the left side is dedicated for support personnel only and is used to search a custom log not listed on this page To do so insert a required log name to the text field and press Show Custom Log functional button If the user has used Logs Collection OO feature code after or during from another phone connected to the same QX IP PBX the call a special log file will be generated containing the details of that call and few last calls done in the system This log file will be internally kept in the system until the next time someone used the Logs Collection feature code again The collected logs will be a part of the System Logs when user downloads them next time so it can be reviewed by appropriate support staff This could be used to collect the logs at the exact moment when a problem has happened System Logs Settings amp Maintenance Diagnostics v Pending Events System Logs User Rights Backup Restore System Logs Settings Remote Logs Settings Logs Archive System Logs SIP User Agent Call Controlling Conference Controller Policy Server Media Stream Voice Mail System SIP User Agent SIP Registration FOIP DSP System Messages CN AC Show Custom Log 14 46 46 Try to send SIP message 10 09 2014 10 46 46 474 GMT UDP 900 bytes buff size 0 from 192 168 74 206 5060 to 192 168 70 25 5060 een a EERE AME Sr SIP 2 0 200 OK Via
454. s will be displayed in the following format username Proxy sipserver port If no SIP registration server or SIP server port is defined corresponding information will not be included in this column If no username is defined the extension number will be displayed instead e Percentage of System Memory indicates the user space in percentages configured for each extension The actual available duration in minutes for the extension voice mails uploaded recorded greetings and blocking messages is also displayed here The available minutes corresponding to the selected user space are dependent on the Voice Recording codec selected from the Voice Mail Common Settings page For example for the same amount of marked out user space selection of the G726 voice recording codec will provide more space for voice mails and user defined voice greetings than the G711 codec selection e External Access indicates whether the GUI Login 3pcc Click2Dial login or Call Relay options are enabled on the extension e Codecs column lists the short information full information is seen in the tool tip about extension specific voice Codecs Extension codec s can be accessed and modified by clicking on the link ofthe corresponding extension s Codecs The link leads to the Extension Codecs page Clicking on each user extension in the Extensions table will open the extension specific Your Extension menu see Manual III Extension User s Guide The Pickup Group
455. sages Greeting Menu Message O Da Dial J Incomin Outgoin rng AA Number Greeting a neds Your Name Out of Office Me Follow i AA Number Blocking Blocking in case of Message Message Me Welcome in case of Message Message multiple AAs Message on the OX IP multiple AAs on en the QX IP PBX O O z O 0 u isten to 1 Listen to Listen to Listen to Listen to Listen to Current O Current Current Listen to Current i Current Current Find Listen to AA i Incoming Outgoing i Current Menu Message Greeting Blockin Blockin Name Out of Office Me Follow AA Greeting Message 5 5 recorded Message Me Welcome Message Message e Message 2 Record a New E Record a New O 2 2 AA Menu e AA Greeting Record a Record a amp ee n Message Record a E Record a Universal Universal Universal Record a Univer a Incomin Outgoin Universal Umyer a Fina Greeting ie ane Out of Office Me Follow Blocking Blocking Name Message Message Me Welcome Restore Default Message Message Restore Default Message AA Menu AA Greeting Message Restore hectori Restore Restore System Restore System System Restore System Default System Default Default i System Default eae Outgoing Teenie Default Find Greeting 1m8 Blocking Out of Office Me Follow Blocking Name Message Message Message Me Welcome Message Message Stop Recording Stop Recording Q O O O Stop or Playback or Playback Stop E Stop Stop Recording Stop Stop Recording or
456. sconnect the call The following options can be configured too e Any input other than in the list above allows configuring the action taken when the caller makes a selection other than options listed in the User Input table If it is configured to No Action then the timer for No Input will reset and it will be counting the No Input time again e No input allows configuring the action taken when the caller doesn t enter anything during the certain period The No Input timeout is equal to Welcome message duration Delay after message Recurring message duration Play Count Play Count Interval If there is no input during that time the action specified for No input will take effect The Dial Timeout specifies the period of time to determine when the user has completed dialing and to begin to process the call The timer will start after the last digit or symbol is entered If the key has been pressed then the call will be processed immediately Incorrect number handling link opens the Edit Incorrect Number Handling page which is similar to Edit Option page to configure the action taken when the user has selected a destination that resulted in a failed call such as an invalid extension number Incorrect number handling link will open the page to configure the action taken when the user has selected a destination that resulted in a failed call such as an invalid extension number Please Note The Incorrect number handling will be act
457. se that do not have symbol in the Call Routing Table When calling from local extensions the calling number for local extension is sipnumber ip_address_of_QX e g 20233 192 168 35 25 only the sipnumber part of the pattern will be parsed among other entries with symbol in the Call Routing Table Best Matching Algorithm All calls through and within a QX IP PBX are made according to call routing patterns that specify a destination based on a dialed number When a user dials a number to make a call the QX IP PBX matches the dialed number against the existing patterns that are specified in the Call Routing table If the dialed number matches only to a single pattern this pattern will be used to set up a call If several patterns have been found to match the number the QX IP PBX uses the Best Matching Algorithm to prioritize the matching patterns Once the patterns are prioritized the pattern with the highest priority will be used as a preferred route for call setup The successive patterns will be used only if the destination specified by a higher priority pattern is unreachable To prioritize the matching patterns the following criteria are sequentially applied to matching patterns The criteria are ordered by their priorities Each consecutive criterion is calculated only for the patterns that take the same value for the preceding criteria that is Criterion 3 is calculated only for patterns that take the same value for Criterion 1 and Criterion 2
458. ser s Guide Intercom Service The AAA Required checkboxes are used to choose one or more of the following Authentication Authorization and Accounting AAA settings e Local Authentication with this checkbox selected callers will need to pass authentication through the Local AAA Table when dialing the current pattern e RADIUS Authentication and Authorization this checkbox is present when a RADIUS client is enabled With this checkbox selected callers will need to pass the authentication through RADIUS server see above when dialing the current pattern o The RADIUS Accounting checkbox is accessible when the RADIUS Client is enabled With this checkbox selected no authentication will take place but CDRs call detail reports of the calls made through this routing record will be sent to the RADIUS server This checkbox selection enables the Client Code Identification checkbox If the authentication is configured based on the caller s address callers will pass the authentication automatically otherwise they will be required to identify themselves by a username and a password e The Client Code Identification checkbox selection activates the code identification feature a caller after dialing the destination phone number may optionally enter and then an Identity Code An Identity Code is an arbitrary digit string entered by the user to identify a specific call or call group The Identity Code is sent with CDR to the RADIUS serve
459. set or they can upload a wave file from the extension s Account Settings page see Manual III Extension User s Guide QX50 QX200 QX2000 SW Version 6 0 x 56 QX50 0X200 0X2000 Manual II Administrator s Guide The Custom Greeting column in the Extensions Directory table displays whether or not a custom greeting user s name is recorded or uploaded Users cannot be accessed through the Extensions Directory and it is implied as being an inactive entry in the event a custom greeting is not recorded or uploaded Warnings will be seen in the Extensions Directory table for Extensions Directory Settings Extensions a Add Edit Delete MoveUp Move Down Name Call to Custom Greeting Description Inactive entries Extension numbers In the Extensions Directory JAMES 4 Warning user s name recording is absent so this entry is not active Team Leader table are made aS a link to move to the corresponding extension s MICHAEL Varning user s name recording is absent so this entry is not active Support Engineer Account Settings page see Manual HI Extension User s Guide ai e user s name recording is absent so this entry is not active Enge This helps the administrator access the extension s settings page where a custom greeting can be manually uploaded Fig II 80 Extension Directory table Move Up and Move Down are used to move the selected record one level up or down in the Extensions Directory table The sequence
460. shire ATL 5000 CISCO 7960 CISCO SPA525G2 CISCO SPA303 CISCO SPA501G CISCO SPA509G Fanvil C58 C58P Fanvil C62 C62P Fanvil F52 F52P Grandstream BT100 Grandstream BT200 Grandstream GXP1400 Grandstream GXP1405 Grandstream GXP1450 Programmable Keys Configuration QX50 QX200 QX2000 SW Version 6 0 x Grandstream GXP2140 Grandstream GXP2160 Grandstream GXP2200 Grandstream GXV3140 Grandstream GXV3175 Grandstream HT286 Grandstream HT386 IpDialog SipTone II Linksys SPA921 Linksys SPA922 Linksys SPA941 Linksys SPA942 Linksys SPA2002 Linksys PAP2T Panasonic KX UT136 Panasonic KX UT123 Panasonic KX TGP550T04 Polycom SoundPoint IP 300SIP Polycom SoundPoint IP 330SIP Polycom SoundPoint IP 331SIP Polycom SoundPoint IP 335SIP Polycom SoundPoint IP 450SIP Polycom SoundPoint IP 501SIP Polycom SoundPoint IP 550SIP Polycom SoundPoint IP 601SIP Polycom SoundPoint IP 650SIP Polycom SoundStation IP 5000 Polycom SoundStation IP 6000 POLYCOM VVX 1500 Polycom VVX 300 310 Polycom VVX 400 410 POLYCOM KIRK wireless server 6000 POLYCOM KIRK wireless server 300 snom 320 snom 360 snom 370 snom 710 snom 720 snom 760 snom 820 snom 821 snom 870 snom M3 snom PA1 snom m9 snom MeetingPoint Swissvoice IP 10S Telematrix IP550 Spectrum Plus Telematrix IP 3300 Telematrix IP9600 MWD5 Thomson ST2030S Yealink SIP T19P Yealink SIP T20P Yealink SIP T21P Yealink SIP T22P Yealink SIP T26P Yealink SIP
461. should be selected appears Connection RSA Key Management Start Stop Add Edit Delete Restart All Active Connections Attention It is not recommended to simultaneously start a static ARTER and a dynamic connection configured to use the same secret key A dynamic connection may capture the static connection peer and vice versa depending on which connection established first to7435M12Li Stop disconnects the selected IPSec connection The state of the IPSec connection will change into Stopped If no record is QX50 QX200 QX2000 SW Version 6 0 x 137 e C Pyg l QX50 QX200 QX2000 Manual Il Administrator s Guide selected the error message One Record should be selected will Fig l 223 IPSec Configuration Connection Settings page appear More than one record may be selected at a time to be stopped Add leads to the Add IPSec Connection wizard where a new IPSec connection can be defined and specified The wizard provides several pages Edit leads to a set of IPSec Connection Properties pages to modify the parameters of the selected IPSec connection The page includes the same components as the Add IPSec Connection page To operate with Edit only one record may be selected otherwise an error message One row must be selected appears Restart All Active Connections restarts all active IPSec connections The State of these IPSec connections will turn into Connected or Activated if the resta
462. sion of the Licensed Materials to a third party in violation of this Agreement You agree that upon such termination you will return the Licensed Materials to the Licensor at its request 4 No Unauthorized Copying or Modification The Licensed Materials are copyrighted and contain proprietary information and trade secrets of the Licensor Unauthorized copying modification or reproduction of the Licensed Materials is expressly forbidden Further you may not reverse engineer decompile disassemble or electronically transfer the Licensed Materials or translate the Licensed Materials into another language under penalty of law 5 Transfer You may sell your license rights in the Licensed Materials to another party that also acquires your Quadro or QX product If you sell your license rights in the Licensed Materials you must at the same time transfer the documentation to the acquirer Also you cannot sell your license rights in the Licensed Materials to another party unless that party also agrees to the terms and conditions of this Agreement Except as expressly permitted by this section you may not transfer the Licensed Materials to a third party 6 Protection And Security Except as permitted under Section 5 of this Agreement you agree not to deliver or otherwise make available the Licensed Materials or any part thereof to any person other than the Licensor or its employees without the prior written consent of the Licensor You agree to use your b
463. splayed on the bottom of all pages For some events the LED will start flashing after a delay e Send Mail an e mail notification about the new event on the QX IP PBX will be sent to the e mail address specified in the Mail Settings page e Send SNMP Trap SNMP notification will be sent to the traphost s listed in the SNMP Trap Settings table see SNMP Trap Settings e Send SMS SMS notification about the new event on the QX IP PBX will be sent to the mobile phone specified in the SMS Settings page Actions that are not allowed for the selected event like mail notification if the PPP link is down or the mail server has been configured improperly are hidden For multiple events editing actions that are not appropriate for least one of the selected events will also be hidden Please Note In case of an IDS Intrusion Detection System intrusion alert only the first possible intrusion in each 10 minute period will initiate an event This helps to avoid flooding the System Events table and flooding the user with various intrusion alerts that result from each possible Denial of Service attack When these events are displayed in the System Events table the user can receive detailed information about the intrusions through a link to the IDS log list If QX IP PBX cannot receive an IP address from the DHCP or PPP servers or cannot register an extension on the SIP or Routing servers or cannot reach an NTP server it raises only one event for the entire
464. sponding checkbox and press Change Password to AS open the corresponding page us Overview ro User Rights Management Maintenance Fig II 273 User Rights Management Users page For Administrator or Local Administrator account the Change Password page contains two parts one for GUI Access Password the other one for Phone Access Password O ew User Rights Backup Restore Firmware The GUI Access Password offers the following components ues o Change Password e The Old Password text field is only present when modifying the Administrator account password and iia requires the current password of the Administrator An A a arene error message prevents entering the wrong password Old Password New Password Maintenance e The New Password text field requires a new password Confirm New Password arme for the Administrator or Local Administrator e Reentering the new password in the Confirm New Password text field will confirm the new password The New Password field is checked against its strength and you may see how strong is your inserted password right below that field Fig II 274 Change Password page Please Note The password can consist of numeric values and symbols Up to twenty 0 20 digits and symbols are allowed The Phone Access Password offers the following components e The Old Password text field is present when modifying the Administrator account password and requires the current password
465. ss indication with in band information e on call acceptance The Calling Party Type of Number drop down list allows you to select the type identifying the origin of call The Called Party Type of Number drop down list allows you to select the type identifying the subaddress of the called party of the call The Called Party Numbering Plan and Calling Party Numbering Plan drop down lists correspondingly indicate the numbering plan of the called party s and calling party s number The Incoming Called Digits Size text field indicates the number of received digits in a range from 0 to 255 required to establish a call When this field has a 0 value the system uses either the timeout defined in the T302 field or the Sending Complete Information element messages to establish a call Independent on the value in this field Sending Complete Information element and the pound sign always result in call establishment The Generate Progress tone on IP checkbox selection will generate the progress tone to IP When Generate Progress Tone to PSTN PBX checkbox is selected QX generates ring tones to callers during ISDN call dialing This feature is mainly applicable to 2 stage dialing mode Enable CLIR Service checkbox selection enables Calling Line Identification Restriction CLIR service which displays the incoming caller ID only if Presentation Indication is allowed on the remote side Otherwise if CLIR service is disabled caller ID will be unconditionall
466. ss is not disrupted until it is completed A power down while upgrading may cause serious damage The update process takes about 5 minutes Normal operation will be stopped during that time Pressing Save will start uploading the image file to the board n and the next page will display results and verification of the Progress uploading lt lt lt image being burned Cancel Uploading The Cancel Uploading button appears when the update Seeciyimege Gina megesin procedure starts and it is used to stop it Save Fig II 283 Firmware Update page This page displays non editable information about the image validity The Image Check field will display invalid if the Overview Diagnostics Systemlogs User Rights image does not correspond to the hardware version SS GaSe Firmware update The Current Software Version field shows the old software version The New Software Version field shows the new It is recommended to backup the configuration prior to upgrading the firmware You can do that right now by clicking the following link Download Configuration Warning Make sure the Firmware Update process is not disrupted until it is completed version of the software image A power down while upgrading may cause serious damage The update process takes about 5 minutes Normal operation will be stopped during that time i i i z Maintenance This page needs to be confirmed in order to continue image ares updating If
467. st streaming The RTP channels are created from RTP Streaming Channels page QX50 QX200 QX2000 SW Version 6 0 x 39 e C Pyg l QX50 QXK200 QX2000 Manual Il Administrator s Guide 3 ACD Agents Table Settings This group is used to configure agents in the ACD group and has the following components PRA Extensions Management Edit Entry The ACD Agents Table lists all agents in the corresponding ACD GoBack group and their statuses ACD Agents Table 700 Add opens the Add Entry page where a new agent may be added sdranced Stings e Naor up we Dome to the group The Add Entry page contains the following ings Agent 1D m components ACD Agents Table 103 Go To Codec Settings Fig ll 55 Extensions Management Edit Entry ACD Agents Table page ACD Agent ID text field requires the name of the agent previously created from the Agents table of ACD Management n pean OS Skills Agents Agents Table of Group 700 Add Entry O Go Back Agent Status drop down list requires the actual status of the agent The following values are available in this list Extensions ACD AgentID 138 Y e Online the agent is logged into agent group and deere Status Giner available for receiving the calls from that group Enable wrap up Timeout 30 e Offline the agent is not logged into the agent group and cannot receive the calls from that group The same agent still can receive the calls from the other
468. stem login success Firmware update failed Firmware update success Fig II 245 Event Settings page epygl The Edit Event Settings page offers the following input options Application displays the application the event refers to Multiple is on shown here if more than one event has been selected for the action EditEvcat salinos assignment Application SYSTEM Name displays the name of the event Multiple is shown here if scription the device has been successfully started after reboot more than one event has been selected for the action assignment i Display notification Description displays additional information about the event ee Multiple is shown here if more than one event has been selected for a send sus the action assignment Action offers radio buttons to choose one of the actions to notify the QX IP PBX administrator when an event s takes place The following actions can be available Fig II 246 Edit Event Settings page e Display Notification A notification link will be displayed on the bottom of all pages and a record is added into the Events table The notification is executed as a link Please Check your pending events The link leads to the System Events page This action also will take place if Flash LED or Send Mail has been selected even if not specifically selected e Flash LED available only for QX50 QX200 The flash LED ORANGE will blink every second and a notification will be di
469. system memory e Video Conferencing activates the Video Conferencing feature on the system This allows up to 16 person video conference calls on QX200 up to 8 person video conference calls on QX50 and up to 104 video conference calls on QX2000 The other participants of conferences can use only audio connection To enter a Feature Key click Add A page with the Feature Key text field is opened Enter the key and press Save The status of the Overview Basic Setup selected feature entry will change to Reboot needed Reboot the QX dace IP PBX and the feature will receive the status Activated ah ae O Go Back To receive a Feature Key register the QX IP PBX device and send a corresponding request to Epygi s Technical Support This request must include the Unique ID that is displayed in the Features page above the features list ure Key yUTHmZVOC3Bq0uzme3dz4AXt89KjKALg Fig II 17 Features Add page Free Trial Activation This page allows activating the QX IP PBX optional features for a trial This page lists all QX IP PBX features that may be activated for a trial characterized by a Feature Description and provided with its Status Expiration Date Time used to specify the trial period Upon expiring the specified period the QX IP PBX will reboot and trial features will disable User has to select the appropriate checkboxes under Activate column specify the needed count under Count Column and save The QX IP PBX
470. t eaaa aa aE nana 102 Ete ring AP Aa a e er a A E O E aN imeNatamdenen 105 PP PA O o PS PE O OO E E E er E 105 Class OF ale A o o E o PP rt Eo EEE gt 106 Call Recording Ss ie 107 NAT Traversal SQ ea 109 General Si NN pases ete E Ao e 109 SIP Parameter S O O uo o o PR E SU oi e 109 RIP ParameteiS anita 109 STUN Parameters aon eer ere ree ree rt eer eee A A E AE EEEE A E ee re eae eres Pre ee ere re 110 NAT EXCIUSION a a ae ee ne ee ee a 110 RTP SQU GS conan 111 PP _ nen 112 ell ra YES dos mo PORC a A o A o o 113 A a PP e o o A 113 Advanced SENES os 113 Vatce MA Common seta aida 113 KEP Skea chatea aparta aparato 114 cs o A a E me nnn A 114 SPCC SEINES A no PP 115 RADIOS Ciento econ Sanare 115 PPP oe 117 CAMU Nociones 117 EXE WA e A A A E E A E 118 AA O 119 Firewalland NNW PAPA o ao E 119 Advanced Fire Wall Se 025 aan 119 IDS TO ars ias 119 aia Ann e O 120 View AL PIGe ie RUS aii 120 hicoming Trate Porron Ward 121 A PR o aU OA 121 Management ACCESS etnies mittee ek ti came ate Ri ea cs ce nc eS ened ee ne cc ace inc nce a Cen cate nce mee ene meee acm pesca 121 Nc AMA o Ao oo E eee 121 QX50 QX200 QX2000 SW Version 6 0 x 5 e C Pyg l QX50 QX200 QX2000 Manual II Administrator s Guide SIP ACCESS no 122 Blocked IPS A o A A o yA o O ee eee 122 Allowed I
471. t for the extension e Allow Presence Subscriptions e Allow Dialog Subscriptions Edit opens a page Watch Access List Edit Entry where the permissions of the added extensions may be modified 2 SIP Settings This page provides two functions It allows an extension on the QX IP PBX to register to an external SIP server The registration to the external SIP server e g ITSP is usually required before the server will allow the call to be received This page also allows for incoming SIP calls to ring an extension Upon receiving a SIP Invite from an external SIP server the QX IP PBX will look to match the called number with the settings in the User Name DID Number field User Name DID Number is the registration user name on the external SIP server or the DID number from the ITSP The user name needs to be unique on the external SIP server This field length is limited to 32 symbols Ton Add Multiple Extensions Bulk Import Extensions Management Edit Entry O Go Back Password indicates the password for the extension registration SIP Registration Settings 103 on a SIP server Confirm Password is used to confirm the password If the entered password does not correspond to the one entered in the Password field the error message The passwords do not match Please try again will appear Go To Line Settings Go To Codec Settings Fig II 33 Extensions Management Edit Entry SIP Settings page SIP
472. t link leads you to the page where the extensions that are allowed to intercept calls should be defined The Allow other users to Barge In to this extension checkbox and the Edit Call Barge In Intercept Access List link appears only if a Barge In feature is activated from the Feature Keys page e The Allow other users to Barge In to this extension checkbox is used to enable the Barge In Service on the extension e The Edit Call Barge In Intercept Access List link leads you to the Call Barge In Intercept Access List page where the extensions that are allowed to barge in to the current extension or intercept calls should be defined Please Note After activating Barge In feature the extensions that are previously configured to intercept calls from the Call Intercept Access List page will be automatically redirected to the Call Barge In Intercept Access List page along with the Barge In options The Edit Watch Access List link leads you to the page where the extensions that are allowed to watch calls should be defined Call Intercept Access List The Call Intercept Access List page is used to define a list of extensions that are capable to intercept the current extension calls and to define the appropriate permissions QX50 QX200 QX2000 SW Version 6 0 x 25 QX50 0X200 0X2000 Manual II Administrator s Guide QX50 0X200 0X2000 Manual II Administrator s Guide The Call Intercept service allows you to intercept the calls assigned
473. table on this page lists all User Name Registration State Registration Date Time registered accounts and account information It will show the bane A a corresponding authentication parameters username and Qxexos 192 65 74 140 5060 11 00 00 08 11 2014 FXO4GW 192 168 74 101 5060 10 19 37 08 12 2014 password and date time of the last registration QXELTI 192 168 74 126 5060 Not Responding ElTitest Not registered N A The Add functional button opens an Add Entry page where a new account can be configured A Username and a Password is required for a new account on this page Fig II 132 External PSTN Gateways Authorization Parameters page To use the shared remote PSTN lines 1 Enable the Use PSTN lines of the other device checkbox 2 Press Save to apply the selection 3 Enter the Authorization Parameters page 4 Create an account using a unique Username and a Password QX50 QX200 QX2000 SW Version 6 0 x 89 QX50 0X200 0X2000 Manual II Administrator s Guide Telephony Menu The Telephony menu allows you to configure the following settings e VoIP Carrier Wizard e Call Routing Table Overview er CallRouting Overview Call Routing i VolP Carrier Lo cal AAA Table a O VolP Carrier Easily configure the SIP trunking account from the Internet Telephony Service Provider ITSP Telephony Call Routing gt Global 5 eed D ial D lrector wins Call Routing Table Define the destination for dialed
474. tables below Required Bandwidth for Standard Packets Packet Needed bandwidth in kbit s using the Codecs Size in msec E G 726 16 G 726 24 G 726 32 G 726 40 G 729a iLBC 13 33 G 722 G 722 1 10 105 58 66 74 82 50 105 74 20 84 37 45 53 61 29 84 53 30 76 30 38 45 53 22 27 76 45 40 74 27 34 42 50 19 74 42 50 71 25 32 40 48 17 71 40 60 67 22 30 37 45 15 20 67 37 QX50 QX200 QX2000 SW Version 6 0 x 14 e C Pyg l QX50 0X200 0QX2000 Manual Il Administrator s Guide Needed Bandwidth for Encrypted Packets when using a SRTP Packet Needed bandwidth in kbit s using the Codecs amen G 711u G 711a G 726 16 G 726 24 G 726 32 G 726 40 G 729a iLBC 13 33 G 722 G 722 1 10 114 66 74 82 90 58 114 82 20 89 41 49 57 65 33 89 57 30 81 33 41 49 57 26 31 81 49 40 76 28 36 44 52 20 76 44 50 74 26 34 42 50 18 74 42 60 72 24 32 40 48 16 22 72 40 Required Bandwidth for Encrypted Packets when a VPN is used Packet Needed bandwidth in kbit s using the Codecs Size in msec G 711u G 711a G 726 16 G 726 24 G 726 32 G 726 40 G 729a iLBC 13 33 G 722 G 722 1 10 148 98 105 118 124 92 148 118 20 105 59 65 74 81 49 105 74 30 90 43 52 60 66 35 41 90 60 40 85 38 45 53 61 30 85 53 50 80 34 41 48 56 26 80 48 60 74 29 37 45 52 22 26 74 45 Date and Time Settings The Date and Time page provides information about the current system time and date The settings may be updated through the international time and dat
475. te Time field displays the time and the date of last registration on the master s device Class of Service The current implementation of Class of Service CoS on QX IP PBX is used to define the permissions that PBX and Conference extensions will have when using call routing rules to make a call The Class of Service feature provides the ability to set restrictions on the call routing rules for each extension The Class of Service functionality allows to permit or deny the attempt of extensions to use certain types of call routing rules Suppose you want for a certain group of PBX Conference extensions to deny the right to make international calls but allow them to make local and long distance calls and for another group of PBX Conference extensions give a permission to make international calls only Class of Service allows to specify which extensions can use which routing rules to make a call QX50 QX200 QX2000 SW Version 6 0 x 106 QX50 0X200 0X2000 Manual II Administrator s Guide For example if an extension is not assigned to a certain class of service and an attempt is made to place a call from that extension using routing rule with the Class of Service enabled then Number dialed does not exist message will be played to the caller The permissions for a group of PBX extensions can be changed easily by modifying the CoS variable for each PBX extension On QX IP PBX the defined CoS variables are associated with PBX Confere
476. ted timeslot s Select one or more timeslots and click on Edit to open the CAS Signaling Wizard that guides through the key configuration parameters specific to the timeslot PE Interfaces QX50 0X200 0X2000 Manual II Administrator s Guide E1 T1 Trunk ISDN Trunk Trunk 1 192 168 74 127 5060 E1 Signaling Type CAS Incoming Interdigit Service Incoming Digits Timeout 2000 0 20000 ms Signaling Standard ITU Edit ForceUpdate Enable Disable Timeslot Enabled Signaling Type DID Enabled Allowed Call Type Route Incoming Call to Cut Through Automatic Ringing Down Country Timeslot 1 W A N A N A N A N A A N A N A Timeslot 2 A N A N A A N A Timeslot 3 A N A N A N A imeslot 4 A N A N A A N A imeslot 5 ip N A N A N A imeslot 6 N A N A A N A imeslot 7 A N A A A N A imeslot 8 A JA A A N A slot 9 N A N A N A N A imeslot 10 A i A N A imeslot 11 N A imeslot 12 imeslot 13 N A imeslot 14 imeslot 15 imeslot 17 imeslot 18 imeslot 19 W imeslot 20 A imeslot 22 N A Fig II 114 Trunk CAS Signaling Settings page The CAS Signaling Wizard offers a possibility to configure the selected timeslot s and provides a variable group of parameters depending on the E1 T1 trunk configuration CAS Signaling Wizard Page 1 allows to configure signaling type settings and consists of following components Allowed Call Type is used to select the allowed call directions incoming outgoing or both
477. tensions Management Edit Entry Class of Service Settings page The Go to Conferences Management link appears only if the Conference feature is activated from the Feature Keys page and leads to the Conferences Management page where the conference extensions can be assigned to use certain class of service The Go to Call Routing Table link leads to the Call Routing Table page where the call routing rules can be assigned to a certain class of service The Class of Service Add Entry page is used to create a new Class of Service and contains the following components e Name text field indicates the name of the class of service This name will be visible in the Extensions Management Class of Service Settings page in the Conferences Management Class of Service Settings page and in the Call Routing Wizard when assigning the classes for Call Routing Table Call Routing Local AAA Table Global Speed Dial SIP Tunnel Class of Service Class of Services Add Entry O Go Back Telephony Name Class1 Description for calls to PSTN the extensions e Description text field requires optional information about the Class of Service Fig Il 1 Class of Services Add Entry page Call Recording Settings The Call Recording service is optional on the QX IP PBX and is activated from Feature page by inserting a feature key The Call Recording is used to record PBX SIP or PSTN calls on the QX IP PBX and store the recorded cal
478. the voice mailbox and uploaded recorded system greetings It shows the free and total space counted in minutes seconds for every extension This page includes the following information Memory Size shows total memory space counted in minutes seconds available on the QX IP PBX and assigned to all extensions The table s links lead the administrator to the extension settings page where User Space may be altered The System Memory row indicates the space occupied by the universal extension recordings Link refers to the Universal Extension Recordings page where universal extension system messages may be uploaded Call History shows the current number of calls with recorded statistic entries Call History Archive field displays the total and used size of archived call statistics of archiving settings and links to it The Conference Memory Status shows total memory space counted in minutes seconds available on the QX IP PBX The table s links lead the administrator to the Conferences Management page where Total Space for the corresponding conference extension may be altered fil Status QX50 0X200 0X2000 Manual II Administrator s Guide System Status LAN WAN Statistics General Network Lines Memory Hardware SIP Registration IP Lines Registration License Status Memory Status Memory size 22 day 22 hour 32 min 50 sec User Space for Extension y Voice Mailbox System Messages 0 sec 0 sec 0 sec 0 sec 1 sec 0 sec 1 mi
479. the Conference ID is limited to 20 digits The Conference ID cannot start with the digit 0 which is a reserved PDEA character Conferences Management Add Entry O Go Back Conference ID 777 The Conference IDs can be used in Auto Attendant to reach a conference on the system To join a conference using its ID dial the Conference ID when in Auto Attendant To add a conference specify the Conference ID and click on Save This will open the Edit Entry page see below Fig II 76 Conferences Management Add Entry page Email Default Settings Mail Default Settings page is used to define the email templates used in the system generated emails to the conference participants Sra ERE Two email templates can be defined on this page Ss Ge pa i i A Mail Default Settings e Conference Notification Default Mail delivered when the Extensions moderator chooses the Send Notification Mail menu option Conference Notification Default Mail e Conference Activation Default Mail delivered by the conference Scheduling system if the Send Mail before Conference Activation option is enabled er a Participants lt participants gt Each template should be defined in the corresponding text field Conference Activation Defaut Maii Additionally functional tokens can be used to automatically insert prole the Conference ID Subject Description Participants Password o Scheduling information as well as a possibility to display th
480. the deletion with Yes The selected recordings will be deleted To abort the deletion and keep the recordings in the inbox select No Attendant Extension Settings For Attendant extensions the Extensions Management Edit Entry page consists of General Settings Attendant Scenario SIP Settings and SIP Advanced Settings pages The SIP Settings and SIP Advanced Settings pages are the same as for the regular extensions described above The General Settings and Attendant Scenario pages are described below 1 General Settings for attendant extension This group requires AA extension information and has the following components Display Name is an optional parameter used to define the Auto Attendant s description Usually the display name appears on the called party s phone display when a call is made or a voice mail is 2 Show on Public Directory sent Go To Codec Settings Percentage of Total Memory 5 With the Enable FAX Forwarding checkbox enabled the system moves the incoming FAX to the selected extension if a FAX tone is detected on the Auto Attendant Fig I 61 Extensions Management Edit Entry General Settings for Auto Attendant page QX50 QX200 QX2000 SW Version 6 0 x 43 QX50 0X200 0X2000 Manual II Administrator s Guide The Extension to forward drop down list is used to choose the extension where the incoming FAX addressed to the QX IP PBX s Auto Attendant will be forwarded The list contai
481. the transmit direction Disable Van Jacobson style TCP IP header comp ression in a Van Jacobson TCP IP header _ in M ee N receive direction isable the connection ID compression option in Van Jacobson style IP header compression both the transmit and the receive direction with this option able the PXCP and PX protocols no negotiation of TCP IP header compression will take place and the header will always be sent uncompressed Disable the connection ID compression option in Van Jacobson style TCP IP header compression with this option PPPD will not compress the connection ID byte from Van Jacobson and will not ask the peer to do so Fig Il 217 Advanced PPP Settings page Disable the IPXCP and IPX protocols this option should only be selected if the peer is not working properly and cannot handle requests from PPPD for IPXCP negotiation SNMP Settings The Simple Network Management Protocol SNMP is an application layer protocol that facilitates the exchange of management information between network devices and is used by network administrators to manage network performance find and solve network problems and plan for network growth On QX IP PBX SNMP agent is running to allow administrators to remotely manage QX IP PBX s network and the device s configuration Remote administration is being performed by means of special SNMP monitoring programs SNMP Manager which can automatically feedback by the certainly configured
482. thm and is used in the Digital Signature standard FIPS number 186 from NIST SHA is an improved variant of MD4 producing a 160 bit hash SHA and MD5 are the message digest algorithms available in IPSEC MD5 Message Digest is a hash algorithm that makes a checksum over the messages The checksum is sent with the data and enables the receiver to notice whether the data has been altered QX50 QX200 QX2000 SW Version 6 0 x 138 QX50 0X200 0X2000 Manual II Administrator s Guide The Diffie Hellman parameter is used to determine the length of the base prime numbers used during the key exchange process The cryptographic strength of any key derived depends in part on the strength of the Diffie Hellman group which is based upon the prime numbers The higher is the group bit rate the better is encryption If mismatched groups are specified on each peer negotiation fails The third page of the IPSec Connection wizard Automatic Keying is used to setup a type of password Shared Secret or the RSA public key to secure your IPSec Connection The functionality of Perfect Forward Secrecy PFS can be added to both Following ways of automatic keying are available e Shared Secret is a type of password consisting of any characters that both of the IPSec Connection partners must know The authentication will be done with this shared secret All encryption functions below will remain concealed Please Note It is also not recommended to start multi
483. time The Records per page are used to select the number of displayed statistic records per page The Previous and Next can be utilized to switch between these pages The Download Call Detail Records links are available below for all Call History tables for administrator s access only and allows you to download the displayed Call History in a text file Call History Settings The Settings page offers the following input options The Enable Call Reporting checkbox enables Call History reporting The selected number of statistics entries will be displayed in the Call History tables QX50 QX200 QX2000 SW Version 6 0 x 151 pya The Maximal Number of Displayed Call Records drop down lists are used to select the number of Successful Missed and Unsuccessful Outgoing statistics entries to be displayed in the corresponding Call History tables If the record numbers exceed the numbers specified in these drop down lists the oldest record will be removed The Download All Call Detail Records link is used to download the entire displayed statistics in a file that can be viewed with a simple text editor This type of Call History file is easy to read and can be displayed in a spreadsheet The Download All Call Detail Records in CSV format link is used to download the entire displayed statistics in CSV Comma Separated Values formatted file The Clear all Records button is used to clear all statistics records ful Status QX5
484. tings DCC will be functional with the extension only during trial period AAA Fig IT 39 Extensions Management Edit Entry License Settings page Please Note These checkboxes can be simultaneously selected on as many extensions as iQall and or DCC Pro Basic Level licenses are available on the QX IP PBX Enable iQall Toggling license checkbox allows you to allocate the iQall Toggling licenses to the corresponding extensions The Go to User Settings link is used to make a quick jump to the extension specific Extension s Main Menu page see Manual III Extension User s Guide The Go to Line Settings link is used to make a quick jump to the IP Lines page of the corresponding extension The Go to Codec Settings link is used to make a quick jump to the Codec Settings page of the corresponding extension Pickup Group Extension Settings Pickup Group Access List The Pickup Group service is used to monitor calls addressed to a certain list of extensions and to pick up calls ringing on the listed extensions This service may be used when a group of extensions are located in the same area so the persons nearby can hear the ringing on one of the extensions This feature allows you to pick up the call ringing on a certain extension by dialing the number of the pickup extension The Pickup Group list is used to define the extensions that can be monitored by calling a certain pickup extension The Access List is used to define PBX SIP o
485. tings for the callers who passed an authorization by phone on the QX IP PBX This checkbox enables Username and Password text fields to insert the custom settings that will stand instead of the source caller s settings when being delivered to the RADIUS server The Authentication on Destination RADIUS Server parameters group is used to insert a USername and a Password followed by the password confirmation to pass authentication on the RADIUS Server of the destination QX IP PBX If these fields are left empty the original authentication settings that users enter for authentication will be used 3 Accounting Settings The Username field is dedicated for accounting services only It is used to insert an identification username for accounting purposes When no username is specified in this field the source username will be used for accounting The Send Accounting messages manipulation radio buttons group is used to select sending both Start and Stop accounting messages or only Stop accounting message QX50 QX200 QX2000 SW Version 6 0 x 116 e C Pyg l QX50 QXK200 QX2000 Manual II Administrator s Guide Dial Timeout The Dial Timeout Settings page is used to adjust the dialing timeout setting The Routing Dial Timeout setting specifies a period of time after the last dialed digit that the system identifies as a completion of dialing If the user does not press any key within the specified timeout the system assumes that the dialing
486. tion and has the following components Add Multiple Extensions Bulk Import Extensions Management Edit Entry Extensions O Go Back Display Name is an optional parameter used to recognize the caller Usually the display name appears on the called party s General Settings phone display whenever a call is performed or a voice mail is sent General Settings 50 Password requires a password for the new extension The extension password may only contain digits If non numeric Show on Public Director symbols are entered an Incorrect Password no symbol characters eee E a lec Y Customize push back number allowed error will prevent making the extension Setings Call Type Auto Y If you are unable to define a strong password press Generate caro P Password to use one of system defined strong passwords The Password field is checked against its strength and you may see how strong is your inserted password right below that field Fig IT 44 Extensions Management Edit Entry General Settings for call park extension Confirm Password requires a password confirmation If the input is not corresponding to the one in the Extension Password field the error will appear Incorrect Password confirm With the Show on Public Directory checkbox enabled the details of the corresponding extension will be displayed in the User Settings table on the Main Page of the Extension s Web Management ac
487. tistic records per page The Previous and Next can be utilized to switch between these pages The Download Call Detail Records links are available below for all Conference Call History tables and allows you to download the displayed Call History in a text file Conference History Settings The Settings page is only displayed when an administrator is logged in The Conference History Settings page offers the following input options QX50 QX200 QX2000 SW Version 6 0 x 156 The Enable Call Reporting checkbox enables conference Call History reporting The selected number of statistics entries will be displayed in the Conference Call History tables The Maximal Number of Displayed Conference Call Records drop down lists are used to select the number of Conference Call Successful and Unsuccessful statistics entries to be displayed in the corresponding Conference Call History tables If the record numbers exceed the numbers specified in these drop down lists the oldest record will be removed Gil Status QX50 0X200 0X2000 Manual II Administrator s Guide EV ts ory erenc Unsuccessful Outgoing Calls Conference History Settings Conferences Successful Calls Y Enable Call Reporting Maximal Number Of Conference Call Records 50 Maximal Number Of Successful Call Records 50 Maximal Number Of Unsuccessful Call Records 50 Download All Call Detail Records Download All Call Detail Records in CSV format The Down
488. to Adjust System Security security report link below vee e FirewallNAT Epygi treats system security with the utmost priority and has taken an active approach to provide users with information and tools to aid in maintaining system security It is highly recommended that users of an do ne IP based system need to be familiar with industry best practices to maintain system security Limitation of Liability and Remedies In no event shall Epygi Technologies be liable for any consequential incidental direct indirect special punitive or other damages including without limitation loss of data loss of phone calls loss of business profits business interruption loss of business information or other pecuniary loss arising out of the use or inability to use the Quadro Fig II 264 Security Diagnostics page The Show Security Report link allows to display the last security audit report This page also contains the following useful links to adjust the system security e User Rights Management 0 IP Line Settings 0 Firewall NAT Call Capture The Call Capture page is used to capture the voice streams on the active calls and the available interfaces on the QX IP PBX FXS and FXO This page consists of two sub pages QX50 QX200 QX2000 SW Version 6 0 x 161 QX50 0X200 0X2000 Manual II Administrator s Guide The Active Calls sub page lists all FXO FXS active calls on the QX IP PBX for the certain moment e Capture Timeout t
489. to an individual extension The extensions that are allowed to intercept calls are defined in the Call Intercept Access List With the special feature codes for details see Feature Codes in the Manual III Extension User s Guide you may pick up a ringing call ofthe extension This page contains the following functional buttons Add functional button opens an Add Entry page where extensions q En may be added to the Call Intercept Access List PRA Call Intercept Access List of Extension 77103 This page requires the extension number in the Address text field SS that will be allowed to intercept calls The wildcard is supported in 7 PBX the Address field to add a group of extensions with one entry PBX 108 The Allow Intercept checkbox on this page allows to select the Intercept option for the added extension Attention Intercepted calls are not displayed in Active Calls table Fig II 27 Call Intercept Access List on the Administrator s Main Page nor are registered in the Call History s Add Extension Add Multiple Extensions Bulk Import Call Intercept Access List Add Entry Extensions Call Type PBX Address 108 wildcard supported Y Allow Intercept Save Fig II 28 Call Intercept Access List Add Entry Call Barge In Intercept Access List The Call Barge In Intercept Access List page is used to define a list of extensions that are capable to Barge In Intercept the current extension calls
490. to download and use Remove to delete the corresponding custom voice message Choose File No file chosen Save Browse opens a file chooser window to browse for a custom voice message for an archive file with the tar gz extension containing the custom attendant scenario and the voice prompt recordings Fig ll 68 Upload Custom Voice Messages page The Attendant Ringing Announcement group allows uploading an optional voice message that is played to callers instead of ring back tones when making calls through an auto attendant The Ringing Announcement can be enabled for both custom and default attendants QX50 QX200 QX2000 SW Version 6 0 x 48 QX50 0X200 0X2000 Manual II Administrator s Guide Please Note The Attendant Ringing Announcement is played to SIP to extension and PSTN to extension calls only The announcement can also be played to SIP attendant SIP and PSTN attendant SIP calls if they are made by a call routing rule for which the RTP proxy is enabled The group offers the following components The Enable Ringing Announcement checkbox enables disables the Auto Attendant optional announcement message When this checkbox is selected but no custom announcement message is uploaded the default message will be played to callers e File selection is used to upload the ringing announcement file The following option is available under this selection Upload new ringing announcement indicates the file name used to up
491. tone characterized by beginning with the key O Administrator Login Allows to modify Auto Attendant greeting and menu messages as well as to manage universal extension messages Enabling disabling the Call Routing rules Allows managing the routing entries in the Call Routing table i e to enable disable certain dialing rules by dialing key combinations pre configured on each routing entry By dialing 9000 you will be required to dial enabler disabler key to enable or disable the routing rule s correspondingly Since multiple routing rules may have the same enabler disabler key combinations the same key may be used as enabler for one routing rule and as disabler for another one dialing the certain key will affect all pre configured routing rules If the routing record has an authorization enabled on the enabler disabler key administrator s password will be required to be inserted after the key Once the administrator s password is dialed system plays a confirmation about the accepted configuration and the state of the certain routing rule s is getting modified If administrator s password has been inserted incorrectly for 3 times no status changes will be applied to any of the routing record s even to those which have no authorization enabled Administrator Login menu has the following sub menus and the management Keys 6 009 administrator s Login 900 900 O Auto Attendant Auto Attendant Universal Extension Mes
492. tors may connect QX IP PBX with Telnet protocol port number 645 and access the logs selected on this page This is done for remote QX IP PBX s diagnostics and is mainly used by Epygi s Technical Support Office To make the QX IP PBX s logs open for remote access appropriate Firewall level or Filtering Rules must be created Checkboxes below on this page are used to select those log types that should be accessible remotely Select only those logs that you wish to have monitored remotely Logs Archive The System Logs Archive page available only for QX2000 shows the archived logs table with time period by Date Clicking on the corresponding date will open the archived system logs table in hourly basis Hour shows the initiation time of the system logs This could be used to collect the logs at the exact moment when a problem has happened The Unpacked size on disk shows the system logs size on disk for the corresponding Date and Hour The following functional buttons are available on this page Download link is used to download the archived system logs file to the PC and opens the file chooser window where the saving location can be specified Delete removes the selected entry from the archived system logs table QX50 QX200 QX2000 SW Version 6 0 x amp Maintenance Y Enable User Logging Y Enable Developer Logging Log Lines to Show 50 Y Mark all logs Comment Consultative Transfer Problem 4 Maintenance Downloa
493. ts will appear at the upper right corner of all management pages The system events and the warning message are visible only for the administrator The warning link which leads directly to the System Events page QX50 0X200 0X2000 Manual II Administrator s Guide will disappear from the management pages if the administrator has marked all new events as read The System Events table is the list of new and read system events System events have corresponding coloring depending on the nature of the event success priority 1 color green low importance failure priority 2 color yellow critical failure priority 3 color red The table shows the Status of the event new or read as well as the name of the application the event refers to event description and the date when the event was received For example if the event was caused by the IDS service the Check IDS link available only for QX50 0X200 appears in the reference row that will lead to the IDS Log page or if the event has occurred due to incorrect mail sending or SIP registration the corresponding links will be seen in the Reference column of the table The administrator can view the detailed log for each event that has occurred The System Events page offers the following components Current System Time displays the local date and time on QX IP PBX Mark all as read marks newly occurred events as read Reset LED switches off the flashing LED if app
494. ttern instead of the EN discarded digits The Prefix field can consist of numeric lado values only A corresponding warning appears if any other symbols are inserted Access code Emergency Code 911 e By pattern text field specifies calls to which the rule Poe coasts L130 7 should be applied If an outbound call has a destination number that matches the specified pattern it will be completed according to the current rule A routing pattern may contain wildcards The complete list of characters and wildcards allowed in this text field is Previous given on the Allowed Characters and Wildcards page The Route Incoming Calls to drop down list allows you to NA T a ae select an extension or Auto Attendant on the QX IP PBX where incoming calls from the configured VoIP Carrier should be routed to For the selected extension there will be an unconditional forwarding set up which will care for incoming calls forwarding from the VoIP carrier to the corresponding Fig II 136 VoIP Carrier Wizard page 3 extension The Emergency Code text field requires the emergency code supported by the specified ITSP By default this field is filled with the information defined in the QX IP PBX s System Configuration Wizard but this field also allows to define an ITSP specific emergency codes In case your system has both local PSTN emergency codes and ITSP codes configured when dialing the certain emergency code QX IP P
495. ttings B Channels page E1 T1 Trunk ISDN Trunk PSTN Gateways Edit functional button opens B channels Edit Entry page which Trunk 1 192 168 74 127 5060 E1 Signaling Type CCS B Channels Edit Ent contains 3 checkboxes nee Ua SE E See A Timeslot 2 FE Interfaces e Enable Timeslot used to enable disable the selected a timeslot s Enable Echo Cancellation e Force Update Timeslot used to apply new settings immediately by restarting the timeslot s e Enable Echo Cancellation used to enable disable the echo cancellation feature on the selected timeslot s Fig II 123 Trunk CCS Signaling Settings B Channels Edit Entry page Please Note A timeslot can be used either for voice or data transfer Timeslot selected for the D Channel receive transmit is missing in the list of B channels The Bearer Establishment Procedure drop down list allows to select the session initiation method on the B channels One of the following possibilities of the transmission path completion prior to receipt of a call acceptance indication can be selected e on channel negotiation at the destination interface e on progress indication with in band information e oncall acceptance The Calling Party Type of Number drop down list allows to select the type identifying the origin of call The Called Party Type of Number drop down list allows to select the type identifying the subaddress of the called party of the call The Called P
496. ttings and Preferences page has two drop down lists to select the Your Locale location and a Regional Settings and Preferences corresponding Timezone QX IP PBX will support Daylight n a 3 Your locale location Armenia v Savings DST correction if it is available for the selected time io aaa Van zone Choose System Language This page also has a manipulation radio button group to italian tal choose English US e System Language selection is available only when the custom Language Pack has been uploaded and it is used to enable custom language for system voice messages or returning back to the default language English Previous Fig II 5 System Configuration Wizard Regional Settings page The Emergency Codes and PSTN Access Code Settings are used to configure the emergency dial plan The Emergency Codes text field requires the PSTN numbers of the emergency or lifeline services Multiple emergency codes separated by commas can be inserted in this field For each emergency code a routing pattern will be generated in the Call Routing Table which will allow faster and easier calls 7 to emergency destinations Please ente all your Emergency Codes separated by commas and PSTN Access Code into following fields System LAN Internet WAN Date and Time Email SMTP Short Text Messaging SMS System Configuration Wizard Emergency Codes and PSTN Access Code Settings The PSTN Access Code drop d
497. ttings defined in the wizard i e corresponding routing rules will be added to the Call Routing table of the FXS Gateway If you need to reboot the FXS gateway use the Reboot functional button in the FXS Gateway Management page QX50 QX200 QX2000 SW Version 6 0 x Interfaces Fe Interfaces Pe Interfaces QX50 0X200 0X2000 Manual II Administrator s Guide El T1 Trunk ISDN Trunk PSTN Gateways IP Lines IP Line Settings IP Phone Templates IP Phones Logo FXS Gateways FXS Gateway Management Add Edit Delete Reboot Restart MAC Address Description 00 f0 00 f0 81 85 QXFXS24 00 f0 00 f0 81 84 QXFXS24 Fig II 104 FXS Gateway Management page ISDN Trunk PSTN Gateways IP Lines IP Line Settings IP Phone Templates IP Phones Logo FXS Gateways FXS Gateway Configuration Wizard O Go Back FXS Gateway Model FXS Gateway Model Epygi QXFXS24 MAC Address 100 t0 00 0 81 34 Description QXFXS24 Fig IT 105 FXS Gateway Configuration Wizard FXS Gateway Model page E1 T1 Trunk ISDN Trunk PSTN Gateways IP Lines IP Line Settings IP Phone Templates IP Phones Logo FXS Gateways FXS Gateway Configuration Wizard O Go Back FXS Gateway Summary FXS Gateway Model Epygi QXFXS24 MAC Address 00 f0 00 0 31 84 Description QXFXS24 Lines Mappings Gateway Lines IP Lines Line 1 IP Line 48 Line 2 IP Line 49 Line 3 IP Line 57 Line 4 IP Line 94 Line 5 IP Line 114 Line 6 IP Line 115 P
498. tween 1024 and 65536 SIP addresses needs to be specified in one of the following formats display name lt username ipaddress port gt display name lt usernameOipaddress gt username ipaddress port A flexible structure of wildcards is allowed In comparison with a wildcard username ipaddress the character stands for only one unknown digit and the character wcarnane stands for any number of any digits For your convenience the following combinations can be used Please Note Wildcards are available for caller addresses only No wildcard characters are allowed for called party addresses Exceptions are addresses in the Supplementary Addresses table that are used by Outgoing Call a username a specified user from any SIP server Blocking and Hiding Caller Information Settings services To use and 2 alone as non wildcard characters use and correspondingly e ipaddress any user from the specified SIP server e any user from any SIP server SIP Tunnel Settings The SIP Tunneling service is used to build a tunnel between QX IP PBXs and to use that tunnel for routing the SIP calls through the remote QX IP PBXs When this service is enabled slave QX IP PBXs should be registered on the master QX IP PBX with the corresponding username password With the appropriate configuration done on the master QX IP PBX the master device can use the slave QX IP PBXs for routing the S
499. two simultaneous calls can take place at a time The QX50 0X200 0X2000 has no own ISDN trunks only shared ISDN trunks are displayed in this page if available The shared trunks lines can be edited from this page Any changes applied in this page will be automatically reflected on the QX ISDN gateway s that share its ISDN trunks The ISDN Trunk Settings page is used to configure the ISDN trunk and their signaling This page offers the following input options The Trunk Settings table lists the available ISDN trunks on the QX ISDN Gateway s and their settings trunk name and interface types The Start and Stop functional links are used to start shutdown the Owenem Punes PS PO EVM Tunk BONT ASIN Gateway selected ISDN trunk s When an ISDN trunk is in a shutdown state ISDN Trunk Settings ISDN calls cannot be placed or received The following virtual ISDN trunks are available Pe Interfaces The Restart functional link is used to bring channel s to the initial A e idle state on both sides When applying one of these options any Tunk 1 2192468 741355060 active traffic on the channel s will be terminated ee Trunks PTMP Point To Multi Poin PTMP Point To Multi Poin runk 3 192 168 74 135 5060 PTMP Point To Multi Poin runk 4 192 168 74 135 5060 PTMP Point To Multi Poin The Copy to Trunk s functional link displays a page used to ln a J runk 2 192 168 74 109 5060 PTMP Point To Multi Poin choose a trunk to which
500. type of firewall but together with deleting dangerous packets or packets containing intrusion attacks IDS generates a log file with information about these dropped packets and the senders responsible for those packets The log can be viewed on the IDS Log page and notifications about them can be sent to the user in various ways such as e mail flashing LED and display notification Firewall and NAT The Firewall Configuration page offers the following components The Enable IDS checkbox selection enables the Intrusion Detection System The Enable NAT checkbox selection enables Network Address Translation The Enable Firewall checkbox selection enables the firewall security service The firewall security level has to be selected Firewall Filtering Rules otherwise the firewall cannot be enabled a EN The Firewall Security radio buttons are the following e Firgwall Configuration e Low Security Everything that is not explicitly forbidden will Y Enable DS Y Firewall Y Enable NAT be allowed This security level doesn t block anything by default It is recommended if the device is already located behind another firewall or if every filter has been configured Lom Security setting is alowed thats not explicity forbidden This policy doesn t block anything per default You have to configure the filters manually This option is recommended if this device is already located behind CO rrectly another firewall or if you are sure that you have confi
501. u will not be automatically redirected to the GUI start page After a successful reboot you need to login again Are you sure you want to reboot the device amp Maintenance Yes Reboot Device Fig II 291 Reboot device page QX50 QX200 QX2000 SW Version 6 0 x 171 e C Pyg l QX50 QXK200 QX2000 Manual Il Administrator s Guide Registration Form The Register Your Device in Technical Support Center page appears when administrating an unregistered QX IP PBX and it has been created for customer support purposes The page requires customer registration at the Epygi Technical Support Dashboard Center It provides several links offering the following a a eee ee ene registration options ias N Register Your Device In Technical Support Center Don t remind me again Register now leads to the Epygi Technical Support System Registration page and requires customer s information to submit the QX IP PBX registration form Remind me later hides the registration notification in the QX IP PBX through System Configuration Wizard or Internet Configuration Wizard until the next administrating activities Fig II 292 Device Registration page Don t remind me again hides the registration notification forever QX50 QX200 QX2000 SW Version 6 0 x 172 QX50 0X200 0X2000 Manual II Administrator s Guide Appendix PBX Services for QX IP PBX s Administrator The following PBX Services are accessible at the dial
502. udes the only ACD Group ID text field which requires the ACD Group number extension The ACD Group ID should not match any existing extension in the Extensions Management table Any newly created ACD Group will automatically appear in the Extensions Management table Skills Agents ACD Management Add Group Go Back Extensions ACD Group ID 888 Save Edit opens ACD Group Extension Settings in the Extensions Management Fig IT 93 ACD Group Management Add Entry page Pressing on the links in the Group ID and Agents List columns of the Groups table will lead you to the ACD Group Extension Settings where group settings and the list of group s agents may be adjusted correspondingly Authorized Phones Database The Authorized Phones Database page is used to create a list of trusted external phones If they are part of the QX IP PBX Authorized Phones database external SIP or PSTN then users are free to access the QX IP PBX Auto Attendant services without requiring authentication When adding a trusted phone to the list an existing extension has to be chosen The parameters extension number and password as well as SIP and Speed Calling Settings will be used automatically for the trusted caller access of the QX IP PBX Auto Attendant A direct connection to the Call Relay menu can be optionally provided The Authorized Phones Database page displays the Authorized Phones Database table where the trusted phones are listed Only
503. uld be selected if the participant is a Conference itself and enables the correct behavior of conference termination e Allow Duplicated Participation checkbox allows multiple participants with the selected Caller ID calling address to join the corresponding conference This is applicable when different participants are using the same shared number to place a call The Edit functional button provides a possibility of editing multiple participants at the same time A Select to modify fields z ee checkbox alongside the fields to be modified needs to be selected WW conference Genial Recording animeen Schedule Sena Notitication Mal to submit changes otherwise the fields will not be updated Participants Edit Entry Conference ID 888 Select to modify fields Participant Name SIP Address Tel Number 11369 sip epygi loc 5986744 Mail Address Participant Type Confirmation Type Y Allow Video Dial Out Activate On Dial In Y Y Participant Indication Y Nested Conference Y Allow Duplicated Participation Save Fig II 302 Conference Settings Participants Multi Edit Entry page New Participants Configuration This page is used to configure settings of participants independently dialed in to the conference Once the new participant connects the conference he will automatically appear in the Conference Progress table and remain there unless disconnected from the conference Max New Participant Count text field requir
504. umber under the Available Lines column the FXS On board Line Settings page specific for the current line is opened and offers the following input options The Caller ID drop down list contains various standards of Caller ID transmissions It is used to send the calling party s information to the phone attached to the selected line E1 T1 Trunk ISDN Trunk PSTN Gateways No Caller ID Line Settings Line 1 FF Interfaces Caller ID e FSK send prior to the first ring i l tandard 2 FSK send between first and second ring e FSK send between the first and second ring P E Y Enable Busy Tone Indication e FSK send both prior to a ring and between the first and second Busy Tone Duration 5_ Y Enable Power Disconnect Indication ring Disconne ct Duration 300 Y msec e DTMF send prior to the first ring e DTMF send between the first and the second ring ROO Y Enable Hot Desking Capability e Combined send both DTMF prior to the first ring and FSK Mii ii between the first and the second rings Never m After B hour s 45 min The QX IP PBX sends the current time date to the called phone together with the caller s information Fig II 108 FXS Line Settings page A group of Remote Party Disconnect Indication parameters are used to configure the private PBX attached to the QX IP PBX FXS port e The Enable Busy Tone Indication checkbox enables a busy tone transmission to the FXS port when the remote party
505. uploaded and is used to remove it and restore the default call queue welcome message The Download call queue welcome message functional link appears only when the custom call queue welcome message is already uploaded and is used to download it to PC and opens the file chooser window where the saving location can be specified Customize Queue Scenario settings are used to define a custom scenario for audio files played in the ACD queue Here you may upload custom audio files and to define the sequence in which they will be played for the person in the queue By selecting this option the default ACD queue messages will be replaced with the scenario defined below Custom Queue Messages table lists all audio files in the custom queue scenario and allows you to add new field Each audio file is characterized by the number of repeats and the timeout when it should start The audio files may be ordered in the list with Move Up and Move Down functional buttons The custom queue will start with the first audio file in this list and will be played in the loop in the order audio files are listed The Add functional button opens an Add Entry page where a new audio file can be defined This page consists of the manipulation radio buttons selection to allow upload a new audio file or to select an already uploaded one Ertemions Skills Agents ACD Management Add Entry e Existing File this selection is used to choose one of the T gt basting File already
506. uploaded custom queue messages to include in the DES scenario The same file may appear in the different tion cw Fa instances of the queue music i e Upload New File used to upload a new audio file The uploaded files should to be in PCMU CCITT u law 8 kHz 8 bit Mono wave format otherwise the system will prevent uploading it with the Invalid audio file or format is not supported warning message Fig IT 54 Extensions Management Edit Entry ACD Group Settings Add Queue Message page Please Note The file name can contain only alphanumerical characters and _ symbols Attention You should have enough memory allocated to the corresponding extension from General Settings in order to be able to upload audio files otherwise error message prevents uploading new files Play Count indicates the number of times the corresponding audio file will be played continuously in the queue Timeout indicates the timeout in seconds between the end of the previous queue audio file in the scenario if any and the beginning of the current audio file For the first audio file in the list this timeout indicates the interval between the beginning of the queue and the beginning of the current audio file s playback Play Background Music checkbox is used to fill in the timeout intervals between the audio files in the scenario with the background music This option requires you to choose the Audio Line in or RTP Channel of broadca
507. uration It is recommended to enable the Oveniew IPies PS PO ELM Trunk ISDN Trunk PSTN Gateway MSN when there are multiple ISDN devices connected to the same ISDN Wizard ISDN bus If the MSN is enabled on this page the next page will require the MSN table configuration o MSN No MSN Previous Fig II 128 ISDN Wizard ISDN PRMP Settings page For MSN service enabled the Routing Settings page is used to assign MSN numbers to the certain destinations on the QX The MSN number can be assigned to the QX IP PBX s extensions to the Auto Attendant or to the routing agent The destination selected from this page will ring upon incoming call to the corresponding MSN number comes in QX50 QX200 QX2000 SW Version 6 0 x 86 pya The fields in the MSN Number column require the MSN numbers allocated to the QX Please Note At least one MSN number should be defined in this page The system displays an error message if the same MSN number is used twice in this page The Route Incoming Call to drop down lists is used to select the destination where the incoming call addressed to the certain MSN number will be routed Choosing the Routing with inbound destination number selection will automatically use the initially dialed number to connect the destination without any additional dialing If MSN is disabled on the ISDN Wizard MSN Settings page the ISDN Wizard Routing Settings page contains only one Route Incoming Call
508. uration Wizard Status on this page is QX50 QX200 QX2000 SW Version 6 0 x 30 QX50 0X200 0X2000 Manual II Administrator s Guide changed to Passed and a Reactivate button appears Using Reactivate button you might re enable the Voice Mail Configuration Wizard so the user will be again prompted about his her personal settings next time entering his her Voice Mailbox Instructions on how to insert the information prompted in the Voice Mailbox Settings are available in the Features Codes see Manual III Extension s Users Guide The Shared Mailbox section is used to setup a mailbox sharing The Edit Voice Mailbox Access List link goes to the page where a list of PBX extensions can be defined for which the mailbox of the current extension will be shared and accessible without password authentication For more details on how to access Shared Mailboxes see Feature Codes e Use External Voice Mail enables the Voice Mail service for the corresponding extension and is used to define a remote Voice Mail Server as a location for the Voice Mails In this case recorded voice mails will be collected on the remote server Radio button selection enables a sub group of manipulation radio buttons n Add Multiple Extensions Bulk Import eo Ifthe remote Voice Mail Server is combined with the SIP Proxy server it is recommended to select Proxy Controlled Mailbox Type With this selection SIP proxy will keep the recorded voice mail on itself When
509. ured stream as an archived tar file which contains two streams receive and transmit of the corresponding stream The files can be then played with an audio application The Remove Capture link is used to remove the captured audio stream Ping Ping sends four ICMP Internet Control Message Protocol requests with a default size of 64 bytes to the destination IP address or host name specified in the text field Ping Target The response times are logged and the round trip time the time required from being sent until being received again is measured The minimum and maximum round trip time and its average as well as the percentage of lost and of received frames results are displayed in the lower area of the page Ping Target requires the destination IP address or host name for the ping request If Use ICMP checkbox is selected an ICMP request will be send to the ping destination MS Windows standard Otherwise if checkbox is not selected a UDP request will be send Linux standard The Start Ping button starts pinging the specified ping target QX50 QX200 QX2000 SW Version 6 0 x Diagnostics System Logs Diagnostics Security Diagnostics Melle mE Ping Traceroute Call Capture Active Calls Interfaces Call Start Time Call Duration Calling Phone Called Phone 06 Aug 2014 15 04 34 8 sec 102 107 Pog Maintenance Capture Timeout sec Start Call Capture Active Calls Interfaces XS y amp Maintenance
510. urrently inactive in the conference but configured to be dialed out also those added manually from the handset by moderator Service is available for Moderators only Dial out to all participants to the conference Initiates the dial out to all participants currently inactive in the conference Service is available for Moderators only Next Phone with Video Capability Shows the next phone with video capability Also switches from automatic mode to manual one Previous Phone with Video Capability Shows the previous phone with video capability Also switches from automatic mode to manual one Automatic Video Switching Mode With this key combination the loudest speaking participant is displayed on all video capable phones If that participant has no video capability a black screen will be displayed Start or Resume Conference Recording Service is available for Moderators only Pause Conference Recording Service is available for Moderators only Stop Conference Recording Service is available for Moderators only Request to Speak With this key combination a listener requests to speak and a notification hand up icon is displayed in the Conference Progress table The moderator can then switch the particular listener either to speaker or lecture mode With a speaker permission granted listener can speak to the conference along with other participants With a lecturer permission granted listener can speak to the conference havi
511. ut the call quality voice codec used to ls pd A Called Phone receive and transmit packets and the close conference call z B in reason The close conference call reason appears to provide 7 ei more information about the call termination reason which o E lt can be a network problem termination by one of the m reee m conference call parties voice mail service activation etc l SS Clicking on the details information will open the RTP 19 Nov 2014 04 33 19 Nov 20 19 Nov 2014 04 33 00 2 sec 234 2 168 4 205 401 Statistics page where all RTP parameters of established o TE ZE 19 Nov 2014 04 32 58 2 sec 12344321 192 168 4 205 409 confere nce call are provide d 08 19 Nov 2014 04 32 57 2 sec 12344321 192 168 4 205 408 19 Nov 2014 04 32 56 2 sec 12344321 192 168 4 205 407 e Authenticated By information details the conference ic 19 Nov 24 04525 1212184205609 participants that passed an authentication on the QX IP PBX as configured in the Local AAA Table Download Call Detail Records Successful Calls Fig II 254 Conference History Successful Calls page The Call Detail column is present only in the Unsuccessful Outgoing Calls table and indicates the reason why the call was unsuccessful The Filter button performs searching within the statistics tables The search may be done with several criteria at the same time The following search criteria are available e The text fields ConfID Fr
512. vice If the master device becomes unavailable which can be caused by power loss reboot or network malconfiguration the second QX IP PBX becomes automatically available and starts to run as a master device Depending on the configuration the second QX IP PBX can remain master or go to the backup mode once the first device becomes available again Attention During failover procedure all active calls will be disconnected and the system will be out of service during 2 5 minutes depending on the number of IP phones connected to the system which is needed for running the applications and rebooting the phones If there are IP phones in the network that are not auto configured by QX IP PBX IP phones not supported by Epygi or IP phones with the changed login name and password you will need to reboot them manually After failover the license keys firmware and language pack are not being transferred from the master to backup QX IP PBX therefore so make sure both QX IP PBXs are configured identically in the redundant network before enabling redundancy mechanism When you login to the device which runs in a backup mode only Redundancy Settings are available All other GUI configuration settings are non editable and automatically synchronized with the master device s configuration To ensure the interaction between the master and slave devices corresponding configuration should be done in the Redundancy Settings on both devices Enable Redundancy check
513. w Password empty Phone Access Password Old Password empty New Password empty Confirm New Password empty Disabled Enabled Server Configuration Assign manually Server Name ftp epygi com Server Port 21 Update Method ftp Username anonymous Password empty Check and notify Every day at 0 00 System Default Value Maximal mail message duration 5 min Ask password before granting local access to mail box disabled Ask password before granting remote access to mail box enabled Send welcome message disabled Play Voice Mail help enabled Automatically play messages enabled Send mails count information message disabled Send date time information message enabled Send beep at the end of message enabled Silent VM recording disabled Send new voice messages via e mail disabled Voice Mail Send notification with attachment Remove Voice Mail On Send disabled Fax Send notification with attachment Remove Fax On Send disabled Send new voice message notifications via SMS disabled Send new voice message notifications via phone call disabled Voice Mail Indication Lamp indication enabled for IP lines only Tone indication enabled for FXS lines only Ringing indication disabled Zero Out enabled Redirect Call Type PBX Redirect Address 00 FAX Redirection disabled Automatic Fax Receiving Mode disabled Out of Office disabled Forward rewind duration 3 seconds Greeting message default U
514. wed e Max maximal port has to be lower than 65536 and higher than the minimal port range Only odd numbers are allowed Please Note RTP RTCP Mapped Port ranges should be greater than or equal to the RTP RTCP port ranges defined on the RTP Settings page STUN Parameters The STUN Parameters page enables automatic NAT configuration through the STUN server and is used to configure the STUN Simple Traversal of UDP over NAT client on the QX IP PBX This page requires the following data to be inserted The STUN Server text field requires the STUN server s hostname or IP address The STUN Port text field requires the STUN server port number The Secondary STUN Server and Secondary STUN Port text fields respectively require the parameters of the secondary STUN server The Polling Interval drop down list contains the possible time intervals between referrals to the STUN server NAT Traversal Settings Use STUN Use Manual NAT Traversal Mapped Host 158 168 75 125 Mapped RTP RTCP Port Range Min 6061 Max 6071 Save Fig II 160 NAT traversal Settings RTP Parameters page General SIP Parameters RTP Parameters NAT Traversal Settings Primary STUN server stun epygi com Primary STUN Port 3478 Secondary STUN server stun epygilab am Secondary STUN Port 3479 Polling interval lhour Y Keep alive interval 120 NAT IP checking interval 300 Save Fig II 161 NAT traversal Settings STU
515. while remaining imperceptible With the special feature codes for details see Feature Codes in the Manual III Extension User s Guide you may dial in to the active calls between the other local PBX user and his call partner and depending on the configuration and the feature code used you may listen to the call additionally be able to speak to the extension user only or to all participants This service offers three options e Listen in with this option you may only listen to the third party s call without being able to speak in the call No sound notification will be heard in the third party s call when you dial in e Whisper with this option you may listen to the third party s call and speak to the extension to which you have barged in Only that extension will hear a sound notification when you dial in e Barge in with this option you may listen to the third party s call and speak to all participants in the call All participants of the call will hear a sound notification when you dial in To use the Barge In service options the Barge In feature should be enabled and configured on the extension from User Extension Settings to which you wish to barge in the call Attention Barge In service calls are not displayed in Active Calls table on the Administrator s Main Page nor are registered in the Call History QX50 QX200 QX2000 SW Version 6 0 x 51 QX50 0X200 0X2000 Manual II Administrator s Guide e C Pyg l QX
516. will be automatically submitted by the QX IP PBX system The trusted user will directly access the QX IP PBX Auto Attendant services The SIP settings of the login extension will be used when making IP calls QX50 QX200 QX2000 SW Version 6 0 x 64 QX50 0X200 0X2000 Manual II Administrator s Guide The Automatically Enter Call Relay Menu checkbox enables direct access for the trusted user to the QX IP PBX Auto Attendant Call Relay menu If the checkbox is not selected a trusted caller will be directed to the Auto Attendant s main menu but will still be able to reach Remote Access Voice Mailbox of the specified extension and Call Relay services see Feature Codes with no authentication Please Note Login Extension drop down list and Automatically Enter Call Relay Menu checkbox have no sense for Auto Attendant with custom scenario configured see Attendant Extension Settings The Description text field allows entering an optional comment Callback Settings The Enable Callback checkbox selection gives the possibility for a specified trusted caller to use the Instant Call Back service see chapter Call Back Services The Callback Call Type drop down list includes possible callback call types PBX PSTN SIP and Auto The Callback Destination text field requires the destination number where QX IP PBX should instantly call back to The value inserted in this field is dependent on the selected callback call type for PBX extension numbe
517. will reboot and trial features activate The syntax for values under Count is the following Licensed Features Redundancy Language Pack Feature Keys Free Trial e IP Phone Support the number for additional IP lines Trial Feat Activati e ACD support enables the ACD Management feature support ne nee eee on the system e 3PCC Support enables the 3PCC feature support on the irewal Expiration Date Time Date 19 Dec 2014 Y Time 00 00 Y system i Unique ID 06 6 0 2 000012 Activate Upgrade Name Additional Features Status e Barge In enables the Barge In feature support on the Debug Enables Debug support Activated syste m 3pcc Support Support for Third Party Call Control No Key found ACD Support Support for Automatic Call Distribution Activated e Redundancy enables the Redundancy feature support on Barge In Support for Barge In Activated the QX 2 0 0 0 a Redundancy Support for Redundancy Activated DCC Pro Support Full support for Pro level Deskto 64 users e Call Recording the number for simultaneous call j a recordings DCC Basic Support Full support for Basic level Desktop 32 users Communication Console e DCC Pro Support the number for licensed Pro level DCC extensions iQall Toggling Support Support for iQall toggling 32 users IP Phone support Support for additional IP Phones 1000 users e DCC Basic Support the number for licensed Basic level DCC exte nsions Autodialer Support Support for
518. with the given numbers already exists in the Extensions Management table SIP Server text field requires the address of the SIP server The field is not limited regarding symbol usage and length as it can be either an IP address ora host address e g sip epygi com SIP Port text field requires the port number to connect to the SIP server The SIP Port may only contain digit values otherwise an error message SIP Port is incorrect will appear If the SIP server port is not specified QX IP PBX will access the SIP server via the default 5060 port Registration on SIP Server checkbox enables the SIP server registration option on the newly created extensions User Extension Bulk Import The Extensions Template Management feature and the PC based Bulk User Extensions Importer tool are used to create and update multiple user type extensions The user extension settings can be divided into two groups common settings of extensions groups for example SIP server name SIP port etc and settings which are different for each extension of these groups for example Display Name Extension Password etc Based on this the following three steps can be used to Add Modify a group of extensions e Configure the common settings for a group of extensions using the QX IP PBX Extension Template Management feature e Based on the common settings of these groups configure the extensions specific settings using the Epygi Bulk User Extensions Importer tool
519. x if you wish to run other firmware version on your SIP phone than the one compatible with the QX IP PBX The Configure IP phones from drop down list is used to select the QX IP PBX s interface where the IP phones are connected Besides LAN and WAN this list also includes all defined VLAN interfaces Plesae Note For QX2000 the Configure IP phones from drop down list appears only if VLAN is configured on the QX2000 The Phones Default Template drop down list is used to select the QX IP PBX default template for the IP Phone which will be used if not selected otherwise on the particular line see IP Phone Templates Supported SIP Phones Below is the list of IP phones supported by QX IP PBX and officially compatible with it The Plug and Play PnP and or auto configuration feature is working for all IP phones listed below e Aastra 9112i e Grandstream GXP2000 e SIPUra SPA 841 e Aastra 9133i e Grandstream GXP2100 e snom 190 e Aastra 480i e Grandstream GXP2110 e snom 200 e Aastra 480iCT e Grandstream GXP2120 e snom 220 e Aastra 9143i 33i e Grandstream GXP2124 e snom 300 QX50 QX200 QX2000 SW Version 6 0 x 69 Aastra 9480i 35i Aastra 9480iCT Aastra 6751i Aastra 6753i Aastra 6755i Aastra 67571 571 Aastra 6757iCT 57iCT Aastra 67301 Aastra 673 1i Aastra 6735i Aastra 67371 Aastra 6739i Aastra MBU400 Akuvox SP R53P Alcatel Temporis IP200 Alcatel Temporis IP600 Alcatel Temporis IP800 AudioCodes 310HD AudioCodes 320HD Berk
520. y reached move the oldest recording to FTP server A AA The FTP Settings group is used to define the FTP server settings where the recordings will be uploaded if configured accordingly Server Name text field requires the FTP server name Fig II 59 Extensions Management Edit Entry Recording Box Storage Settings Server Port text field requires the FTP server port number Username and Password text fields require the FTP server authentication parameters Path on Server text field requires the location on the server where the recordings will be stored Naming Scheme text field requires the naming scheme of the files to be uploaded to the FTP server This scheme helps to distinguish files among others and to avoid possible overwriting of the files This text field may contain any distinctive text and also offers a list of variables e caller_dispname caller s display name e caller_username caller s username caller_fullname caller s full name in the username host port format e callee_dispname called user s display name callee_username called user s username callee_fullname called user s full name in the username host port format e duration duration of the call e time_hour hour when the call recording started e time_min minute when the call recording started e time_sec second when the call recording started e date_year year when the call recording started date_
521. y displayed When the Alternative Disconnection Mode checkbox is not selected QX will disconnect the call as soon as the disconnect message has been received from the peer When the checkbox is selected QX s user may hear a busy tone when peer has been disconnected QX50 QX200 QX2000 SW Version 6 0 x 88 QX50 0X200 0X2000 Manual II Administrator s Guide e C Pyg l QX50 QX200 QX2000 Manual Il Administrator s Guide P Asserted Identity The Disable P Asserted Identity radio button disables the P Asserted Identity feature for both incoming and outgoing calls The Override CLID with P Asserted Identity radio button selection enables SIP P Asserted Identity support For the calls from SIP to ISDN if Invite SIP message contains a P Asserted Identity then the CallerID on ISDN is sent with the original Caller ID which comes from the identity field SIP user agent should check for the existence of the P Asserted Identity then the P Preferred Identity then the Remote Party ID to fill the identity field For the calls from ISDN to SIP with restricted Caller ID the SIP Invite message contains P Asserted Identity field with the value from the Caller ID on ISDN The SIP From field contains anonymous The Use Redirecting Number Info Element with P Asserted Identity radio button selection enables full support of the SIP P Asserted Identity For the calls from SIP to ISDN if the SIP Invite message contains a P Asserted Identity or a
522. y the extension s login see Manual III Extension User s Guide Besides this the details of the extension will be displayed in the Public Directories on the snom and Aastra SIP phones Leave this checkbox unselected if the extension is reserved or not used or when the extension serves as an intermediate unit for call forwarding etc The Percentage of Total Memory drop down list allows you to select the space for the extension s voice mails and uploaded recorded greetings and blocking messages The maximum value in the drop down list is equal to the maximum available space for voice messages on QX IP PBX When editing an existing extension and decreasing the voice mailbox size the system will check the present amount of voice mails in the mailbox of the extension If the memory required for these voice mails exceeds the size entered the system will suggest either to remove all voice messages from the extension s voice mailbox or to select a larger size so that the existing voice messages can be stored in the mailbox The Enable Ringing Simulation checkbox is available on virtual extensions only and enables extra ring tones played to the caller before the voice mail of the called virtual extension gets activated If this checkbox is not enabled the voice mailbox will get activated immediately the call arrives The ring tones will be played during the timeout specified in the Ringing Simulation Timeout text field The Edit Call Intercept Access Lis
523. y when the custom call queue welcome message is already uploaded and is used to remove it and restore the default call queue welcome message The Download call queue welcome message functional link appears only when the custom call queue welcome message is already uploaded and is used to download it to PC and opens the file chooser window where the saving location can be specified Upload new call queue message allows updating the active call queue message played when a caller is being held in the queue downloading it to the PC or restoring the default one The Remove call queue message functional link appears only when the custom call queue message is already uploaded and is used to remove it and restore the default call queue welcome message The Download call queue message functional link appears only when the custom call queue message is already uploaded and is used to download it to PC and opens the file chooser window where the saving location can be specified Choose File button opens the file chooser window to browse for a new Call Queue welcome message file The uploaded files should to be in PCMU CCITT u law 8 kHz 8 bit Mono wave format otherwise the system will prevent uploading it with the Invalid audio file or format is not supported warning message The system also prevents uploading if there is not enough memory available for the corresponding extension which will cause the You do not have enough space warning messa
524. zed Phones Database table 4 Press the Delete button on the Authorized Phones Database page 5 Confirm the deletion by clicking on Yes or cancel the action by clicking on No Call Back Services With Call Back service callers can save a call charge when calling to and through QX IP PBX QX IP PBX provides the possibility of creating a list of those trusted callers that are allowed to make free of charge calls to QX IP PBX s Auto Attendant or through its Call Relay menu to the third party SIP or PSTN destination Two types of Call Back services are available on the QX IP PBX Pre configured Call Back and Remote Call Back Configuration Pre Configured Call Back For Pre configured Call Back a list of trusted callers must be configured in the QX IP PBX s Authorized Phones Database using Web Management The Call Back service should be enabled and a valid callback destination should be specified for each caller To use Pre configured Call Back the caller registered in the Authorized Phones Database should simply call to the QX IP PBX s Auto Attendant through SIP or PSTN let the call to ring twice and then hang up Call Back will be instantly activated and QX IP PBX will call back to the defined Call Back destination By answering the incoming call caller will be connected to the Auto Attendant menu Please Note Depending on the call back destination make sure that there is at least one PSTN line routed to the Auto Attendant from the FXO Settin

Manual II: Administrator`s Guide

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