SETU ATA211 System Manual
SETU ATA211 VoIP-FXS-FXO Adaptor System Manual
Documentation Disclaimer Matrix Comsec reserves the right to make changes in the design or components of the product as engineering and manufacturing may warrant. Specifications are subject to change without notice. This is a general documentation for all models of the product. The product may not support all the features and facilities described in the documentation. Information in this documentation may change from time to time.
Contents Introduction..................................................................................................................................................... 1 Welcome! ............................................................................................................................................................ 1 About this System Manual ..................................................................................................................................
Making a Second Call ..................................................................................................................................... 108 Call Transfer .................................................................................................................................................... 109 Call Toggle ...................................................................................................................................................... 111 Call Waiting ......
CHAPTER 1 Introduction Welcome! Welcome to the world of Telecom Solutions. Thank You for choosing SETU ATA211. This product is designed to give you the highest performance, combined with real ease of use. We hope you will make optimum use of this intelligent, intuitive, feature-packed VoIP Phone Adapter. Please read this document before installing SETU ATA211. About this System Manual This document contains detailed information and instructions for installing, configuring, and operating SETU ATA211.
Configuring Advanced Settings: Contains instructions for configuring the more advanced features and facilities of SETU ATA211. Features of SETU ATA211: Describes in detail, the features of SETU ATA211, along with instructions for configuring and using these features. System Status: Describes the status indicators for the LAN and WAN ports, SIP trunk registration, and details of the System Firmware.
The words ATA, ATA211, SETU ATA211, System and gateway means SETU ATA211. Some of the terms specific to this manual that you will encounter freguently are defined below: • Admin: A person who installs, configures and maintains SETU ATA211. Admin can also be the user. Admin has full accessibility of the system. • User: A person who uses SETU ATA211. User has limited access to the system • Caller/Calling party: A person making a call. • Callee/Called party: A person receiving a call.
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CHAPTER 2 Knowing SETU ATA211 The Overview ATA211 is a VoIP-FXS-FXO Adapter. It allows you to make and receive voice calls over IP Network as well as PSTN network using a conventional telephone instrument. ATA211 uses SIP (Session Initiation Protocol) protocol to make voice calls over the IP network. It converts the voice traffic into IP packets for transmission over the internet.
For the ease of configuration, SETU ATA211 has a an embedded, HTTP Web server with an graphic user interface, which you can access by connecting a computer to SETU ATA211. Ports and Connectors Port Name Connector Description Power DC Jack Power Adapter, to connect 12VDC, 1.00A power adapter. LAN RJ45 To connect a standalone computer or a LAN Switch. WAN RJ45 To connect to the IP network over a DSL Modem or Router or a LAN Swtich FXS RJ11 To connect a standard telephone instrument or a PBX.
Applications of SETU ATA211 ATA211 finds its application in following scenarios: 1. Residential Application - Home users use SETU ATA211 to make low-cost international calls to family, friends and relatives. IP Network FXS ATA211 Broadband Modem/Router LAN FXO Proxy PSTN 2.
3. Business Application - Corporate Offices use SETU ATA211 to make low-cost international calls to their overseas customers, employees, suppliers, etc. IP Network Proxy FXS1 PBX WAN ATA211 Broadband Modem/Router Switch LAN FXO PSTN 4. Peer-to-Peer Calls Application - Corporate Offices also use SETU ATA211 to communicate between various branch offices, at no cost..
CHAPTER 3 Getting Started Preparing for Installation Package Contents SETU ATA211 Adaptor 12V, 1.25Amp (Country Specific) Quick Start and User Card Ethernet Cable (RJ45) CD containing System Manual, Quick Start and User Card Two Screws M 7/30 Two Screw Grips RJ11 Line Cord Warranty Card Set 1. Unpack SETU ATA211. 2. Verify the package contents. Contact your vendor/Service Provider/system administrator, if any of the above listed items is missing or damaged.
Safety Instructions Take every measure to reduce the risk of fire, electric shock and injury to the person handling this product. • • • • • • • Read all the instructions given in this manual. Always switch off the product, unplug the product from the wall outlet when handling it. Removing covers or opening the system or incorrect reassembly increases the risk of electric shock. Never open the system in power ON condition. Use the adapter provided with the product.
Connecting SETU ATA211 • Connect a standalone computer/ LAN Switch to the LAN Port of SETU ATA211 using an Ethernet cable. • Connect the WAN Port of SETU ATA211 to the IP Network—a DSL modem or Router or a LAN Switch—using the Ethernet cable provided to you. • Connect a telephone to the FXS Port of SETU ATA211, using a standard telephone cable with an RJ11 plug. You may also connect a PBX to the FXS port of SETU ATA211.
• On successful completion of initialization cycle, each LED will glow as per the call conditions as summarized below: LED Status Meaning FXS Continuously ON Indicates that FXS port is either Off-hook or in speech during incoming call / outgoing call. Blinks fast 200ms ON, 200ms OFF Indicates that FXS port is in Error state during incoming/outgoing call.
Configuring SETU ATA211 SETU ATA211 provides access to system configuration at two levels: Admin and User. A distinct set of features and facilities can be configured at each of these levels. SETU ATA211 can be configured using its Web-based graphic user interface (GUI) and from the phone connected to its FXS port. While the entire system configuration can be done using the GUI, you can configure only certain parameters from the phone.
If required, change the IP Address and the Subnet of the computer. You may also connect the LAN port of SETU ATA211 to a computer on LAN. If you connect the LAN port of SETU ATA211 to a LAN Switch, make sure that the IP Address of LAN port of SETU does not conflict with the IP Address of any device on the LAN, and SETU ATA211 is in the same Subnet as the LAN computer.
On the Login page of Jeeves, you are offered two Login modes: Admin and User. To log in as Admin, • In the Login as box, select Admin. • In the Password field, enter 1234, the default Admin Password. • Click the Login button. • On successful login, the Admin mode page will open. • The left pane shows the links Basic Settings, Advanced Settings, Supplementary Services, and Status. Basic Settings are sufficient to get your SETU ATA211 into operation. • Click the links to expand.
• In the Login as combo box, select User. • In the Password field, enter 1234, the default User Password. • Click the Login button. • On successful login, the User mode page will open. • The left pane displays User Settings and Status links. • Click the User Settings link to expand. The sub-links to the functions and features that are allowed to be configured from the User mode appear on the left pane: Supplementary Features, Speed Dialing, Network, User Password, Syslog, and Factory Defaults.
• Dial the Admin Password (default: 1234). You will get programming tone. • Dial the System Command for the feature/facility you want to configure. For example: To change IP Address of the LAN Port dial 12#* For example, to change the IP address to 192.168.1.120, dial 12192168001120#* • When you dial a valid command, SETU ATA211 plays a confirmation tone for 3 seconds, and plays the programming tone. • Dial 0 to exit configuration mode.
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CHAPTER 4 Basic Settings The Basic Settings enable you to configure SETU ATA211 for basic functions. As Basic Settings cover much of your configuration requirements, you will be able to operate and use the system efficiently, when you configure Basic Settings. To configure Basic Settings, • Click Basic Settings link. The links to the different basic parameters appear on the left pane. There are two ways to configure Basic Settings: • Using the Wizard.
• Click the Wizard icon on the top right of your screen. • The Next button takes you to the next page, saving the changes you made on the current page. • The Back button returns you to the previous page. • The More button expands parameters on the page. • The Less button collapses parameters on the page. • The Default button assigns factory set values to all the parameters on the page. • The Quit button allows you to exit the Wizard at any stage, saving changes you made before exiting.
2. Click More button to view all parameters on the page. 3. Select Admin Language and User Language. Default: English. SETU ATA211 can display the Admin and User pages of the GUI, Jeeves, in the following languages: • English • Italian • Spanish • French • German • Portuguese All the Admin and User Pages of the GUI will appear in the language you selected, when you login again as Admin or User. You can also select a Language on the Login page; however, it will be applied for the current session only. 4.
6. To synchronize Date and Time of SETU ATA211 with that of the country where it is installed, select Time Zone from the drop down list. Default: (GMT+05:30) Kolkata, Chennai, Mumbai, New Delhi. 7. To use a public internet time server for date and time synchronization, select an internet based time server as NTP Server. You may select any of the three reliable public internet time servers1 supported by SETU ATA211: • ntp1.cs.wisc.edu • time.windows.com • time.nist.gov Default: Ntp1.cs.wisc.
SIP Trunks SETU ATA211 supports three SIP trunks, which may be configured as Proxy or Non-Proxy, i.e. Peer-to-Peer trunks. You may register with three different ITSPs and use their services. 1. Click the link SIP. The parameters of SIP Trunk 1 appear on your screen. 2. Click More button to view all parameters of SIP Trunk 1 on this page.
Trunks Settings 1. Select the check box Enable to use the SIP Trunk. Default: Disabled. You may disable the SIP trunk, if you do not want to include it in Call Routing. 2. Select SIP Trunk Mode according to your installation. Default: Peer-to-Peer. • Select Proxy, if you want to register this SIP trunk with an ITSP or a Registrar Server. • Select Peer-to-Peer, if you want to use the trunk for Peer-to-Peer (non-proxy) calls.
• You can configure as many as 500 number strings, which are stored against an Index number. • In the Number field, enter the peer-to-peer number string—prefix or entire number—that will be dialed. The number string must not exceed 24 characters. Default: Blank. If the number to be dialed out is , for example, 123@abc.com, you must enter 1234 in this field. • To identify the number string you configured, enter a name in the Name field.
• the Click Submit to save entries. • Close the window. 3. Enter the SIP ID. This is the ID which remote parties will use to call this SIP Trunk. Default: Blank. The SIP ID may be a number or text consisting of a maximum of 40 characters. If you have defined the trunk mode as Proxy, enter the SIP ID provided by your ITSP. For example, if SIP URI provided by the ITSP is 12345@abc.com, enter 12345 in this field. If you have defined the trunk mode as Peer-to-Peer, enter the desired SIP ID. 4.
The valid range for the port is 1024-65534. Default: 05060. 10. To add ‘rinstance’ in REGISTER? Message, keep this feature enabled. Default: Enabled. ‘rinstance’ is any random value which can be used by the SETU ATA211 to fetch its own contact binding, i.e. to know the Registration Expiry Timer assigned by the server. When you enable ‘rinstance’ in Register, SETU ATA211 will generate any random value of 'rinstance' and include in the REGISTER message.
• Route to fixed destination number through FXO: The system will place all calls received on the SIP Trunk to a Fixed Destination Number through the FXO port. If you select this option, define the destination number in the Fixed Destination Number field. • Route to FXS. If not answered within ring timer, route to fixed destination number through FXO: The system will place all calls received on the SIP Trunk on the FXS port.
• For the number you entered, define the Maximum Digits. The number received must not exceed the maximum digits you have defined here. Default:24 • For the number you entered, select the Destination Route. It may be the FXS or FXO port to route the Called Party number. Click Submit and close the window. To know more see “Called Party Number Table”. 2. Enter the Fixed Destination Number in this field, if the incoming call routing option requires the call to be finally placed at a fixed destination.
• Click the arrow icon. • The PIN Authentication Table page opens in a new window. • Enter the PIN to be authenticated against each Index in the Table. PIN number can be a of maximum 4 digits. The characters allowed are: 0-9,*, #. • • If you have finished entering, click Submit. Close the window. To know more about this feature, see “PIN Authentication”.
To do this, • Click the arrow icon. • The Digest Authentication Table page opens in a new window. • Enter the User ID to be authenticated along with its corresponding Password against each Index in the Table. The User ID may be a maximum of 40 characters. The User Password may be a maximum of 16 characters. • • If you have finished entering the User IDs and their corresponding User Passwords in this Table, click Submit. Close the window. To know more about this feature, see “Digest Authentication”.
Vocoders 1. Select Vocoders in the order of preference from the multiple selection box. Vocoders are the various voice codecs used to compress the data in RTP packets for optimum use of bandwidth and for ensuring voice quality. You can set 4 Vocoder options in the order of preference. The Vocoders supported by SETU ATA211 in the order of preference, i.e. 1st to 4th, listed in the Used Codecs box are: • G.711 A - Law • G.711 Law • G.729 • G.
Advanced 1. Select the default SIP Transport for outgoing SIP messages from the following options: • UDP: Outgoing messages are transported using UDP. • TCP: Outgoing messages are transported using TCP. • TCP (Fallback to UDP): TCP is used for outgoing messages. However, if the TCP connection fails, the system will attempt to send the message again over UDP. Default: UDP To use TCP or TCP (Fallback to UDP), to must enable SIP over TCP in the “System Parameters” page. 2.
• Make a list of numbers that need to be modified before being dialed out from this SIP trunk. Make a list of the corresponding modified numbers that the system should substitute the dialed numbers with. • Click the arrow icon. • The Automatic Number Translation table opens in a new window. In this table, you can store as many as 24 Dialed Numbers and their corresponding Substitute Numbers, at Index Numbers 01 to 24.
• In the Dialed Number column, enter the numbers that need to be modified when dialed out from this SIP trunk. • In the Substitute Number column, enter the numbers which the system should dial out in place of the dialed numbers. Make sure you enter a Dialed Number and its corresponding Substitute Number at the same Index number in the table. For example, if you entered 001 as Dialed Number at Index 01, you must enter its corresponding Substitute Number 1 also at Index 01.
9. Configure the Pause Timer (sec), if you want to insert a delay before the digits of a number string is out dialed from the SIP trunk. The range of the Pause Timer is from 1 to 9 seconds. Default: 2 seconds. 10. Select an appropriate Call Hold Method that is compatible with your ITSP proxy server/remote peer.You may select: • RFC 2543 • RFC 3261 Default: RFC 3261 11.
FXO Port SETU ATA211 supports a FXO Port, • • which may be interfaced with the analog trunk from PSTN and used to route incoming calls to FXS port or SIP trunks. which may be interfaced with FXS port of the PBX. To configure the FXO port parameters, 1. Click the link FXO. 2. Click More button to view all parameters on this page. General 1. Select the Enable check box to use the FXO Port. Default: Disabled. You may disable the FXO Port, if you do not want to include it in Call Routing. 2.
The name you assign may consist of a maximum of 24 characters. Incoming Call 1. Select an appropriate Incoming Call Routing option for the FXO Port. Default: Route to FXS. You may select: • Ignore Incoming Calls: Select this option, if you do want to receive incoming calls on FXO port of ATA211. • Route to FXS: The system will place all calls received on the FXO port on the FXS port. The phone connected to the port will start ringing for the duration of the Ring Timer (default: 45 seconds).
• Route to FXS. If not answered within ring timer, answer the call, collect number & route as per incoming call route selection: The system will place the incoming call on the FXO port to the FXS port. The phone connected to the FXS port will start ringing for the duration of the Ring Timer (default: 45 seconds). If the call is not answered within the timer, ATA answers the call and gives dial tone to the caller.
• For each index number, • Enter the desired Number (max. 24 characters) you want to store in the table. The first Index number 001, is reserved for No Match Found. Start your entries from Index 002. • For the number you entered, define the Minimum Digits, which the system should wait to receive before considering it as a valid number. Default:01 • For the number you entered, define the Maximum Digits, which the system should wait to receive before considering it as End of Dialing.
• • For calls made from the FXS port, select the destination route for the option If dialed number does not match with either of the programmed number in the Dialed Number Table & call is made from FXS, route the call through. Default: SIP Trunk 1. • For calls made from the FXO port, select the destination route for the option If dialed number does not match with either of the programmed number in the Dialed Number Table & call is made from FXO, route the call through. Default: SIP Trunk 1.
• The PIN Authentication Table page opens in a new window. • Enter the PIN to be authenticated against each Index in the Table. PIN number can be a of maximum 4 digits. The characters allowed are: 0-9,*, # • • If you have finished entering, click Submit. Close the window. To know more about this feature, see “PIN Authentication”.
Advanced 1. ATA uses Answer Supervision signaling to indicate to the FXS port that the call made through the FXO port has been answered by the remote party. Answer Supervision is used in the application where a PCO machine is connected to the FXS port of ATA211. This signalling is used to indicate the event of call maturity (when the called party has gone Offhook) so that the PCO machine can start billing.
In the absence of this signal, the call will not be considered as disconnected, even when the caller goes On-hook. This will result in inaccurate billing. To resolve this, ATA211 supports 'Disconnect Signaling Detection'. When there is an outgoing call through the FXO port or there is an incoming call on the FXO port, and call gets matured, if the called party disconnects the call, the disconnect signal will be generated.
In such cases, if the PSTN sends some signaling in the form of Disconnect Tone, and if this is detected by ATA211, ATA can use this to indicate to user 'X' that the remote party 'Y' has gone On-hook, to disconnect the call. • Keep the Disconnection Tone Detection check box to if you want ATA to detect Disconnect Tone of the PSTN. Default: Enabled. • In the Disconnect Tone ON Time and Disconnect Tone OFF Time fields, set the Disconnect Tone Cadence as supported by the PSTN. Default: 0750 msec.
• The Automatic Number Translation table opens in a new window. In this table, you can store as many as 24 Dialed Numbers and their corresponding Substitute Numbers, at Index Numbers 01 to 24. • In the Dialed Number column, enter the numbers that need to be modified when dialed out from the FXO port. • In the Substitute Number column, enter the numbers which the system should dial out in place of the dialed numbers.
8. Configure the Flash Timer, as per your requirement. The Flash timer signifies the time period for which the loop current breaks. SETU ATA211 uses this event to activate various features such as Call Hold, Call Transfer, etc. Default: 600 msec. 9. You may configure the Ring Cadence-OFF Timer for the FXO port to set OFF time for Ring cadence.
• • • • • 600 + 2.16 µF 900 + 1 µF 900 + 2.16 µF 600 + 1 µF Global complex impedance Default: 600 16. If required, you may adjust the Speaking Volume (Transmit) on the FXO port to increase or decrease the volume of your voice being transmitted to the remote party. Select the appropriate Speaking Volume from the drop-down list. Default: 0 db. 17.
FXS Port SETU ATA211 supports an FXS port. Depending upon your requirement, you can either connect an standard, analog telephone instrument to the FXS port, or interface the FXS port with the FXO port of a PBX. To configure the FXS port parameters, 1. Click FXS. 2. Click More button to view all FXS port parameters on this page. General 1. By default Call Routing is enabled on the FXS port. You may disable the FXS Port and if you do not want to include it in Call Routing.
2. You can assign a Name to the FXS port, which will be displayed to the called party (if the instrument of the called party supports display name functionality). The name you assign may consist of a maximum of 24 characters. Default: Blank 3. You can assign a Number to the FXS port. In case of Peer to Peer calls this number will be displayed to the called party. The length of the number string may have a maximum of 24 characters. Default: Blank. Outgoing Calls 1.
• Click the arrow icon. The Dialed Number Table opens in a new window. In this table, entries are stored at Index numbers from 002 to 100. For each index number, • Enter the desired Number (max. 24 characters) you want to store in the table. • For the number you entered, define the Minimum Digits, which the system should wait to receive before considering it as a valid number.
For calls made from the FXS port, select the destination trunk for the option If dialed number does not match with either of the programmed number in the Dialed Number Table & call is made from FXS, route the call through. Default: SIP Trunk 1. For calls made from the FXO port, select the destination trunk for the option If dialed number does not match with either of the programmed number in the Dialed Number Table & call is made from FXO, route the call through. Default: SIP Trunk 1.
• If you have finished entering, click Submit to save. • Close the window. To know more about this feature, see “PIN Authentication”. 3. Select the Auto PSTN Fallback check box, if you want ATA to automatically route the calls through the FXO port when the internet (ethernet) link is down and the SIP trunks cannot be used for call routing. Default: Disabled. Class of Service 1. Select the features of SETU ATA211 that you want to allow in Class of Service4 (CoS) of the FXS port.
Advanced 1. Select the appropriate Answer Signaling Type on the FXS port. Answer Signalling is required when SETU ATA211 has a PCO/Billing machine connected on its FXS port. When a call is made through SETU ATA211 and the called party answers the call, call maturity signal must be generated on the FXS port of SETU ATA211 so that the PCO/Billing device connected to its FXS port can start billing.
Default: Polarity Reversal. • Open Loop Disconnect Timer: If you selected Open Loop Disconnect, you may set the value of this timer, as required. Default: 500 milliseconds. 3. Select the Subscriber Type for SETU ATA211. Default: Gateway. When SETU ATA211is interfaced with a service provider server—ITSP, the Matrix ETERNITY IP-PBX, or any other PBX—that supports supplementary services that require dialing of Flash, like Call Hold, Call Transfer, Call Waiting, etc.
8. If required, you may adjust the Speaking Volume (Transmit) on the FXS port to increase or decrease the volume of your voice being transmitted to the remote party. Select the appropriate Speaking Volume from the drop-down list. Default: 0 db. 9. You may also adjust the Listening Volume (Receive) on FXS port to increase or decrease the volume of the remote party's voice being transmitted to you. Select the required Listening Volume level from the drop-down list. Default: 0 db. 10.
Passwords The system can be programmed by the Administrator or the User, by logging into Jeeves as Admin or User with their respective Passwords. The default Admin and the default User Password are the same, 1234. Since the Admin Login gives the you access to configure all the parameters of SETU ATA211, while the User login has access to only user specific parameters, you may change the default password and use different Passwords for Admin and User Login. The password must not exceed 4 digits.
When you change the Admin Password, you will be logged out of Jeeves. Restoring Default Admin Password? In case you forget the Admin password, you must set the password to the default value. To do this, 1. Lift the handset of your phone. 2. Dial #*** You will get dial tone after successful execution. 3. Replace handset. The Admin password will be set to the default value, if the option ‘Default Admin Password when User defaults the ATA? is enabled in the System Parameters.
4. In User Password, 5. Enter the new password in the New Password field. 6. Type the new password again for confirmation in the Re-Enter to Confirm field. 7. Click Submit to save. To change User Passwords from User Mode: 1. Log in as User. 2. Click the link User Settings on the left pane to expand. 3. Click User Password, 4. Enter the current password in the Old Password field. 5. Enter the new password in the New Password field. 6.
3. Replace handset. Dialing this command string will also reset the Admin Password to factory default, if the option ‘Default Admin Password when User defaults the ATA? is enabled in the System Parameters. To avoid this, make sure you disable this option before you default the User Password.
Network Parameters 1. Click Network. 2. Click More button to view all Network parameters on this page. Routing Mode • Select Routing Mode, according to your installation scenario as NAT or Bridge. Default: NAT. • Select NAT as Routing Mode, if the LAN and WAN port of SETU ATA are in different network segments.
• You will need to configure NAT as Routing Mode, when the SETU ATA211 is at the edge of the network and multiple hosts are connected behind ATA, and you want to share a common internet connection between all hosts. Select Bridge as Routing Mode, if the LAN and WAN port of SETU ATA are in the same network segment. Selecting Bridge mode will disable Network Address Translation (NAT) on the WAN port. When you set SETU ATA211 in Bridge Mode, you need not configure the LAN port IP Address and Subnet Mask.
WAN 1. Select the Connection Type according to the IP addressing scheme of the network which SETU ATA211 is connected: DHCP, PPPoE, and Static. • Select DHCP, if the network uses a DHCP server to assign the IP address, Subnet Mask and Gateway address to SETU ATA211. • Select PPPoE, if the network uses PPPoE. Enter PPPoE User ID (max. 40 characters) and Password (max. 24 characters). You may also enter the PPPoE Service Name (max. 24 characters), if provided.
DNS Server DNS stands for Domain Name Server which is used to resolve domain name into IP address. You can select either Static or Automatic. Default: Static. 1. Select Static if you want to configure DNS manually. • Enter Primary DNS Address. • Enter Secondary DNS Address, if available. Secondary DNS Address will be considered when the request to Primary DNS server fails. • Enter the DNS Domain Name. 2.
Advanced 1. If the SETU ATA211 is connected in VLAN network, configure the VLAN/CoS. This parameter enables the SETU ATA211 to add VLAN header to the packets generated by it. The VLAN header consists of the VLAN ID (12-bit) and Class of Service (CoS, 3-bit) for prioritization of traffic6. • Select the VLAN/CoS check box to enable VLAN ID tagging on all packets generated by the system. Default: Disabled. • Enter the VLAN ID that you have assigned to the VLAN in which the SETU ATA211 is connected.
3. SETU ATA211 will send all the RTP packets with RTP QoS setting, enter the RTP DiffServe/ ToS as per your requirement. Valid range is from 00-63, Default: 46 4. MAC address selection provides you two options: Unique and Clone. Default: Unique MAC Address. If you select Unique MAC Address, the system will use the unique MAC address assigned to it as source MAC address on all Ethernet frames. Select Clone MAC Address, if you want the system to use MAC address other than unique MAC address.
CHAPTER 5 Advanced Settings Advanced Settings include the features and facilities listed below: 1. 2. 3. 4. 5. 6. 7. 8. 9. 10. 11. 12. 13. System Parameters Called Party Number Table Dialed Number Table Peer-to-Peer Numbers Digest Authentication Daylight Saving Time Static Routing Factory Defaults Restart Syslog PCAP Software Upgrade Auto-Configuration System Parameters Certain parameters of SETU ATA211 are applied system-wide and are not specific to a port.
3. Click the link System. The System Parameters page opens. 4.
• System Name: You can assign a name to SETU ATA211, as 'System Name'. This name has significance when multiple SETU ATA211 are connected in the same LAN network. The Name you assign may contain a maximum of 24 characters. Default: Matrix SETU ATA211 • First Digit Wait Timer: The First Digit Wait Timer signifies the time for which the system waits for receiving the first digit after going Off-hook from the FXS Port. On expiry of this timer, the system will give error tone to the user.
Select the Forward Error Correction check box to enable. Default: Disabled. • Voice Activity Detection: Voice Activity Detection is a software application which allows data network, carrying voice traffic over the internet, to detect the absence of audio and conserve bandwidth by preventing the transmission of 'Silent Packets' over the network. Select the Voice Activity Detection check box if you want SETU ATA211 to apply Voice Activity Detection (also called ‘Silence Suppression'). Default: Disabled.
Make sure you configure the NAT Type on the SIP Trunk as STUN. See “SIP Trunks”. • Router's Public IP Address: Routers public IP address specifies the public IP address of the NAT router behind which system is located. Default: Blank. You need to configure this field only if the system is located behind the NAT router and a Static IP Address is assigned as Public IP Address of the Router. Make sure you configure the NAT Type on the SIP Trunk as Router’s IP Address. See “SIP Trunks”.
• 100rel/PRACK: This parameter is to be configured if you want to support reliable transmission of (SIP) provisional responses over UDP. Select the 100rel/PRACK Enable check box, if you want the SETU ATA211 to use 100rel SIP extension for reliable transmission of SIP provisional responses and to use PRACK (Provisional Acknowledgement). Default: Disabled.
Called Party Number Table Called Party Number Table is used for routing incoming calls on the SIP trunks on the basis of the Called Party Number received, without answering the call. You must configure this table, if you have selected Route as per called party number as the Incoming Call Routing option for the SIP trunk. In the Called Party Number Table, you can store a maximum of 10 numbers. Each number is stored at an Index in the table.
To be considered as a valid number, the number received must be equal to or more than the Minimum digits configured. If the number received matches the Minimum digits but exceeds the maximum digits, the system will strip off the extra digits and route the number. If the incoming CLI does not match with any entry in Called Party Number Table (i.e. the number does not exist in the table), the call will be routed through the destination route for the first entry at Index 001, i.e.
• For the number you entered, select the Destination Route. It may be the FXS or FXO port to route the Called Party number. 5. Click Submit save your entries.
Dialed Number Table The Dialed Number Table is a list of numbers or part numbers (you must configure), with a preferred SIP trunk / FXO port / FXS port for each number. When the user dials a number from the FXS port, SETU ATA211 checks the Dialed Number Table for a match using the best fit logic. If a match is found, it uses the Destination Route selected for the number in the table to make the outgoing call. In the Dialed Number table, upto 100 numbers can be stored.
Configuring Dialed Number Table If you have not already configured the Dialed Number Table for the FXS port, you may do so now. 1. Login as Admin with your Password. 2. Click the Advanced Settings link on the left pane to expand. 3. Click Dialed Number Table. The Dialed Number Table page opens. 4. For each Index number, • Enter the desired Number you want to store in the table. The number must not exceed 24 characters; all ASCII characters are allowed.
• At the bottom of the table, select Destination Route for dialed numbers that do not match with any of the entries configured in this table (No Match Found) by the Source Port. • For calls made from the FXS port, select the destination route for the option If dialed number does not match with either of the programmed number in the Dialed Number Table & call is made from FXS, route the call through. Default, SIP Trunk 1.
Peer-to-Peer Numbers Calls made between two IP clients without the intervention of a Proxy Server are defined as Peer-to-Peer Calls (P2P calls). P2P call can be made from FXS or FXO port. As the Peer-to-Peer call application does not require a SIP server, voice communication using this application is 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.
• For the number you entered, in the Destination Address field in the table, enter the IP Address of the WAN Port of ATA connected at Location B. In this case, 190.1.1.100 • For the number you entered, configure the digit length in the table. As the number you want to dial, 3001, has four digits, configure Minimum Digit for this number as ‘4’. Also configure the Maximum Digit as ‘4’.
Configuring Peer-to-Peer Calling To use Peer-to-Peer calling, do the following: • enable the SIP trunk. • set the SIP Trunk Mode to Peer-to-Peer. • configure the number callers must dial to reach this SIP trunk as its SIP ID. • set the Incoming Call Routing option on the SIP trunk as Route to FXS. • configure Outgoing Call Route on the FXS port. • configure Incoming Call Routing on the FXO port. • configure the Peer-to-Peer table.
If the number to be dialed out is , for example, 1234@abc.com, you must enter 1234 in this field. • In the Name field, enter a name to identify the number string you configured. It may be the name of your contact or any name you wish to assign to the number string. The name may consist of 24 characters (maximum). Default: Blank. The name you configure here will not be used in SIP signaling.
Digest Authentication 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 authorised 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 the recipient are connected.
in the Digest Authentication Table you configure in the New Delhi office, you must configure the User Name and Password for the dedicated SIP trunks at the offices in Mumbai, Kolkatta, Chennai. • With Digest Authentication configured at all branches, whenever an incoming call is received on the SIP trunk, the SIP trunk will challenge the identity of the caller. When the digest credentials sent by the calling device matches with the Digest Authentication Table, the call will be allowed on the SIP trunk.
• Enter the user name assigned to the caller/calling device in the User ID field. SETU ATA211 will use this User ID to match the digest credentials sent by the caller/calling devices when challenged. Make sure the User ID you enter here and the User ID assigned at the calling end are the same. The User ID can be up to 40 characters long. Default: Blank. • Enter the password to authenticate the user ID in the User Password field. The password may consist of a maximum of 24 characters. Default: Blank.
Daylight Savings Time Daylight Saving Time (DST) is the practice of advancing clocks so that afternoons have more daylight and mornings have less. Typically clocks are adjusted forward one hour near the start of spring and are adjusted backward in autumn. Many countries of the world9 use it, though the start and end dates of DST vary with location and year.
• Day-Month Wise. Select this option if the DST in your country starts and ends on a particular day of the month. For example, if DST starts on the Second Sunday of March and ends on the First Sunday of October. OR • Date-Month Wise. Select this option if the DST in your country starts and ends on a particular date of the month. For example, if DST starts on October 12 and ends on March 15. Default: Day-Month Wise. 7. If you selected the Day-Month Wise option, configure the Start and End time for DST.
are advanced by one hour at 01:00 hours GMT at the start of DST and set back by one hour at 01:00 hours GMT when DST ends. 1. Select the Daylight Saving Time Enable check box. 2. Set the Time Offset as 60 minutes. 3. Select the option Day-Month Wise as DST Type. 4. Configure the DST Start as follows: • • • • Select 5th as the Ordinal. Select Sunday as the Day. Select March as the Month. Set Time to 01 Hours and 00 Minutes. 5. Now, go to the option DST End, and configure as follows.
Static Routing Table Static Routing Table is required when you have more than one router (Gateway) in your network and you want SETU ATA211 to send packets to multiple routers/gateways for different types of calls. To illustrate this with an example, two Local Area Networks, Network A and Network B, are connected through Frame Relay/ MPLS network to give access to local resources and also to make Peer-to-Peer calls. 59.162.252.82 SIP Proxy 192.168.1.0/24 A B 192.168.2.
The Static Routing Table in SETU ATA211 resolves this. The Static Routing Table defines the appropriate Gateway Address (or Router’s LAN Address) where the IP packets are to be sent. This Table makes it possible to route different types of outgoing calls (Peer to Peer or Proxy) made to different subnets to the Gateway as per the called network. In the Static Routing Table, you must configure: • The address of the final Destination where the call is to be made.
The Static Routing Table allows you to configure up to 5 entries. Each entry is stored against an Index number. For each entry, you must configure the following fields: • Destination Address: This is the address of the final destination where the call is to be made. This can be a device IP Address or Network Address. • Subnet Mask: This is the mask to be applied on destination address. • Gateway Address: This is the IP address of the node where the IP packets are to be sent.
Restart SETU ATA211 If you need to restart SETU ATA211, you may do it using Jeeves, instead of switching OFF and switching ON the system again. When you restart the system, all active calls will be disconnected and the ports in use will be released. The system configuration however, will not be affected. To restart the system using Jeeves, 1. Login as Admin with your Password. 2. Click the link Advanced Settings on the left pane to expand the links. 3. Click Restart. 4.
Reinstate Factory Defaults SETU ATA211enables you to restore factory settings to all the programmable parameters using the option Factory Defaults. You can reinstate Factory Defaults by logging in as Admin or User. The factory set values will be assigned to all parameters depending on the type of login, Admin or User.
An alert message will be displayed, "This option will assign default values to all the programmable parameters and will Restart. Do you want to continue?" 4. Click OK. The system will restart. To restore Factory Defaults from User mode, 1. Log in as User. 2. Click Factory Defaults link under User Settings. 3. Click OK on the alert message: "This option will assign default values to all the programmable parameters and will Restart. Do you want to continue?" The system will restart. Using Commands 1.
Software Upgrade SETU ATA211 provides you the facility to upgrade the system software at the click of a button. SETU ATA211 supports an embedded FTP11 server which can be used for Uploading and Downloading System files, configuration files, drivers etc. To update configuration settings and firmware, 1. Login as Admin with your Password. 2. Click the link Advanced Settings on the left pane to expand. 3. Click System Upgrade. The following message box will be displayed. 4.
• • • ppp: contains files for PPPoE configuration. system: contains the current firmware/software. web: contains html files. 6. Click any of these folders to view the files that SETU ATA211 is currently using. You may either delete the existing files or copy the existing files to another location (as backup). 7. Copy the new configuration files from their location and paste these new files in these folders. Wait for the file transfer to complete. 8. Restart the system after uploading the files.
Syslog (Debug) Debugs are logs of actions and events that take place on any computer system. These logs are useful for troubleshooting and system security. SETU ATA211 supports Syslog12 Client for debugging. Syslog Client enables the system to send debug messages in syslog format to the remote 'Syslog Server' on IP network. You can view the system debug messages on the remote server. Each debug message includes the MAC Address of SETU ATA211 which is sending debug messages in 'syslog server'.
3. Click Syslog. 4. Configure the following parameters: • Syslog: Select the check box to enable. Default: Disabled. When the Debug flag is enabled, the system will send the debug messages to the Syslog Server IP address. Debug report can be viewed on the Syslog Server or any other application which can capture the Syslog messages/debug sent by the system. • Syslog Server Address: Port: Enter the IP Address of the Syslog Server.
To debug the System from User mode, 1. Log in as User. 2. Click the Syslog link, under User Settings on the left pane. 3. To configure Syslog settings, refer the steps 4 and 5 described for the Admin mode.
PCAP Trace PCAP or packet capture consists of intercepting and logging the traffic passing over a digital network or a part of a network. PCAP intercepts each packet in the data streams that flow across the network, and can decode and analyze its contents. PCAP can be used, among others, to monitor the network, analyze network problems, debug client/server communications, debug network protocol implementations.
Refer to the following examples to know how to set the Filters. Examples of Filter settings: • To capture packets of source port 5060, set the filter as src port 5060 • To capture packets of destination port 80, set the filter as dst port 80 • To capture packets transmitted from IP address 192.168.1.176, set the filter as src host 192.168.1.176 • To capture packets received from IP address 192.168.1.176, set the filter as dst host 192.168.1.176 • It is not mandatory to set Filters.
10. Now, you can open the trace files using Wireshark/Ethereal or any other similar software which supports opening of trace files. To use PCAP Trace from User mode, 1. Log in as User. 2. Click the PCAP link, under User Settings on the left pane. 3. To configure PCAP settings, refer the steps 4 to 10 described for the Admin mode.
Auto Configuration You can use Auto-Configuration to have the system automatically download the configuration files stored at a central location: TFTP/HTTP/HTTPS Server. This feature is useful for ITSPs and where multiple SETU ATA211 are connected. ITSPs can store the configuration files of each SETU ATA211 that they provide to their customers on the Auto Configuration Server (ACS). The configuration files of each ATA are to be stored in a folder named by its MAC Address.
3. Click Auto Configuration and configure the following parameter. 4. Auto Configuration: Select the desired method to be used to fetch configuration files. Default: Never. • Never: Select this option if you do not want Auto-Configuration process to be done during power ON. • At Each Power ON: Select this option if you want SETU ATA211 to perform Auto-Configuration at each power ON.
• HTTPS: Select the radio button, if you want this protocol for the process of Auto-Configuration. HTTPS is preferred when security is the concern during Auto-Configuration. 6. Server Address: Specify the Auto-Configuration Server Address. The Auto-Configuration Server Address resembles the IP Address/Domain of ACS where all configuration files of SETU ATA211 are stored. The server address can be configured using DHCP.
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CHAPTER 6 Features of SETU ATA211 Call Hold Call Hold enables you to put an on-going conversation on hold, and call another person or receive a call from another person. You can retrieve the call you put on hold, after the conversation with the other party has ended. You can also retrieve the call you put on hold in the middle of the conversation with the other party.
Making a Second Call You can make a second call, by putting the current call on hold. • If the first call is a SIP call the second call can be a SIP or a PSTN call. But, if the first call is a PSTN call the second call can only be a SIP call. • To use this feature, make sure that the Call Hold feature is enabled in the Class of Service of the FXS port. See “FXS Port”. Making a Second Call • • • • • You are in speech with party A. Dial Flash to put party A on Hold. You get feature tone.
Call Transfer Using Call Transfer you can connect the person you are in speech with, with another party. SETU ATA211 offers two types of Call Transfer: • Call Transfer - Blind: You can transfer the call to the desired party, without informing the party of the transfer. • Call Transfer - Attended: You can transfer the call to the desired party after consulting the party and or obtaining their consent for transfer.
• • • • • • When party B answers the call, you will be in speech with party B. You may now transfer the call if party B consents. Dial Flash-#7 to transfer the call to party B. On successful transfer, you will get confirmation tone. Party A will get connected with party B. Replace the handset of your phone. To know the call transfer result, remain Off-hook after dialing the transfer target’s number (party B).
Call Toggle Call Toggle allows you to have two simultaneous telephone conversations, talking to two persons alternately. You can toggle between: Two SIP calls, a SIP call and a PSTN call. The parties for Call Toggle can be: • Two outgoing calls • Two incoming calls • One outgoing call and one incoming call To use this feature, make sure that Call Toggle is enabled in the Class of Service of the FXS port. See “FXS Port”. Using Call Toggle • • • • • • • • You are in speech with A and you want to talk to B.
Call Waiting When your phone is busy, the Call Waiting feature notifies you about another incoming call in the form of beeps. The Call Waiting feature of SETU ATA211 allows you to: • reject/ignore the waiting call. • hold the current call and attend to the waiting call. • disconnect the current call and attend to the waiting call. Configuring Call Waiting To use this feature, make sure that the Call Waiting feature is enabled in the Class of Service of the FXS port. See “FXS Port”.
2. Click the Supplementary Features link. 3. Keep the Call Waiting check box enabled. 4. Click Submit to save the settings. 5. You may log out. Using Commands • • • • • • Lift handset. Dial #19-Admin Password (default: 1234). OR Dial #18-User Password (default: 1234). Dial 72-1-#* to enable. Dial 72-0-#* to disable. Replace handset. Using Call Waiting When call waiting is enabled and you are busy on a call with party A, • You will hear beeps indicating another incoming call.
• 114 disconnect your call with party A, by dialing Flash-#5 and answer the waiting call. The beeps will stop, you will be in speech with party B. Party A will be disconnected.
Conference Three-party Conference, also referred to as Three-Way Calling, is a telephone call, in which you can have two other persons participate in the call. You can initiate a conference by calling the first person, and then put the first person on hold to call the second person. You can also include another person when you in the middle of speech with a person. Conference can be made between PSTN and SIP calls or between two SIP calls.
Do Not Disturb (DND) If you do not want to receive any calls on your phone, you may set the Do-Not-Disturb feature on your phone. When you set DND, all incoming calls on your phone will be rejected, but you can continue to make outgoing calls. Configuring Do-Not-Disturb To use this feature, make sure that Do-Not-Disturb feature is enabled in the Class of Service of the FXS port. See “FXS Port”. Using Jeeves To set or cancel DND from Admin mode, 1. Log in as Admin. 2.
2. Click the Supplementary Features link, under User Settings on the left pane. 3. Select the Do-Not-Disturb check box to set DND. Default: Disabled. To cancel, clear the check box. Using Phone • • • Lift handset of your phone. Dial #19-Admin Password (default: 1234). OR Dial #18-User Password (default: 234). • • • Dial 61-1-#* to set. Dial 61-0-#* to cancel. Replace handset.
Call Forward When you are away from your phone13, but would like to answer your calls, you can use the Call Forward feature to have SETU ATA211 forward incoming calls to another number. SETU ATA211 supports the following Call Forward options, which you can set on your phone: • Call Forward-Unconditional: All incoming calls received on the SIP Trunk are forwarded to the desired destination number, automatically without waiting for a response from your phone.
3. Click the Supplementary Features link. 4. Select the check box for the Call Forward option you want to set: • • • Call Forward-Unconditional Call Forward-Busy Call Forward-No Reply. In the corresponding box enter the desired destination number (up to 40 characters) you want the calls to be forwarded to. 5. If you selected Call Forward- No Reply, you may also change the duration of the no-reply Timer, if required. The range of the Call Forward No-Reply Timer is 01 to 99 seconds. Default: 45 seconds.
• • Call Forward-Busy Call Forward-No Reply. In the corresponding box enter the desired destination number (up to 40 characters) you want the calls to be forwarded to. 4. If you selected Call Forward- No Reply, you may also change the duration of the no-reply Timer, if required. The range of the Call Forward No-Reply Timer is 01 to 99 seconds. Default: 45 seconds. 5. Click Submit to save.
• Replace handset. Call Forward- No Reply To configure destination number for Call Forward- No Reply: • • • • • Lift the handset of your phone. Dial #19-Admin Password (default: 1234) OR Dial #18-User Password (default: 1234) Dial 55-Destination Number-#*. Replace handset. To set the Timer for Call Forward-No Reply: • • • • • Lift handset of your phone. Dial #19-Admin Password (default: 1234) OR Dial #18-User Password (default: 1234) Dial 57-Time-#*. Valid range of the Timer is 01-99 seconds.
Caller ID Restriction If you do not want your caller ID to be displayed to the called party, you can do so using this feature. Configuring Caller ID Restriction Using Jeeves 1. Log in as Admin with your Password. 2. Click the Basic Settings link on the left pane to expand. 3. Click SIP to open the SIP trunk parameters page. 4. Select the desired SIP Trunk by clicking the SIP trunk number tab. 5. Click More button to view all parameters on the page. 6. Scroll to Advanced. 7.
Selective Trunk Access Regardless of the trunk selected for ‘Outgoing Call Routing’, you can make outgoing calls from any trunk by dialing the Access Code for that trunk. 1. To dial a number from FXO, • • • • Lift the handset of your phone. Dial #83. Dial the desired number. Talk, when the remote party answers the call. 2. To dial a number from SIP1: • • • • Lift the handset of your phone. Dial #84. Dial the desired number. Talk, when the remote party answers the call. 3.
Speed Dialing As the name itself suggests, this feature offers you a quick way to dial a number. Using this feature, you can make an outgoing call to a particular number by dialing a short code instead of dialing the entire number string. Speed Dialing feature works on the basis of the Speed Dialing Table, which you must configure separately for each port. The Speed Dialing Table consists of Index Numbers that serve as short codes.
Configuring Speed Dialing To use Speed Dialing you must configure the Speed Dialing Table separately for each port. To configure Speed Dialing for the FXS port from the Admin mode: 1. Log in as Admin. 2. Click the Supplementary Services link on the left pane to expand the links. 3. Click Speed Dialing. The Speed Dialing Table page opens. You can configure upto 20 entries in this table. Against each index number: • • • • Enter the Name (Max 24 characters) of the contact you wish to speed dial.
2. Click the Speed Dialing link. The Speed Dialing Table opens. 3. To configure the Speed Dialing Table, follow the same instructions as provided for configuring the table from the Admin mode.
Supplementary Services of Service Provider When SETU ATA211 interfaced with a service provider server—ITSP, the Matrix ETERNITY IP-PBX, or any other PBX—that supports supplementary services that require dialing of Flash14, like Call Hold, Call Transfer, Call Waiting, you may choose to access the features of the service provider, or to access primarily, the features of SETU ATA211.
Multi Stage Dialing Multi-stage Dialing is useful when SETU ATA211 connected to a SIP Server being used for networking the PBXs of multiple sites. The extension users of the networked PBXs can dial the entire number string, both the destination number and the extension number of the destination PBX together. The SETU ATA211 will split the string into two stages. It will dial out the destination number first, and on receiving the answering signal from the PBX, it will dial the extension number.
• • At the India office, PBX3 is interfaced with SETU ATA211, which is assigned SIP ID 23. • At the China office, PBX4 is interfaced with SETU ATA211, which is assigned a SIP ID 24. Now, from the office in Canada, where PBX1 is installed, the extension user 101 of PBX1 wants to call extension 102 of PBX2 in Australia.
Configuring Multi-Stage Dialing For this feature to work, at each site where the SETU ATA211 is installed, do the following: • Make sure that Auto Attendant or Direct Inward Dialing is enabled on the FXO port of the PBX on which SETU ATA211 is interfaced. • Enable Automatic Number Translation on the desired SIP Trunk of SETU ATA211. • Configure the Automatic Number Translation Table. • Set the DTMF ON Time, DTMF OFF Time. • Enable ‘Play Routing Tone’ in the System Parameters, if required.
The Automatic Number Translation Table opens in a new window. The Automatic Number Translation Table can accommodate upto 24 Dialed Numbers and their corresponding Substitute Numbers, which are stored against Index numbers 01 to 24. In the Dialed Number column, enter the numbers that users will dial out using the SIP trunk. In the Substitute Number column, enter the number which the system should dial out in place of the number dialed by the users.
When any extension user of PBX1 dials '22 + Extension Number (of PBX2)', for example, 22101 (to reach the extension 101 of PBX2), • SETU ATA -21 will first dial out ‘22’, which is the SIP ID of the called ATA and wait for answer' (22~). The caller will only hear routing tone. • PBX2 on the called side answers the call (DID/Auto-Attendant must be enabled on the FXO port).
Disconnect the Call You can disconnect a call by using an access code (default: #82). When the call disconnect access code is dialed, the system will release the port engaged in the call. Using Disconnect the Call • • • • A is in speech with B A dials #82 (access code for Disconnect the Call). The current call is released. A gets error tone. You can disconnect calls using access codes on the FXO and FXS port only.
Making New Call Making New Call feature enables you to make a new call in the middle of an active call, by dialing an access code. ATA will release the current call and you will get the dial tone, enabling you to dial digits for making the new call. Using Making a New Call 1. A is in speech with B. 2. A dials #81 (access code for Making New Call). 3. The current call is released, A gets dial tone. 4. A can dial C’s number. 5. A is in speech with C.
PIN Authentication SETU ATA211 supports PIN Authentication for outgoing calls from the FXS port and for incoming calls on the FXO port and SIP trunks. You may use PIN Authentication on the FXO port / the SIP trunks / FXS port to restrict access to SETU ATA211 to specific callers. The PIN Authentication feature works on the basis of the PIN Authentication Table, which you must configure.
3. Click PIN Authentication. The PIN Authentication Table page opens. You can configure upto 100 entries in this table. This Table is common for all port types. • Enter the PIN to be authenticated against each Index in the Table. The PIN can have a maximum of 4 digits. The characters allowed are: 0 to 9,*, # • 136 If you have finished entering the PIN Numbers, click Submit to save your entries.
CHAPTER 6 System Status You may view the status of the System, the Network ports (LAN and WAN), SIP Trunks and Auto Configuration from Jeeves. You can also check the status of certain parameters, like the Registration status of SIP Trunks, IP Address and Subnet Mask of the LAN and WAN port, on the display of the Phone connected to FXS port by dialing the relevant “System Commands”. Viewing System Status from Jeeves To view System Status on Jeeves from Admin Mode, • Log in as Admin.
• Log in as User. • Click Status link to open the page. SIP • 138 The following parameters will be displayed for SIP trunks 1, 2 and 3.
• Status: In this field, different status messages may be displayed, each of which are described briefly in the table below. Status Description Disable Shows that SIP Trunk is disabled. Registering Shows that SIP Trunk is enabled and is waiting for response from the SIP server. Active Shows that SIP Trunk is registered with the SIP server. Failed Shows that some error has occurred in the SIP Trunk and no calls can be made using it (applicable only in case of Proxy Account).
Network LAN • IP Address: This field displays the current IP Address of the LAN port. • Subnet Mask: This field displays the current Subnet Mask Address of the LAN port. • MAC Address: This field displays the MAC Address of the LAN port. WAN 140 • • Routing Mode: This field displays the Routing Mode selected, that is NAT or Bridge. IP Address: This field displays the current IP Address of the WAN port. • Subnet Mask: This field displays the current Subnet Mask Address of the WAN port.
• NAT Type: This field displays the NAT Type, if STUN is enabled in SETU ATA211. Commonly used NAT types are: • Unknown • Open • Conenat • Restrictednat • Portrestrictednat • Symmetricnat • Symmetricfirewall • Blocked • IP Address fetched using STUN: This field displays the IP address fetched using STUN, if STUN server address is configured. • SIP Port fetched using STUN: This field displays the SIP port fetched using STUN if STUN server address is configured.
Firmware • Version-Revision: Current Software Version-Revision of SETU ATA211 is displayed here. • Kernel Date: The Date of current Kernel of the system is displayed here. . Auto Configuration • Status: This field displays the type of Auto Configuration option selected. • Next Resync: This field displays when the system will resync next for Auto Configuration. • Last Resync: This field displays the time when the system last resynchronized with the Auto Configuration server.
Go On-hook at the confirmation tone. Your phone rings, and the IP Address will be displayed on your phone as: 192168001001 To view the WAN Subnet Mask, Dial 33-#* and go On-hook at the confirmation tone. Your phone rings and WAN Subnet Mask is displayed as: 255255255000 To view the IP address of LAN port, Dial 32-#* Go On-hook at the confirmation tone.
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Appendix Call Progress Tone Generation Dial Tone Code Country Ring Back Tone Busy Tone Error Tone1 Error Tone2 Freq. Cadence Freq. Cadence Freq. Cadence Freq. Cadence Freq. Hz second Hz second Hz second Hz second Hz second Australia 425*25 cont. 400*25 0.4on 0.2off 0.4on 2.0off 425 0.375on 0.375off 425 0.375on 0.375off 425 1on 1off 02 Argentina 425 cont. 425 1.0on 4.0 off 425 0.3on 0.2off 425 0.3on 0.4off 425 1on 1off 03 Belgium 425 cont. 425 1.0on 3.
Dial Tone Code Country Ring Back Tone Busy Tone Error Tone1 Error Tone2 Freq. Cadence Freq. Cadence Freq. Cadence Freq. Cadence Freq. Hz second Hz second Hz second Hz second Hz Cadence second 32 UK 350+440 cont. 400+450 0.4on 0.2off 0.4on 2.0off 400 0.375on 0.375off 400 0.4on 0.35off 0.225on 0.525off 400 1on 1off 33 USA/Canada 350+440 cont. 440+480 2.0on 4.0off 480+620 0.5on 0.5off 480+620 0.25on 0.25off 480+620 1on 1off Note1: Cont.
CADENCE (In Seconds) Frequency (Hz) T1ON T1OFF T2ON T2OFF Australia 25 0.4 0.2 0.4 2.0 02 Belgium 25 1.0 3.0 03 Brazil 25 1.0 4.0 04 China 25 1.0 3.0 05 Egypt 25 2.0 4.0 06 France 25 1.5 3.5 07 Germany 25 3.5 5.5 0.79 1.1 08 Greece 25 1.0 4.0 09 India 25 0.4 0.2 0.4 2.0 10 Israel 25 2.0 3.0 11 Italy 25 1.0 4.0 12 Japan 25 1.0 2.0 13 Korea 25 1.0 3.0 14 Malaysia 25 2.0 3.0 15 New Zealand 25 2.0 3.0 16 Poland 25 2.0 3.
ATA211. Can I still surf the internet when ATA211 is power down? Ans. No. You can not surf Internet if ATA211 is power down. You must power ON the ATA211 to enable Internet surfing on your PC. Else, connect the PC directly to the internet. Q.5 I am using the ATA211 to make VoIP calls and surf internet. I have connected my PC to LAN port of ATA211. Can I still make VoIP calls when my PC is shut down or not connected? Yes. You can still make VoIP calls when your PC is shutdown or not connected. Ans. Q.
There are two important passwords for the configuration of ATA211 viz Admin Password and User Password. Admin password protects the access to whole configuration while User password protects the access to only few configuration of ATA211. If you forgot the user password but you still have access to admin password, you can login to Web Jeeves and change the user password. But, if you forgot the admin password, you need to default the Password using access code '#***' from the Phone, connected to the ATA211.
Acronyms CLI Caller Line Identification CPT Call Progress Tone CPTG Call Progress Tone Generation CWT Call Waiting Tone DHCP Dynamic Host Configuration Protocol DND Do Not Disturb DNS Domain Name Service DST Daylight Savings Time DTMF Dual Tone Multi-Frequency FEC Forward Error Correction FTP File Transfer Protocol GMT Greenwich Mean Time IC Incoming call IP Internet Protocol ISP Internet Service Provider ITSP Internet Telephony Service Provider ITU International Telecommuni
WAN Wide Area Network VoIP Voice Over Internet Protocol Matrix SETU ATA211 System Manual 151
Features at a Glance Description Feature Code To enter programming mode for Admin #19-Admin Password To enter programming mode for User #18-User Password To exit programming mode 0 Access Code for ‘Using Supplementary Services of Service Provider’, that allows user to use both Gateway and Network types.
Product Specifications Port Name Application No.
Telephony features Voice Calls using SIP proxy and Voice calls without using SIP proxy (Peer-to-Peer Calling) Call Waiting Caller ID Blocking Call Forwarding 3-Party conference Call Transfer Call Hold Caller ID: Generation and Display Answer supervision Disconnect supervision Speed Dialing Do Not Disturb (DND)- Incoming calls can be rejected Polarity Reversal- Useful when some billing machine is connected to ATA Call Toggle- Used to toggle between active and held call Auto PSTN fallback Support If the Ethe
Unit Weight 0.45 kgs (1.10 lbs) Shipping Weight 1 kg (2.20 lbs) Power Supply External Adapter: 12VDC@ 1.25Amp.
System Commands Command Meaning Allowed to? #19-Admin Password To enter programming code for Admin Admin #18-User Password To enter programming code for User User 0 To exit programming mode Admin and User #0 Admin and Access Code for ‘Using Supplementary Services of Service Provider’, that allows user to use both Gateway User and Network types.
Warranty Statement Matrix warrants to its consumer purchaser any of its products to be free of defects in material, workmanship and performance for a period of 15 months from date of manufacturing or 12 months from the date of installation which ever is earlier. During this warranty period, Matrix will at its option, repair or replace the product at no additional charge if the product is found to have manufacturing defect.
out of use of or inability to use such product, even if Matrix has been advised of the possibility of such damages or losses or for any claim by any other party. Except for the obligations specifically set forth in this Warranty Policy Statement, in no event shall Matrix be liable for any direct, indirect, special, incidental or consequential damages whether based on contract or any other legal theory and where advised of the possibility of such damages.
Open Source Licensing Terms and Conditions • The firmware of this product also includes some of the Open-Source software released under GNU General Public License (GPL) Version 2. Terms of this license is printed in full below. GNU GENERAL PUBLIC LICENSE Version 2, June 1991 Copyright (C) 1989, 1991 Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA Everyone is permitted to copy and distribute verbatim copies of this license document, but changing it is not allowed.
We protect your rights with two steps: (1) copyright the software, and (2) offer you this license which gives you legal permission to copy, distribute and/or modify the software. Also, for each author's protection and ours, we want to make certain that everyone understands that there is no warranty for this free software.
distribute such modifications or work under the terms of Section 1 above, provided that you also meet all of these conditions: a) You must cause the modified files to carry prominent notices stating that you changed the files and the date of any change. b) You must cause any work that whole or in part contains or is part thereof, to be licensed as parties under the terms of this you distribute or publish, that in derived from the Program or any a whole at no charge to all third License.
machine-readable copy of the corresponding source code, to be distributed under the terms of Sections 1 and 2 above on a medium customarily used for software interchange; or, c) Accompany it with the information you received as to the offer to distribute corresponding source code. (This alternative is allowed only for noncommercial distribution and only if you received the program in object code or executable form with such an offer, in accord with Subsection b above.
infringement or for any other reason (not limited to patent issues), conditions are imposed on you (whether by court order, agreement or otherwise) that contradict the conditions of this License, they do not excuse you from the conditions of this License. If you cannot distribute so as to satisfy simultaneously your obligations under this License and any other pertinent obligations, then as a consequence you may not distribute the Program at all.
programs whose distribution conditions are different, write to the author to ask for permission. For software which is copyrighted by the Free Software Foundation, write to the Free Software Foundation; we sometimes make exceptions for this. Our decision will be guided by the two goals of preserving the free status of all derivatives of our free software and of promoting the sharing and reuse of software generally. NO WARRANTY 11.
GNU General Public License for more details. You should have received a copy of the GNU General Public License along with this program; if not, write to the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. Also add information on how to contact you by electronic and paper mail.
168 Matrix SETU ATA211 System Manual
Index Numerics 100rel 72 G General Request Timer 72 A AC Impedance 47, 55 I Inter Digit Wait Timer 69 C Call Release Timer 69 Call Transfer-Attended 109 Call Transfer-Blind 109 Class of Service 53, 65 Cloned 66 Connection Type DHCP 63 PPPoE 63 Static 63 Current Date 141 Current Time 141 L Language Selection 20 LEDs 6 D Debug configure system debug 97 Dialed Number Table Configuring Dialed Number Table 74 Destination Route 40, 51, 77 Maximum Digits 29, 40, 51, 74, 77, 28, 40, 51, 74, 77 Number 25, 28,
V VLAN header 65 VLAN/CoS Layer RTP CoS 65 SIP CoS 65 Voice Activity Detection (VAD) 70 VoIP Silence Disconnect Timer 69 Volume 48, 56 W Warranty Statement 157
Magyarországon a Matrix Telecom Ltd. képviselete, Matrix termékek importőre, kizárólagos forgalmazója: 1095 Budapest, Mester u. 34. Telefon: *218-5542, 215-9771, 215-7550, 216-7017, 216-7018 Fax: 218-5542 Mobil: 30 940-1970, 20 949-2688 E-mail: delton@delton.hu Web: www.delton.hu Version 3, February 2011 www.matrixtelecom.
