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SETU VFXTH March 2014

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  • 1. Hearty Welcome!
  • 2. Technical Training SETU VFXTH “VoIP – FXO – FXS Gateway”
  • 3. Overview Interfaces Port Configuration Hardware Architecture LED Indications Installation Do’s and Don'ts Applications Programming using Phone Programming using PC Incoming call Management Outgoing Call Management Advance Settings Maintenance Status
  • 4. A Versatile VoIP-FXO-FXS Gateway A Gateway that provides voice service over IP network using SIP protocol An effective and flexible solution for accessing Internet based telephone services & corporate Internet systems across established LAN Developed to fulfill requirements of SOHO (Small Office Home Office) users & small/ medium scale enterprises Overview
  • 5. Interfaces Overview Port Configuration Hardware Architecture LED Indications Installation Do’s and Don'ts Applications Programming using Phone Programming using PC Incoming call Management Outgoing Call Management Advance Settings Maintenance Status
  • 6. Interfaces
  • 7. Port Configuration Overview Interfaces Hardware Architecture LED Indications Installation Do’s and Don'ts Applications Programming using Phone Programming using PC Incoming call Management Outgoing Call Management Advance Settings Maintenance Status
  • 8. Configurations VoIP Channels FXO Ports FXS Ports FXO Ports Label FXS Ports Label SETU VFXTH0016 16 0 16 0 P01-P16 SETU VFXTH0024 24 0 24 0 P01-P24 SETU VFXTH0032 32 0 32 0 P01-P32 SETU VFXTH3200 32 32 0 P01-P32 0 SETU VFXTH0808 16 8 8 P01-P08 P09-P16 Total 32 SIP Trunks supported in all Configurations SETU VFXTH Configurations
  • 9. Hardware Architecture Overview Interfaces Port Configuration LED Indications Installation Do’s and Don'ts Applications Programming using Phone Programming using PC Incoming call Management Outgoing Call Management Advance Settings Maintenance Status
  • 10. SETU VFXTH1616 Hardware Architecture Input Supply : DC Power Jack 24 V, 2.5A Power Supply OP V : +5V +3.3V, -27V -87V Ethernet Port FXS Modules: Each Module supports 2 extensions to be connected (Total 8) FXO Modules (Total 8) VoIP module : CODEC IC & SDRAM. Total 4 such Modules Each supporting 8 channels 32 BIT RISC PROCESSOR FLASH 32MB 128 MB RAM CPLD
  • 11. LED Indications Overview Interfaces Port Configuration Hardware Architecture Installation Do’s and Don'ts Applications Programming using Phone Programming using PC Incoming call Management Outgoing Call Management Advance Settings Maintenance Status
  • 12. Total 34 LEDs in SETU VFXTH1616 Power LED : At Power On Power LED will Turn On (Continuous Green) 32 Port LEDs : FXO and FXS Port LEDS At Initialization: P01-P32 : OFF After approx 16 sec P01-P03 Glow Continuous Red After approx 20 sec remaining P04-P32 Glow Red Continuous After 5 Sec : P01 - P32 LED will be Off LED Indications
  • 13. 32 FXO/FXS Ports LED indications during normal functioning Continuously Off Port Idle / Disable 400 ms Red on - 200 ms off - 400 ms RED on - 3000 ms off (2 Blinks) Incoming Ring Event 400 ms on- 400 ms off (continuous) Red Off-Hook Event (Dialing State) Continuous On Red Speech LED Indications
  • 14. System LED (STS)
  • 15. Installations Do’s and Don’ts Overview Interfaces Port Configuration Hardware Architecture LED Indications Applications Programming using Phone Programming using PC Incoming call Management Outgoing Call Management Advance Settings Maintenance Status
  • 16. Dust Proof, Moisture Free Location Away from electromagnetic Sources Ventilated Location Path to Static Charges Stable Mains Supply Proper Mains Earth Proper Telecom Earth Installation DO’s
  • 17. Installation DONT’s
  • 18. Applications Overview Interfaces Port Configuration Hardware Architecture Programming using Phone Programming using PC Incoming call Management Outgoing Call Management Advance Settings Maintenance Status LED Indications Installation Do’s and Don'ts
  • 19. Broadband Modem/Router FXO1…FXO16 PSTN Network 2001 2016 FXS1…FXS16 IP Network Ethernet Stand Alone Call Possibilities 1. FXS  IP Network 2. FXS  PSTN Network Stand Alone Application
  • 20. Broadband Modem/Router FXO1…FXO16 PSTN Network FXS1…FXS16 IP Network Ethernet FXS ports of SETU VFXTH are connected to FXO Ports of PBX. Extensions of PBX thus can avail the PSTN & VoIP Networks of SETU VFXTH PBX FXS1…FXSN SETU VFXT1616 In Front of PBX Application
  • 21. Behind the PBX Application Broadband Modem/Router FXO1…FXO16 IP Network EthernetFXO ports of SETU VFXTH are connected to FXS Ports of PBX. Extensions of SETU VFXTH thus can use the Trunk of SETU VFXTH PBX FXS1…FXSN 2001 2016 FXS1…FXS16 PSTN N/W SETU VFXT1616
  • 22. Analog Extension PBX Over IP Application Ethernet PBX FXS1…FXSN 2001 2016 PSTN N/W SETU VFXT1616 FXO1… FXO16 FXO1… FXO16 Ethernet 2001 FXS1… FXS16 FXO1… FXO16 SETU VFXT1616 IP Network Broadband Modem/Router Broadband Modem/Router
  • 23. Peer to Peer Calling Ethernet 2001 2001 FXS1… FXS16 FXO1… FXO16 FXS1… FXS16 Ethernet FXO1… FXO16 2016 IP Network SETU VFXT1616 SETU VFXT1616
  • 24. PSTN Call over IP (Long Distance converted to Local Call) Ethernet 2001 2001 FXS1… FXS16 FXO1… FXO16 FXS1… FXS16 Ethernet FXO1… FXO16 2016 IP Network SETU VFXT1616 SETU VFXT1616 Mumbai Delhi
  • 25. Programming Using Phone
  • 26. Certain parameters of SETU VFXTH can be configured by dialing system commands from a telephone connected to the FXS port You can configure certain network parameters like IP address, Subnet Mask, Connection Type, set the system to default and also view current IP address, Subnet Mask, Connection Type, DNS and Gateway address by dialing system commands Programming Using Phone
  • 27. SE Login Connect Analog Phone to FXS port of SETU VFXTH OFF - Hook the phone Hear Dial Tone [Toooooooooooooooooooooo] Dial Command “#19 – SE Password” for Login Default SE Password is “1234” Enter System Commands to perform different functions Dial “00#*” to Exit from Programming mode
  • 28. Commands 11 – IP Address – #* (To change IP Address) 12 – Subnet Mask – #* (To change Subnet Mask) 10 – Code – #* (to change the connection type) [1 – static, 2 – DHCP, 3 – PPPoE] 31 – Code – #* (To Enable1/Disable0 VLAN Tag) 51 – Reverse SE Password – #* (To Restore Factory defaults)
  • 29. 21 – #* (To view IP Address) & go On – Hook 22 – #* (To view Subnet Mask) & go On – Hook 23 – #* (To view Gateway Address) & go On – Hook 24 – #* (To view DNS Address) & go On – Hook 20 – #* (to view the connection type) & go On – hook 27 – SIP Trunk Number (1 – 9) – #* & go On – hook (To view the status of SIP Trunk) Commands
  • 30. programming Using pc
  • 31. Web Jeeves Login from Local Network Network Switch 192.168.50.200 192.168.50.33 SETU VFXTH is located on Local IP
  • 32. Internet 203.88.123.231 SETU VFXTH is located on Global IP PC with internet connection Web Jeeves Login from Public Network
  • 33. Internet WAN: 203.88.123.231:80 PC with internet connection IP : 192.168.1.151 Subnet : 255.255.255.0 Gateway : 192.168.1.1 LAN: 192.168.1.1 Router’s port:80 is forwarded to IP Address of SETU VFXTH Web Jeeves Login from Public Network
  • 34. Programming Built – in Web server GUI based software called Jeeves Accessible using any web browser Default IP of Ethernet Port is 192.168.001.136 Default SE password is 1234
  • 35. Programming
  • 36. Enter Ethernet Port IP Address of SETU VFXTH Programming
  • 37. Enter Password for Login (Default: 1234) Login Page
  • 38. Home Page
  • 39. Basic Settings
  • 40. Network Port Parameters Select Region of system installation and accordingly Call progress tone country wise
  • 41. • This parameters can be programmed as per existing data network • Connection type : 1. Static: IP address, Subnet mask & Gateway Address assigned Manually 2. DHCP: IP address, Subnet mask & Gateway Address assigned automatically by DHCP server 3. PPPoE: Select this option if your ISP provides internet services using PPPoE, If you select this option you must enter the User ID, password and service name in PPPoE parameters Network Port Parameters
  • 42. Network Port Parameters Select connection type of SETU VFXTH and according to the connection type program the IP details
  • 43. Login Password Password for Jeeves/FTP/Telnet can be minimum of 4 characters and Maximum of 16 characters long All ASCII characters are allowed except white space & ( ) ; “ ‘ < > | dot (.)
  • 44. Date – Time Settings Click on arrow to Set date and time manually Set SNTP server address here to sync date & time with SNTP server
  • 45. MWI (Message Wait Indication on SIP Trunk If you have subscribe for MWI on SIP trunk for the voice mail service by your ITSP then Program Message retrieval number provided by ITSP and port number ion which MWI is to be sent
  • 46. MWI (Message Wait Indication on SIP Trunk
  • 47. Incoming call management - SIP trunk - FXO Port
  • 48. • The process of routing calls originated on FXO port and SIP trunks to the destination port in SETU VFXTH takes place in two steps: 1. Determination of destination number 2. Determination of destination port Incoming Call Route
  • 49. Outgoing on which Number Destination Number determination
  • 50. FXO Port
  • 51. Incoming Call Route options on FXO Port Destination Number Determination on FXO Port
  • 52. 2 different routings defined here 1. Route all IC calls (with CLI) 2. Route all IC calls (without CLI) Without Any Destination Number Define destination port for routing calls
  • 53. • Without any Destination Number • To the Fixed Destination Number • On the basis of Calling Party Number • After answering the call and collecting the digits Destination Number Determination on FXO Port
  • 54. • Incoming call on the FXO port • All calls received on the FXO port are directly routed to the fixed destination port, configured for this port, regardless of the destination number Without Any Destination Number
  • 55. FXO 022 2631725 SETU VFXTH Without Any Destination Number 2001 FXS No destination number will be provided, only Destination port will be applied
  • 56. • Incoming call on the FXO port • Call is routed to the Fixed destination number programmed on that particular trunk line using the Destination port programmed for that trunk • Destination port can be FXS port, FXO port or SIP Trunk Route To a Fixed Destination Number
  • 57. FXO Fixed Destination Number: 471 SIP 471@matrix- pbx.dynalias.org 0265 2630555 Route To a Fixed Destination Number SETU VFXTH
  • 58. • Incoming call on the FXO port • Calls are routed to a specific number according to the calling party number • When there is an incoming call on the FXO port, SETU VFXTH will match the calling party number with the entries of the calling party number based table, if a match is found, the call is routed to the destination number Route on the basis of Calling Party Number
  • 59. FXO SIP Route on the basis of Calling Party Number Calling Number Destination Number 02652630555 471 02226471110 472 SETU VFXTH 471@matrix- pbx.dynalias.org 0265 2630555
  • 60. • Incoming call on the FXO port • Incoming calls are answered and dial tone is played to the caller, allowing the caller to dial the desired number • The number dialed by the caller is considered as the destination number and dial it out using the destination port programmed After Answering the call & collecting the digits
  • 61. FXO After Answering the call & collecting the digits SETU VFXTH Dial Tone SIP 471 471@matrix- pbx.dynalias.org 0265 2630555
  • 62. SIP Trunk
  • 63. Incoming Call Route options on SIP trunk Destination Number Determination on SIP Trunk
  • 64. Destination Port Options on SIP trunk Destination Port Determination on SIP Trunk 2 different routings defined here 1. Route all IC calls (with CLI) 2. Route all IC calls (without CLI)
  • 65. • Without any Destination Number • To a Fixed Destination Number • On the basis of Calling Party Number • To the Called Party Number Destination Number Determination on SIP Trunk
  • 66. • Incoming call on the SIP Trunk • All calls received on the SIP Trunk are directly routed to the fixed destination port, configured for this port, regardless of the destination number Without Any Destination Number
  • 67. SIP 022 2631725 SETU VFXTH 2001 FXS Without Any Destination Number No destination number will be provided, only Destination port will be applied
  • 68. • Incoming call on the SIP Trunk • Calls are routed to the Fixed destination number programmed on that SIP trunk using the Destination port programmed for that SIP trunk • Destination port can be FXS port, FXO port or SIP Trunk Route to a Fixed Destination Number
  • 69. SIP Fixed Destination Number: 0265 2630555 FXO 0265 2630555 Route To a Fixed Destination Number SETU VFXTH 471@matrix- pbx.dynalias.org
  • 70. • Incoming call on the SIP Trunk • Calls are routed to a specific number according to the calling party number • When there is an incoming call on the SIP trunk, SETU VFXTH will match the calling party number with the entries of the calling party number based table, if a match is found, the call is routed to the destination number Route on the basis of Calling Party Number
  • 71. SIP FXO Route on the basis of Calling Party Number Calling Number Destination Number 471 02652630555 472 02226471110 SETU VFXTH 471@matrix- pbx.dynalias.org 0265 2630555
  • 72. • Incoming call on the SIP Trunk • Incoming calls are routed to a desired number depending upon the called number received in the SIP ID of request URI of the INVITE message To the Called Party Number
  • 73. SIP SETU VFXTH FXO 0265 2630555 0265 2630555 To the Called Party Number 203.88.142.221 02652630555@ 203.88.142.221
  • 74. Destination Port Determination Outgoing By which Trunk
  • 75. • SETU VFXTH supports different methods of determining the destination port for the calls originated on FXS Port, FXO Port and SIP trunks, they are: 1. Fixed 2. On the basis of destination number 3. On the basis of calling party number (Not Supported on FXS Port) Destination Port Determination
  • 76. Destination Port Determination on FXS Port Destination Port Options on FXS Port
  • 77. Destination Port Determination on FXO Port Destination Port Options on FXO Port
  • 78. Destination Port Options on SIP trunk Destination Port Determination on SIP Trunk
  • 79. Outgoing Call Management - FXS Port - FXO Port - SIP trunk
  • 80. FXS Port For OG Call we can allow or block outgoing calls, enable flag to Block the Outgoing from this trunk
  • 81. FXO Port Enable flag to Block the Outgoing call from this trunk, Apply ANT with Dialed & substitute number string
  • 82. SIP Trunk Options Related to Outgoing calls through SIP trunk
  • 83. STUN
  • 84. When the VoIP port (WAN) is located behind a NAT Router & SIP Messages need to forwarded to the Public Internet STUN specifies the mechanism required for NAT traversal in SIP messages. STUN server facilitates traversing through most NATs except symmetric NATs STUN (Simple Traversal of UDPs through NATs)
  • 85. Illustration of STUN
  • 86. Illustration of STUN STUN Request STUN Request STUN Response To:115.118.161.163:5060 Payload:115.118.161.163:5060 STUN Response To: 192.168.50.161:5060 Payload:115.118.161.163:5060 Source:192.168.50.161:5060 Source: 115.118.161.163:5060 STUN Server
  • 87. STUN Program the STUN Server IP Address & Port here
  • 88. Select NAT type as STUN if you want to use IP address fetched using STUN STUN
  • 89. Status page will display the IP address, port number and NAT type fetched using STUN STUN
  • 90. Router Public IP Address
  • 91. Port Forwarding: Since STUN doesn’t work with symmetric NAT , as an alternative to STUN Port Forwarding can be done in the router and Router’s Public address that is configured can be used as Source Port IP Address VoIP Port Parameters: Router’s Public IP Address
  • 92. Router Public IP Address Use NAT type as Router Public IP address
  • 93. Router Public IP Address Program Router Public IP Address here
  • 94. Router Public IP Address Status page will display the Router Public IP address programmed in the system parameter page
  • 95. P2P Call One Device is on Public IP and Other Device installed behind NAT 192.168.200.210 Internet SETU VFXTH IP: 192.168.200.195 G/W : 192.168.200.210 Router separates Private and Public Network Private IP Public IP 203.88.142.218 Port Forward in Router LAN port of Router WAN 203.88.142.221
  • 96. *Linksys is a wholly owned subsidiary of Cisco Systems, Inc. Router Configuration : Example Router’s Network Parameters
  • 97. Port Forwarding: Router’s SIP and RTP Ports are forwarded to Private IP of SETU VFXTH *Linksys is a wholly owned subsidiary of Cisco Systems, Inc. Router Configuration : Example
  • 98. Peer to Peer Calling
  • 99. • Making an IP call without the intervention of a proxy server is called peer to peer calling • As peer to peer calling does not require a proxy server, voice communication using this application can be done virtually free of cost • The major cost savings offered by this application makes it a very attractive mode of inter – branch or intra – office voice communication Peer to Peer Calling
  • 100. Peer to Peer Calling Program SIP trunk mode as peer to peer for peer to peer calling Enable SIP trunk
  • 101. Peer to Peer Calling Program the peer to peer table with destination number & destination address (IP address of opposite location) Click here to add new entry to the table Click here to delete entry from the table
  • 102. Proxy Calling
  • 103. Requirement for Proxy Calling Proxy server authenticates the clients for outgoing calls through it What is required for authentication? SIP ID Authentication ID Authentication Password Registrar Server Address Registrar Server port
  • 104. Proxy Calling Select SIP Trunk as Proxy and assign the authentication credentials provided by service provider Enable the flag
  • 105. Proxy Calling If this flag is enabled, SETU VFXTH will send the REGISTRAR MESSAGE to Registrar Proxy as applicable
  • 106. SIP Registration On enabling the flag of SIP Registration, following parameters are to be taken care of This is the time period after which system will send registration request to maintain binding with Registrar Server. Valid range: 00001-65535. Default:3600 Seconds When a registration attempt fails, system resends request to registrar server after this timer’s expiry. Valid range: 00001-65535. Default:10 Seconds
  • 107. SIP Registration System will get unregistered with the current server & will register with the alternate server, if fallback occurs while sending INVITE message when Switch Registration to Alternate Server on Fallback is enabled
  • 108. Registrar Settings If you want the system to send DNS SRV query to the configured domain server, enable this flag
  • 109. What is DNS SRV? Dialing by domain names lets a SIP user have a single public “SIP Address” which can be redirected at will to their current location. SRV records maintain stability and also opens up the possibility to use your own domain, regardless of the domain of the SIP service you are using
  • 110. SIP Registration Enable the flag, if your service provider supports multiple servers in its network
  • 111. Advance Settings
  • 112. • Access code is a string of digits dialed to use a feature • SETU VFXTH users can access the features and facilities by dialing the access code assigned to them from a phone. User can 1. Enable/Disable a feature 2. Access Supplementary feature 3. Enter into the programming mode • SETU VFXTH provides default access code for all features, you can change it to suit your preferences Access Codes
  • 113. Access Codes Access codes can be changed from here
  • 114. Access Codes Access codes can be changed from here
  • 115. • This feature provides the flexibility to allow or deny dialing of a particular number or a set of numbers from a particular port or all ports • Allowed Denied number logic makes use of two number lists: 1. Allowed Numbers List: this is the list of numbers that can be dialed out from the SIP trunk (default number list – 7) 2. Denied Numbers List: this is the list of numbers that are to be restricted from being dialed out from the SIP trunk (default number list – 8) Allowed – Denied Numbers
  • 116. Allowed – Denied Logic on FXS Port Apply allowed denied list on FXS Port & program the number list for allowed & denied numbers
  • 117. Apply allowed denied list on FXO port & program the number list for allowed & denied numbers Allowed – Denied Logic on FXO Port
  • 118. Allowed – Denied Logic on SIP Trunk Apply allowed denied list on SIP trunk & program the number list for allowed & denied numbers
  • 119. • This feature is used to translate the dialed number string to preprogrammed number string • ANT can be used to modify, add or delete the prefix of the destination number string • For this feature we need to configure dialed number string and substitute number string in number list table • ANT feature is applied on destination ports (On all SIP trunks and FXO Ports) Automatic Number Translation
  • 120. Apply ANT on FXO port and program the table number Automatic Number Translation
  • 121. Automatic Number Translation Apply ANT on SIP Trunk and program the table number
  • 122. ANT table Automatic Number Translation Examples on how to program
  • 123. • SETU VFXTH supports feature ‘Black listed Callers’ which enables you to block incoming calls from specific numbers and addresses on the SIP trunks • This feature is applicable on source port only • To use this feature, user must configure the numbers of unwanted callers in a number list • Enable the Reject Calls from Blacklisted Caller check box on the SIP trunks on which you want to apply this feature Black Listed Callers
  • 124. Black Listed Callers Apply black listed caller feature on selected SIP trunk and define the number list for the same Black Listed Callers
  • 125. • It’s a record for the calls, containing information about the gateway’s usage when call was made • Maximum of 2000 call record entries can be stored • Call record entries are stored in FIFO logic • User can set different filters as required and generate Call Detail Record (CDR) report • Call records can be cleared manually at any time Call Detail Record (CDR)
  • 126. • It is possible to get following details of a call with CDR 1. Date of call origination 2. Time of call origination 3. Calling number 4. Called number 5. Duration of call 6. Source port 7. Destination port 8. Disconnected by 9. Cause 10. PIN number 11. Remarks Call Detail Record (CDR)
  • 127. • Below mentioned filter can be programmed for CDR 1. The port from which the calls originate (Source Port) 2. The port on which the calls terminate (Destination Port) 3. Calls made on particular dates 4. Calls made at a particular time 5. Calls of a certain duration 6. Calls of certain called party numbers 7. Calls of certain calling party numbers 8. Calls made with PIN authentication 9. Calls made without PIN authentication Call Detail Record (CDR)
  • 128. Call Detail Record (CDR) Set filter parameters for CDR here Click here to clear all call records Click on download to get Zip file containing CDR in .csv and .txt format
  • 129. CDR can also be viewed from Jeeves Call Detail Record (CDR)
  • 130. • PIN authentication is a security feature to restrict access to the system and prevent possible misuse of resources • User can use the PIN authentication on the source port to establish identity of callers before their call is processed by SETU VFXTH • PIN authentication can be used on the source port only if the incoming call routing for the source port is set to After answering the call and collecting digits • To use this feature it must be enabled on the source port and the PIN authentication table must be configured PIN Authentication
  • 131. • The PIN authentication table stores up to 500 PIN numbers and their corresponding authentication passwords • If PIN authentication is enabled on source port, SETU VFXTH answers the Incoming call and plays a feature tone, it waits for the caller to dial the PIN number and password, it matches them with the PIN authentication table, if match is found it allows the call to be processed • In case of wrong PIN entered, SETU VFXTH allows the caller to make two more attempts, if the caller fails to dial correct PIN and password in all attempts, the system disconnects the call PIN Authentication
  • 132. PIN Authentication – FXO Port Select routing type ‘after answering the call and collecting the digits’ for PIN authentication feature to use Enable this flag for prompting caller to enter PIN
  • 133. PIN Authentication Enter PIN number & PIN password, system checks PIN entered by the caller during call with the entries in the PIN authentication table, if match found then only the call will be processed further
  • 134. • Digest authentication is a challenge – based authentication service of SIP to authenticate the identity of the originator of SIP request in the INVITE message • The recipient of the request can ascertain whether or not the originator of the request is authorized to make the request • When the digest credentials of the originator – User Name and Password – in the INVITE message are authenticated and accepted by the recipient, the originator and recipient are connected • You may use the digest authentication to restrict access to SETU VFXTH to specific callers, prevent unwanted or malicious calls Digest Authentication
  • 135. • When this feature is enabled on a SIP trunk for all Incoming calls 1. SETU VFXTH will challenge the identity of the calling party 2. When the calling party sends its credentials, SETU VFXTH authenticates the credentials by matching it with its Digest Authentication table 3. If a match is found, the calling party will be authenticated and the call will be allowed on the SIP trunk 4. If no match is found, SETU VFXTH will consider it as invalid authentication information and reject the call Digest Authentication
  • 136. Enable apply flag in SIP trunk to use digest authentication Digest Authentication
  • 137. Enter Digest credentials (User ID and User Password) of calling party Digest Authentication
  • 138. • Static Routing Table is required when you have more than one router (Gateway) in your network and you want SETU VFXTH to send packets to multiple routers/gateways for different types of calls • If you have only one router connected in the network , you need not configure this table & LAN interface of router will act as the default gateway for the system Static Routing
  • 139. Program the static routing table with the details, if the match is found here then gateway will send the packets to defined gateway address opposite to the destination address Static Routing
  • 140. • Prefix to domain name conversion is used when a user sets call forward or makes a blind transfer on SIP, this feature is applicable only when the destination port is SIP • SETU VFXTH supports multiple SIP trunks & FXS ports, when a FXS port user dials a SIP number, SETU VFXTH routes the call to the IP network using the SIP trunk determined by the routing mechanism. The FXS user can dial only numbers not domain names, therefore it becomes necessary that the domain names be assigned prefix codes which the FXS user can dial Prefix to Domain Name Conversion
  • 141. • User need to program prefix v/s domain name in the table • This table is not checked for making an outgoing call, but it is checked when some FXS port has set call forward and only number is programmed or user is doing blind transfer • For example prefix in the table is programmed as *123 and domain name as abc.com and destination number for call forward is *1239974 then it will be replaced by 9974@abc.com Prefix to Domain Name Conversion
  • 142. Define prefix and domain name in the table Prefix to Domain Name Conversion
  • 143. • If call disconnection is signaled by your CO network in the form of disconnect tone on the FXO Ports • You must enable Disconnect Tone Detection on the FXO port and select the Disconnect tone type • To enable the system to detect the disconnect tone accurately, you must configure the cadence and frequency of the disconnect tone type you selected, as supported by the CO network Disconnect Tone
  • 144. Enable disconnection tone detection here Disconnect Tone
  • 145. Disconnect Tone Program the disconnect tone cadence here
  • 146. • SETU VFXTH supports dialing of emergency numbers from all ports, Emergency numbers and their respective routing groups must be configured in the emergency number table • User can configure up to 10 numbers of emergency services such as ambulance, fire brigade, police etc. • By default, No emergency numbers are loaded in the system, in the emergency number table Emergency Numbers
  • 147. Click here to add new entry to the table Click here to delete entry from the table Click here to Edit entry of the table Emergency Numbers
  • 148. Features
  • 149. • If any FXS port want to use supplementary services then these services must be activated in COS for particular FXS port as well as at SIP services provider in case of SIP account calling • SETU VFXTH offers following telephony features, which they can access by dialing access codes 1. Call Hold 6. Blind Transfer 2. Call Forward 7. Attended Transfer 3. Call toggle 8. Do Not Disturb (DND) 4. Call waiting 9. Hotline 5. Conference Class Of Service
  • 150. Enable required feature from Class of service on particular FXS port Class Of Service
  • 151. Enable the supplementary services after enabling the feature in COS Supplementary Services
  • 152. • When SETU VFXTH is interfaced with service provider server – ITSP or other PBX that supports supplementary services that require dialing of Flash like call hold, call transfer, call waiting, you must select the subscriber type according to the extent of feature access you want on the FXS port connected to the system Subscriber Type
  • 153. • Select Network if you want to use supplementary services supported by the other PBX, you can access the service provider features by dialing FLASH, you will not be able to access the local features of SETU VFXTH • Select Gateway if you want to use supplementary services supported by the SETU VFXTH, in the gateway mode you will also be able to access the supplementary services of the service provider which require dialing of FLASH Subscriber Type
  • 154. Select the subscriber type of your choice Subscriber Type
  • 155. Signaling Loop Start Connector RJ-45 Off-Hook Line Impedance 600 Ω / 900 Ω / Complex No. of Long Loop Extension 4 Loop Limit 1800 (Max) Excluding Telephone Set On-Hook Voltage (Tip/Ring) -48 V Off-Hook Current 25 mA (Max) Ringing Voltage Trapezoidal 60 VRMS/25Hz and Sinusoidal 52VRMS/25Hz FXS Port
  • 156. REN 3 DTMF Detection ITU-T Q.24 CLI Presentation DTMF, FSK ITU-V23 & FSK Bellcore Protection Over Voltage Secondary Protection Return Loss >18 dB Longitudinal Balance >50 dB Transmission Level Adjust Tx Gain : -3dB to +6dB; Rx Gain : -3dB to +6dB Answer Signaling on FXS Battery Reversal Disconnect Signaling on FXS Battery Reversal & Open Loop Disconnect FXS Port
  • 157. Hardware settings on FXS port FXS Port
  • 158. General settings on FXS port FXS Port
  • 159. First Digit & Inter Digit wait timer FXS Port
  • 160.  First Digit Wait Timer: • Signifies the time for which the system waits for receiving a first digit after going off – hook from FXS port • On expiry of this timer, system will give error tone to the user • It is programmable from 01 to 99 seconds (Default: 15 seconds) FXS
  • 161.  Inter Digit Wait Timer: • Signifies the time period between 2 consecutive digits while the system is receiving the digits from caller • On expiry of this timer, ATA1S will process the digits dialed so far by the user • it is programmable from 01 to 99 seconds (Default: 5 seconds) FXS
  • 162. Return Loss >18 dB Longitudinal Balance >50 dB Transmission Level Adjust Tx Gain: -15 dB to +10 dB Rx Gain: -15 dB to +10 dB Call Maturity Delay & Polarity Reversal Answer Supervision on FXO Battery Reversal Disconnect Supervision on FXO Battery Reversal & Open Loop Disconnect FXO Port
  • 163. REN 3 DTMF Detection ITU-T Q.24 CLI Presentation DTMF, FSK ITU-V23 & FSK Bellcore Protection Over Voltage Secondary Protection Return Loss >18 dB Longitudinal Balance >50 dB Transmission Level Adjust Tx Gain : -3dB to +6dB; Rx Gain : -3dB to +6dB Answer Signaling on FXS Battery Reversal Disconnect Signaling on FXS Battery Reversal & Open Loop Disconnect FXO Port
  • 164. Hardware settings on FXO port FXO Port
  • 165. FXO Port General settings on FXO port
  • 166. • This feature enables callers to disconnect the current call and make a new call using SETU VFXTH without getting disconnected from the system • This feature is useful when you want to make multiple calls without getting disconnected each time their call ends • This feature is applicable only on the FXO port and only when After answering the call and collecting digits is selected as the destination number determination method • If you have enabled Connect source port when number is out dialed on the FXO port, you will not be able to provide this feature to callers Making a new call using access code
  • 167. • To make a new call using access code  In speech with the current call  Dial #91  Current call will disconnect  Dial the new number you want to call  Speech will be establish on the new call as called party answers the call  While in speech dial #91 again to make another new call Making a new call using access code
  • 168. Enable the flag to allow user making new call using access code Making a new call using access code
  • 169. • SETU VFXTH enables user to disconnect a call using an access code • When the call disconnect access code is dialed, SETU VFXTH releases the port engaged in the call • This feature is applicable only when destination number determination method is selected as After answering the call and collecting digits • If you have enabled Connect source port when number is out dialed on the FXO port or have enabled Connect source port when 183 is received on SIP on the SIP trunk, you will not be able to provide this feature to users Disconnecting a call using access code
  • 170. Disconnecting a call using access code Enable the flag to allow call disconnection using access code
  • 171. Enable the flag to allow call disconnection using access code Disconnecting a call using access code
  • 172. • SETU VFXTH supports direct dialing of IP addresses from the source port. To provide IP dialing facility to the users, you must configure a SIP trunk or a SIP group for IP dialing • IP number can be dialed with dot ’.’ as entered by ‘*’ while dialing it • For e.g. to dial IP address 192.167.100.1 dial as 192*167*100*1 from the Phone at FXS • When an IP address is dialed from the source port of SETU VFXTH, the system does not check the destination port determination method you have configured for that port, instead it routes the dialed IP address through the SIP trunk or SIP group you configured for IP dialing IP Dialing
  • 173. SIP trunk or SIP trunk group can be defined for IP dialing IP Dialing
  • 174. 100rel and SIP PRACK SIP PRACK (SIP Provisional Acknowledgement) is a method to enable reliability for SIP 1XX messages The Called Party answers the PRACK by 200 OK and PRACK is only for 1XX messages other than 100 Trying Generally PRACK message flows from Calling Party to Called Party
  • 175. System Parameters Enable Provisional acknowledgement for all 1xx messages other then 100 trying
  • 176. SIP Timers • SIP Invite Timer • SIP Provisional Timer • General Request Timer
  • 177. SIP Invite Timer • It is the time for which SETU VFXTH waits for a response from the called party after sending INVITE message • This time starts after sending INVITE message to the called party and stops on receipt of provisional response or final response or when the user goes ON- Hook, on expiry of the timer the call is disconnected • The range of SIP INVITE Timer is 10 - 80 seconds (Default: 30 Seconds)
  • 178. SIP Provisional Timer • It is the time for which SETU VFXTH waits for final response after receiving provisional response from the called party • This timer starts on receipt of provisional response from the called party and stops on receipt of final response from the called party or when the user goes ON-Hook, on expiry of the timer the call is disconnected • The range of SIP Provisional Timer is 10 - 180 seconds (Default: 60 Seconds)
  • 179. • It is the time for which SETU VFXTH waits for the response of a transaction request • This timer starts on initiating a transaction • This timer stops on receipt of a response for the request • On expiry of timer, the SETU VFXTH clears the transaction • The range of SIP Provisional Timer is 10 - 60 seconds (Default: 20 Seconds) General Request Timer
  • 180. System Parameters Program the timer values according to the requirement
  • 181. SIP Over TCP • The SIP over TCP option allows you to send/receive the SIP messages over TCP • SIP over TCP is applicable for both Proxy and Peer to Peer • By Default SIP messages transported over TCP • Disable the flag to send SIP messages over UDP
  • 182. SIP Over TLS • The SIP over TCP option allows you to send/receive the SIP messages over TLS. TLS protects SIP signaling against loss of integrity, confidentiality and against reply • SIP over TLS is applicable for both Proxy and Peer to Peer • By Default SIP over TLS is enabled • Disable the flag to disable SIP over TLS
  • 183. System Parameters Program SIP TCP, UDP and TLS port values. Default: 5060 for TCP, UDP and 5061 for TLS
  • 184. Server Port Server ports can be changed to any value from 1021 to 65,535
  • 185. Management/Security Server ports can be changed to any value from 1021 to 65,535
  • 186. Certificate
  • 187. Certificate • SETU VFXTH supports certification for TLS, Web Server, Firmware Upgrade, Configuration Upgrade and TR-069. • SETU VFXTH supports two types of Certificates: Self-Signed Certificate and CA Signed Certificate.
  • 188. Self – Signed Certificate • A self-signed certificate is created by the clients themselves or by the Servers and then given to their clients. • It means that you yourself become the Certificate Authority (CA), create a CA Certificate and sign it. • The self-signed certificate is faster to create but is not signed by a trusted CA Organization. • The self-signed certificate must be installed in the trusted list of clients that connects over TLS with the Server. Because the certificate has been self signed, the signature is not likely to be in the clients’ trust file, hence, they need to add it.
  • 189. Certificate Generate self signed CA certificate by entering the required details below Once you generate self-signed certificate, you must send it to your clients so that they install it in their trusted list. Click generate to generate new certificate for entered details
  • 190. Certificate System will show generated certificates under trusted root CA
  • 191. System Certificate • After creating a Self-Signed CA Certificate, you can either, • Generate a System Certificate for your clients. These System Certificates can then be given to the respective clients OR • The Clients can prepare their own System Certificates. For this you need to send them the CA Certificate created by you OR • Generate a Certificate Signing Request (CSR), if you want the Certificate to be signed by a third party If the clients prepare their own certificates, you need to send your CA Certificate to all the clients. The clients must upload the same in their system. Similarly, all the clients must send their CA Certificates to you and you must upload the same in your system. To avoid this, it is recommended that you create the Certificates and then provide it to your clients
  • 192. Enter details to generate system certificate If you want to get a CA Signed Certificate, you need to do the following: 1. Generate and enroll the Certificate Signing Request (CSR). 2. Get the Certificate Signing Request (CSR) verified and signed by the Certified Authority (CA).
  • 193. Certificate List of available system certificates User can also upload the certificates
  • 194. Certificate Define the certificate to be used for desired application
  • 195. Maintenance
  • 196. Firmware Browse the ZIP file having new firmware files & click on Upgrade button to upgrade the system firmware Program the details for Auto firmware upgrade Upgrade firmware automatically from Matrix Server
  • 197. Configuration Browse the ZIP file having configuration files & click on Upgrade button to upgrade the system configuration Program the details for Auto configuration Click on Backup Configuration to save config.zip file
  • 198. • Debugs are logs of actions and events that take place on system, these logs are useful for troubleshooting and system security • SETU VFXTH supports Syslog client for debugging, Syslog client enables the system to send debug messages in Syslog format to the remote ‘Syslog server’ on the IP network • Syslog uses the UDP as transport protocol • To be able to use this feature, you must enable ‘Syslog’, configure the Syslog Server Address and define the server port on which the Syslog will listen for debug messages System Debug
  • 199. System Debug ug events be viewed he screen Click debug settings to set parameters for debug and to start debug in PC/Laptop connected to SETU VFXTH
  • 200. System Debug Program the IP address and port number of PC/Laptop where Syslog server is installed Debug for Port: clear the check box to disable the debug for the port which is not needed
  • 201. • SNMP – Simple Network Management Protocol • SNMP protocols supported – SNMPV1, SNMPV2C, SNMPV3 • SETU VFXTH is having built in SNMP Server (SNMP Server). It receives SNMP requests and generates SNMP responses or notifications • SNMP Manager usually network management station. It generates SNMP requests and receives SNMP responses and notifications. The SNMP manager is an SNMP client SNMP
  • 202. SNMP Program SNMP details
  • 203. System Port Activity System port activity like Idle, Inactive, Disable, Dial, Speech, ringing, Incoming Call Proceeding, Remote Held, Error
  • 204. • PCAP or Packet capture consists of intercepting and logging the traffic passing over the network, PCAP intercepts each packet in the data streams that flow across the network, and can decode and analyze its contents • A maximum 2MB of packets can be captured and stored in the system • SETU VFXTH also supports filters and promiscuous mode for capturing packets • If promiscuous mode is enabled, SETU VFXTH will capture all network traffic and if disabled then system will capture only traffic that is directly related to SETU VFXTH (to or from SETU VFXTH) PCAP Trace
  • 205. PCAP Trace Click here to start the PCAP trace Click here to stop the PCAP trace Once the PCAP is captured save the trace file on your PC/Laptop Click here to Enable Promiscuous mode Enter the filter details here
  • 206. • Select source port and destination port with source number and destination number. • When Call button from GUI is pressed system will call source number first and when answered by source port it will ring on destination port & speech path can be checked • Clicking on call button will also lead the programmer to system port activity page to monitor the status of the port during call progress Manual Call Test
  • 207. Manual Call Test
  • 208. • SETU VFXTH supports the AC Impedance Test for clear, audible and echo-free speech over FXO Ports. • This test helps you to set the most appropriate values for the FXO Port Parameters —AC Termination Impedance, CO Termination and CO Line Type— to correct the line impedance mismatch between the AC Termination Impedance presented by the FXO Port of SETU VFXTH to the line and the CO Termination Impedance presented by the Central Office to the line. • While the test is being conducted, you will hear pulsating tone on all the ports of the system. (Mute the microphone of destination landline number or mobile number when call is answered by destination number) AC Impedance Test (FXO)
  • 209. AC Impedance Test (FXO) Enter phone number on which system will make the call in order to complete the test Select the FXO port on which you want to run the test Click on start test and wait for the results For more details click help
  • 210. Default System Click OK to factory default the gateway
  • 211. Soft Restart Click OK to Restart SETU VFXTH
  • 212. • TR-069, also known as CPE WAN Management Protocol (CWMP), is a remote management protocol used for secure communication between the Customer Premises Equipment (CPE) and an Auto-Configuration Server (ACS) for various functionalities such as:  Auto-configuration and dynamic service provisioning  Firmware Management  Status and performance monitoring  Diagnostics • SETU VFXTH supports TR-069. Using TR-069, service providers can manage SETU VFXTH remotely for the functions described above. TR – 069
  • 213. TR – 069 Program TR-069 details
  • 214. STATUS
  • 215. System Detail Version Revision details
  • 216. Firmware Status Last Firmware up gradation details if scheduled firmware upgrade is ON
  • 217. Configuration Status Last Configuration up gradation details if scheduled firmware upgrade is ON
  • 218. Network Status IP details status of IP configured in SETU VFXTH
  • 219. FXO Port Status Line connection status on FXO
  • 220. SIP Trunk Status SIP trunk Status