SETU ATA2S System Manual
|
|
|
- Marcus Jennings
- 10 years ago
- Views:
Transcription
1 SETU ATA2S System Manual Magyarországon a Matrix Telecom Ltd. képviselete, Matrix termékek importőre, kizárólagos forgalmazója: 1095 Budapest, Mester u. 34. Telefon: * , , , , Fax: Mobil: , [email protected] Web:
2 Documentation Information This is a general documentation and it covers many models with different specifications. A particular product may not support all the features and facilities described in the documentation. Matrix Telecom reserves the right to revise information in this publication for any reason without prior notice. Information in this documentation may change from time to time. Matrix Telecom makes no warranties with respect to this documentation and disclaims any implied warranties. While every precaution has been taken in preparation of this system manual, Matrix Telecom assumes no responsibility for errors or omissions. Neither is any liability assumed for damages resulting from the use of the information contained herein. Matrix Telecom reserves the right without prior notice to make changes in design or components of the equipment as engineering and manufacturing may warrant. Neither Matrix Telecom nor its affiliates shall be liable to the purchaser of this product or third parties for damages, losses, costs or expenses incurred by purchaser or third parties as a result of: accident, misuse or abuse of this product or unauthorized modifications, repairs or alterations to this product or failure to strictly comply with Matrix Telecom s operating and maintenance instructions. All rights reserved. No part of this system manual may be copied or reproduced in any form or by any means without the prior written consent of Matrix Telecom.
3 Contents Section 1: Introduction... 5 Welcome... 7 Packing List... 9 Warranty Statement Protecting the System Introducing the System Getting Started Section 2: Features and Facilities Auto Configuration Dial Plan FXS Port Jeeves Password Peer-to-Peer Calling Phone Book Resetting the ATA SIP Accounts Software Upgrade Status Supplementary Services System Name Time Settings VoIP Basics WAN Explanation WAN Port Section 3: Appendices Appendix A: Troubleshooting Appendix B: Applications Appendix C: Product Specifications Appendix D: Glossary Appendix E: Features at a Glance Appendix F: System Commands
4 Index Notes Programming Register
5 Section 1: Introduction
6
7 Welcome Matrix Welcome to the world of telecom solutions from Matrix and thanks for purchasing a Matrix product. We want you to get the maximum performance from our product. If you run into technical difficulties, we are here to help. But please consult this system manual first. If you still can t find the answer, gather all the information or questions that apply to your problem and with the product close to you, call your dealer. Matrix dealers are trained and ready to give you the support you need to get the most from your Matrix product. In fact, most problems reported are minor and can be easily solved over the phone. In addition, technical consultation is available from Matrix engineers every business day. We are always ready to give advice on application requirements or specific information on installation and operation of our products. The system manual is divided in following sections: Section 1: Introduction Section 2: Features and Facilities Section 3: Appendices We suggest the first time users to read this system manual in the following sequence and then remaining chapters: Section 1 Section 2 (In hierarchy given below) Getting Started 18 Jeeves 44 Resetting the ATA 59 Peer-to-Peer Calling 55 Supplementary Services 70 Status 66 Time Settings 82 FXS Port 34 SETU ATA2S System Manual 7
8 WAN Port 93 VoIP Basics 83 SIP Accounts 60 Dial Plan 32 Phone Book 58 Software Upgrade 65 Auto Configuration 29 System Name 81 VoIP ATA Phone Adaptor and SETU ATA2S are used as synonymous in this manual. 8 SETU ATA2S System Manual
9 Packing List The ideal sales package for SETU ATA2S is as mentioned below: Sr. Accessories Qty. 01 SETU ATA2S 1 02 System Manual 1 03 Adaptor 12V, 1.25Amp Mounting Screw 30/ Screw Grip 2 06 Warranty Card set 1 07 Support Card 1 08 RJ11 Cable 2 09 RJ45 Cable 1 10 Mounting Template 1 11 SETU ATA2S CD 1 Please make sure that these components are present. In case of short supply or damage detection, contact the source from where you have purchased the system. =X=X= SETU ATA2S System Manual 9
10 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. Any replacement product or part/s may be furnished on an exchange basis, which shall be new or like-new, provided that it has functionality at least equal to that of the product, being replaced. All replacement parts and products will be the property of Matrix. Parts repaired or replaced will be under warranty throughout the remainder of the original warranty period only. This limited warranty does not apply to: 1. Products that have been subjected to abuse, accident, natural disaster, misuse, modification, tampering, faulty installation, lack of reasonable care, repair or service in any way that is not contemplated in the documentation for the product, or if the model or serial number has been altered, tampered with, defaced or removed. 2. Products which have been damaged by lightning storms, water or power surges or which have been neglected, altered, used for a purpose other than the one for which they were manufactured, repaired by customer or any party without Matrix s written authorization or used in any manner inconsistent with Matrix s instructions. 3. Products received improperly packed or physically damaged. 4. Products damaged due to operation of product outside the products specifications or use without designated protections. Warranty valid only if: Primary protection on all the ports provided. Mains supply is within limit and protected. Environment conditions are maintained as per the product specifications. 10 SETU ATA2S System Manual
11 Warranty Card: When the product is installed, please return the warranty card with: Date, signature and stamp of the customer. Date, signature and stamp of the channel partner. Matrix assumes that the customer agrees with the warranty terms even when the warranty card is not signed and returned as suggested. The Purchaser shall have to bear shipping charges for sending product to Matrix for testing/rectification. The product shall be shipped to the Purchaser at no-charge if the material is found to be under warranty. The Purchaser shall have to either insure the product or assume liability for loss or damage during transit. Matrix reserves the right to waive off or make any changes in its warranty policy without giving any notice. If Matrix is unable to repair or replace, as applicable, a defective product which is covered by Matrix warranty, Matrix shall, within a reasonable time after being notified of the defect, refund the purchase price of the product provided the consumer/purchaser returns the product to Matrix. In no event will Matrix be liable for any damages including lost profits, lost business, lost savings, downtime or delay, labor, repair or material cost, injury to person, property or other incidental or consequential damages arising 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. Neither Matrix nor any of its distributors, dealers or sub-dealers makes any other warranty of any kind, whether expressed or implied, with respect to Matrix products. Matrix and its distributors, dealers or SETU ATA2S System Manual 11
12 sub-dealers specifically disclaim the implied warranties of merchantability and fitness for a particular purpose. This warranty is not transferable and applies only to the original consumer purchaser of the Product. Warranty shall be void if the warranty card is not completed and registered with Matrix within 30 days of installation. 12 SETU ATA2S System Manual
13 Protecting the System Matrix Installation Precautions: Do not installed In direct sunlight and excessive cold or humid places. Do not installed at places at where sulfuric gases produced in areas where there are thermal springs, etc. because it may damage the equipment or contacts. Do not installed at places in which shocks or vibrations are frequent or strong. Do not installed at dusty places or places where water or oil may come into contact with the system. Do not obstruct area around the system (for reasons of maintenance and inspection be especially careful to allow space for cooling above and at the sides of the system). Important Safety Instructions: When using your telephone equipment, basic safety precautions should always be followed to reduce the risk of fire, electric shock and injury to persons, including the following: Read and understand all instructions. Unplug this product from the wall outlet before cleaning. Do not use liquid cleaners or aerosol cleaners. Use a dry and soft cloth for cleaning. Do not use this product near water, for example, near a bathtub, wash bowl, kitchen sink, or laundry tub, in a wet basement, or near a swimming pool. Do not open the system in power ON condition. Do not place this product on an unstable cart, stand or table. The product may fall, causing serious damage to the product. WAN and Lifeline interfacing cables should not touch the exposed power line cable. This product should be operated with proper supply voltage. If you are not sure about supply voltage, contact authorized dealer. It is advisable to give proper, stabilized power. Do not allow anything to rest on the power cord of product or AC- DC Adapter. Do not place this product where the cord will be misused by people walking on it. SETU ATA2S System Manual 13
14 Do not overload wall outlets and extension cords as this can result in the risk of fire or electric shock. To reduce the risk of electric shock, do not disassemble this product. Take it to a qualified serviceman when some service or repair work is required. Opening or removing covers may expose you to dangerous voltages or other risks. Incorrect reassembly can cause electric shock when the appliance is subsequently used. Unplug this product from the wall outlet and refer servicing to qualified service personnel under the following conditions: a) When the power supply cord or plug is damaged or frayed. b) If liquid has been spilled into the product. c) If the product has been exposed to rain or water. d) If the product does not operate normally by following the operating instructions. Adjust only those controls, which are covered by the operating instructions because improper adjustment of other controls may result in damage and will often require extensive work by a qualified technician to restore the product to normal operation. e) If the product has been dropped or the cabinet has been damaged. f) If the product exhibits a distinct change in performance. Do not use the telephone of the product to report a gas leak in the vicinity of the leak. =X=X= 14 SETU ATA2S System Manual
15 Introducing the System Matrix offers the Voice over IP-Analog Telephone Adaptor, VoIP ATA or Matrix Phone Adaptor (SETU ATA2S) which is a SIP based VoIP product. Sending voice signals over the Internet is called Voice over IP or VoIP. Session Initiated Protocol, SIP is an internationally recognized standard for implementing VoIP. VoIP-ATA is an Analog Telephone Adapter that allows a user to make voice call over IP network using a conventional telephone instrument. VoIP-ATA has one IP port, for WAN connection. WAN port is used to connect VoIP-ATA to the Internet using broadband modem, cable modem, router, etc. The VoIP calls are made through WAN port. VoIP-ATA has two FXS ports to which conventional telephone instruments (SLTs) can be connected. Call routing mechanisms for IC calls from IP to FXS and OG calls from FXS to IP are included in VoIP-ATA. The IC call routing mechanism helps the user to route the calls to either or both the FXS ports whereas the OG call routing mechanism helps the user to route the call through desired SIP Account which is helpful when services from two different ITSPs have been taken, each having cost benefit in some terms over the other. When a telephone number (A E.164 number or a SIP number without domain name. i.e. the prefix of the SIP telephone number e.g. Suppose a user with SIP number SIP: [email protected] is to be called then the caller should dial 12345) is dialed by the user, the VoIP-ATA converts this into an IP call using SIP protocol and makes a call to the callee. Using Phone book, the user can also dial complete SIP number by just dialing feature code and corresponding index number. User can also use Auto Configuration feature if he wants to avail facility of configuration of ATA by ITSP from central location. Refer Feature List which includes important features supported by VoIP-ATA. SETU ATA2S System Manual 15
16 Feature List: Configuration of the ATA through Web-pages Auto Configuration Resetting the ATA User Access with Password Synchronization with Time Server Supplementary Services with T.38 Fax capability WAN Port IP Address programming Use of two SIP accounts, simultaneously Routing of Dialed Number as per Dial Plan Speed dialing using Phone Book feature Peer-to-Peer Calling Status page display of main parameters By default, Static IP Address of ATA2S is : Before proceeding further, please ensure that you have an Internet access and a SIP Account already set up. SETU ATA2S Photograph: 16 SETU ATA2S System Manual
17 Relevant Topics: 1. Glossary VoIP Basics 83 =X=X= SETU ATA2S System Manual 17
18 Getting Started Overview The Matrix ATA allows you to use the analog telephone to make phone calls over the Internet. The ATA connects to the Internet through a DSL Modem or Router as shown below: FXS2 FXS1 WAN Internet Network This chapter gets the ATA up and running quickly. The details which we have skipped to make this brief can be found elsewhere in the manual. It is divided into four sections: 1. Getting to know the phone adapter. 2. Instruction for connecting the phone adapter. 3. Basic steps for configuration. 4. Making phone calls. Getting to know the phone adapter The Matrix ATA s ports are located on side panel and LEDs are on front panel. Both are explained below: Connection of Right Side panel ports. Left Side panel LEDs. 18 SETU ATA2S System Manual
19 Right Side Panel Ports Following picture shows how the ports are connected to various interfaces: Matrix Telephone Cable Power Adaptor Ethernet Cable FXS2 FXS1 12VDC 1A(Max) ETHERNET (2) (1) (3) (4) Ethernet Switch No. Port Name Connector Description 1 FXS 1 RJ 11 To connect analog telephone/fax. 2 FXS 2 RJ 11 To connect analog telephone/fax. 3 Power DC Jack 4 Ethernet RJ 45 with two LED s Power Adaptor, 12Volt 1.25Amp. DC. This is a WAN Port. Connection to cable modem/ ADSL modem or router or LAN switch. Ethernet activity LED (Green) is glowing. PC can be connected from Ethernet switch to access the ATA. SETU ATA2S System Manual 19
20 Left Panel LEDs The ATA has 5 LEDs; First for PWR, Second for FXS 1, Third for FXS2, Fourth for Registration Status of SIP1 and Fifth for SIP2. These LEDs are used to signify various events. LED Label LED Indication PWR Green FXS1 Red FXS2 Red SIP1 Red SIP2 Red The reset sequence: At power ON, PWR LED is glowing continuously, whereas all other LEDs glow for few seconds time which depends on the timing cycle from boot loader program to init the sequence. On successful completion of initialization cycle, each LED is glowing as per the normal conditions. Normal conditions: The LED indications should be as per following table: LED Status Meaning ON IC Call/OG Call-Speech and Off- Hook. Error condition during IC/OG Fast Blinking FXS1 call/busy Condition. Slow Blinking IC Call-Ring Event/OG Call-Number being dialed. OFF FXS port On-hook. ON IC Call/OG Call-Speech and Off- Hook. Error condition during IC/OG Fast Blinking FXS2 call/busy Condition. Slow Blinking IC Call-Ring Event/OG Call-Number being dialed. OFF FXS port On-hook. 20 SETU ATA2S System Manual
21 SIP1 SIP2 ON SIP1 Active and ATA registered with the SIP server. Fast Blinking SIP1 Active but ATA not registered with the SIP server. Slow Blinking SIP1 Inactive. ON SIP2 Active and ATA registered with the SIP server. Fast Blinking SIP2 Active but ATA not registered with the SIP server. Slow Blinking SIP2 Inactive. Where, OFF ON Slow Blinking Fast Blinking IC, OG : Continuously OFF : Continuously ON : 500ms ON, 500ms OFF : 200ms ON, 200ms OFF : Incoming, Outgoing Instruction for connecting the phone adapter This section describes the instructions on, how to connect the Matrix ATA to internet network and telephones. Unpack the box. Get satisfied with the contents and the condition of the parts. Refer to Packing List. If parts are OK, proceed with connections as shown below: Mount your ATA in a safe and convenient location where cables for your network and phone system are accessible. For this, mechanical drawing can be used as a reference, which is provided end of the chapter. Connect a standard analog telephone instrument to FXS1 and FXS2 ports, using telephone cables provided. Insert one end of the Ethernet cable into the WAN port of ATA and connect the other end of the Ethernet cable to Ethernet Switch which is connected to router or DSL Modem. Insert one end of the Ethernet cable into the Ethernet Switch and connect the other end of the Ethernet cable to the PC. Check the voltage from the power point from where the supply is to be accessed. It should be between VAC, 47-63Hz. SETU ATA2S System Manual 21
22 Insert the DC output terminal of power adapter into the Power jack of the ATA and connect the 230VAC pins of the power adapter to a wall outlet for 230VAC. When you power ON the ATA, observe that all LED s are ON for few seconds and then SIP1, SIP2 LED s are flashing. Basic Steps for Configuration WAN Port: Default WAN IP Address for ATA is ABSOLUTELY NEEDED! Use WAN IP Information from your LAN Admin. or ISP IP Address (must) IP Mask (must) Gateway DNS IP Address (must) DNS Domain Name Note that subnet of WAN and Internet (Router) are same. Read the IP of PC, by clicking on Network Neighborhood Properties TCP/IP Connection Properties for Windows98. Choose the unique IP address of the PC such that its subnet is same as subnet of IP address of ATA, provided by ISP. PLEASE CHECK! The subnet of your PC IP Address must be same as of ATA. For example, if IP address of ATA is , then PC IP Address port should be programmed as X. where X can be any number from 1 to 254, except 10. By doing this, subnet remains same. Pick up the handset of the analog phone connected to ATA. Enter the programming access code #19 followed by password You get programming tone. Enter the WAN IP Address, using command 11-<WAN IP Address>-#*. For e.g. to enter IP Address Enter command 11-< >-#*. You get confirmation tone followed by system restarting. 22 SETU ATA2S System Manual
23 Start Internet Explorer (IE6 with SP2, Service Pack) of PC connected to the Ethernet Switch. Type in the IP Address programmed above in the Address field. On the login page, enter default password Click on SIP Account from the menu provided on left side of the web page. Get Information from your ITSP for SIP Account SIP ID Registrar Server Address Registrar Server Port Authentication User ID Authentication Password Outbound Proxy Server Address Outbound Proxy Server Port Enter above information in SIP Account1 field. If these information is not available, keep the values as default. For e.g. Registrar Server Port = Enable SIP Account for SIP1 and also enable Outbound Proxy, if information is programmed. If you have a second internet phone Service Account (SIP2), then use above fields for SIP2 port and enable. It allows you to use second phone (or Fax) connected at FXS2 port. Submit page. You can have a look at all above parameters and software version information by clicking on Status from the menu. ATA is ready for further IP calling service! User can also use Auto Configuration feature if he wants to avail facility of configuration of ATA by ITSP from central location. Refer Appendix A: Troubleshooting or default the system and start again, if you cannot establish connectivity from your ATA. Making Phone Calls After SIP1 and/or SIP2 LED is ON, just dial the SIP ID numerical number e.g which is registered to the same ITSP provider. If SIP account is [email protected] then the SIP ID number is 4567 and SIP service domain is Provider.com. SETU ATA2S System Manual 23
24 PSTN number can be called directly, to the area which your ITSP Service Provider supports. You can also use feature of Speed Dialing and Peer-to-Peer Calling as explained below: Speed Dialing: The numbers which you need to dial frequently can be programmed in the Phone Book table. Suppose the number you want to dial is entered at index 05 of the phone book, then go OFF- Hook and dial #8-05. The number will be dialed! Peer-to-Peer Calls: Peer-to-Peer number can be dialed even if ATA is not registered to any service provider. If Peer-to-Peer is enabled and FXS port number is programmed, you can call another ATA by dialing its FXS port number. You can also call another ATA or IP- Phone by dialing its IP Address if Peer-to-Peer is enabled. For e.g. Just dial 192*168*1*21, to call the IP number Mechanical dimension of SETU ATA2S: 24 SETU ATA2S System Manual
25 SIMADO ATA2S mm(3.15 Inch) mm (4.09 Inch) mm (1.06 Inch) =X=X= SETU ATA2S System Manual 25
26
27 Section 2: Features and Facilities
28
29 Auto Configuration Matrix ATA configuration includes some basic settings like Registrar Server Address, Authentication user ID, and User Password which are provided by the Service Provider (ITSP). Also some special services like Call Forward are provided by the ITSP. Hence, it is desirable that the ATA is configurable by the ITSP from the central location. Matrix ATA supports this by providing Auto Configuration feature. The ITSP will keep the folders of configuration files in Central Server for different ATA units as per LAN/WAN MAC Address number of the ATA. When the customer connects the ATA to the ITSP s network, the ATA gets automatically configured and the customer starts using the services. How it Works? ATA makes use of TFTP server to down load the configuration files. User has to just enter the Server Address provided by ITSP and specify the config. Folder path provided by ITSP, in the Web page for Auto Configuration. Then after selecting option for using the feature at power ON, Click on submit the page. Three options for using the feature are provided: User can select that the ATA is not to be configured provided as default. User can select that the ATA is to be configured, whenever it is switched ON. User can select that the ATA is to be configured, when switched ON next time. ATA can get the Address of FTP Server from the DHCP Server also if FTP Server Address is not programmed in ATA. The MAC Address displayed on the Status Page is programmed at the factory and can not be changed or deleted by the user. If the FTP server address is not found, the auto-configuration process is aborted and the default configuration files or the current ones are used by the ATA. If all the required files are not available in the folder, then there will be incomplete configuration download to the ATA. Also only those downloaded files whose version will match to the version of ATA will SETU ATA2S System Manual 29
30 be updated. For example, 7 configuration files are necessary and if only 5 files are present, then these 5 files only will be downloaded by the ATA and will be updated in ATA which may not meet the required Configuration of the ATA. When Auto Configuration is complete, the Status page displays following information: In Status field suitable message is displayed such as: No Tftp Server Specified Auto Configuration Fail Auto Configuration Success In Statistics field, suitable message regarding how many files are down loaded successfully and total number of configuration files are provided by the system. For example, if X=Number of files down loaded and Y=Total number of files updated. Then the message displayed will be as: Files Downloaded/Configured = X/Y How to program? Click on Auto Configuration from the menu on the Welcome Page. Auto Configuration: Parameter Value/Default Description At Next Power ON Blank At Power ON Blank Select this option if you want to Auto Configure your ATA when you Power it ON next time. Select this option if you want to Auto Configure your ATA whenever you Power it ON. Never Default = Selected. Select this option if you do not want to Auto Configure your ATA. TFTP Server N.N.N.N, Enter here the server address Address 40 characters Max. N = 001 To 255 Or Domain Name Default = Blank provided by ITSP. Do not enter if it is to be allocated by DHCP server, on request by ATA. Config Folder Path 40 Characters max. Default = Blank Provided by ITSP. Enter here the path of the folder in which the configuration files are stored. For e.g. Matrix/SetuATA. 30 SETU ATA2S System Manual
31 Important Points: ATA will not restart, if user clicks on Submit on this page. This feature can also be used when ATA is connected to an IP PBX. Relevant Topic: 1. Status 66 =X=X= Matrix SETU ATA2S System Manual 31
32 Dial Plan Generally Matrix ATA user can assign two SIP Account numbers to two FXS Ports, one for each. Hence if Call from FXS1 needs to use SIP2 Account, then it can be done by assigning SIP2 to FXS1 in the FXS Port parameters screen. But the ATA has also facility to allocate SIP account, as per the number dialed. The database for this mapping is called Dial Plan. Dial Plan has in all 10 entries. Each entry has a number and corresponding SIP account which the system will use to make an OG call. If the number string dialed by the user matches with any of the number string in the Dial Plan, the call is made using the corresponding SIP Account. This flexibility is helpful in following case: Example: If Numbers are mapped with SIP Account as given below: Index Number SIP Account SIP SIP2 : : : 10 Then, a person in the Cyber cafe can call to Africa on 1267 even if another person is talking to US on SIP1 Account on 1234 number. The ATA allocates SIP2 account to the person who wants to call Africa, based on closest best fit criteria. Thus Dial Plan works as powerful routing tool. Similarly, if the user dials a number say , the call should be made using SIP2 as this number best fits with the number string in index 02. If entries are not matching or the table is blank, the OG call will use default SIP1 Account. Click on Dial Plan. 32 SETU ATA2S System Manual
33 Dial Plan: Index 1 to 10 Number SIP Account Value/Default 16 digit Maximum is from 0 to 9. Default = Blank SIP 1 and SIP 2 Default = SIP1 Relevant Topics: 1. FXS Port SIP Accounts 60 Description This is the index number at which entry is done. Enter the number string which the user needs to dial frequently. Enter the SIP Account Number provided by ITSP. =X=X= Matrix SETU ATA2S System Manual 33
34 FXS Port Introduction: For efficient interfacing with Analog telephone connected at FXS1 and FXS2 and for efficient use of many features like Echo cancellation, VAD, etc., some parameters are programmed on FXS port. This chapter gives detailed procedure of programming. The ATA supports only DTMF dialing (Tone Dialing). ATA supports four types of Caller Number Presentation viz. None, DTMF (not in this version), BellCore FSK and V.23 FSK. ATA supports Calling Party Control to disconnect the Trunk which is still busy when remote called party has gone ON-Hook. It is implemented by: CPC period of 500 Millisecond. ATA supports programming at Input Gain (-6dB to +6dB), Output Gain (-6dB to +6dB), Programmable Flash Timer, Forward Error Correction, Programmable Ring Timer, Programmable Inter-Digit Timer and fixed First Digit Wait Timer. Some parameters are described as below before explaining programming: FXS Port Name: This name is sent to the callee in the Display Name field in the SIP call. The callee shall be able to identify the FXS port from which the call is made along with the SIP number provided by the ITSP for the SIP Account. To elaborate: Suppose the ITSP has provided a SIP number [email protected]. When the call is made by FXS1 or FXS2, the callee gets the same number. Hence he is unable to identify that who is the actual caller; FXS1 or FXS2. The Station name feature resolves this problem by assigning xyz name only to FXS1 and calling from port FXS1. FXS Port Number: It is used for Peer-to-Peer calls only. Call Progress tone: Different types of tones are played during specific event of establishing the call, speech duration and termination of the call. As per ITU standard these tones are of different specification for different country. For example, Dial Tone for India consists of 34 SETU ATA2S System Manual
35 400Hz modulated by 25Hz, but for USA/CANADA, it is Hz, without modulation. Based on specific frequency, modulating frequency and cadence, tones are categorized as shown below: Dial Tone Ring Back Tone Busy Tone Error Tone Confirmation Tone Programming Tone CO Call Waiting Tone (CCWT) ATA plays the Dial Tone when it is ready to accept the first digit of a remote address to make an outgoing call. Normally played when FXS goes OFF-Hook. ATA plays the Ring Back Tone when the called device is alerting the callee. ATA plays the Busy Tone when the callee is busy. ATA plays Error Tone when, there is no registration with proxy server or user has performed some wrong operation. It remains for 7 secs. But when it is used as Error Tone due to invalid programming, then it remains for only 3 secs. and it is called Programming Error Tone. ATA plays Confirmation Tone, to indicate that valid command is received by the system and it has taken necessary action. ATA plays Programming Tone which, prompts the user to enter fresh commands during programming. This tone, remains continuously till user dials digit 0. But ATA plays this Programming Tone as Feature Tone when user has activated some feature like Call Forward, Do Not Disturb (DND), etc. and it remains for 7 secs. ATA plays the CCWT when an incoming call arrives while the user is connected to another party. The time duration and the tone, when it is matured are as per following table 1: SETU ATA2S System Manual 35
36 Name Timer (Sec.) Matures to Dial Tone 7 Error Tone Standby RBT 60 Error Tone Standby Busy Tone 30 Error Tone Standby Error Tone 7 Error Tone Standby Confirmation Tone 3 Dial Tone/Feature Tone Programming Tone 7 Error Tone Standby CCWT 60 Error Tone Standby Timer column shows that for how long the tone will continue and last column Matured to shows type of tone after timer is expired. It goes in stand by after error tone completes. For example, ATA plays error tone for 7 secs. and then goes in stand by. Refer Table 2 for tone specification for different countries, at the end of this chapter. Just select country name from the list of countries in the Combo Box for FXS1 and FXS2 to program the Call Progress Tone. Ring Type: The Matrix ATA supports different cadence for the ring when incoming call lands on FXS1 or FXS2. These cadence is standardized for different country. For example, India supports: 25Hz, 0.4 ON 0.2 OFF, 0.4 ON 2.0 OFF (Sec.). USA/Canada supports: 25Hz, 2.0 ON 4.0 OFF (Sec.). In order to support a specific ring type, user has to just select the country. The system will select the cadence as per table 3, at the end of chapter. Default country is India. First Digit Wait Timer: First digit wait timer signifies the time for which the system waits for the user to dial something once the FXS port goes OFF-Hook. On expiry of this timer, the system will give error tone to the user. 36 SETU ATA2S System Manual
37 Internal Digit-Digit Wait Timer: (Timer of Relevance) This is the time during which if user does not dial a digit after pressing a digit, the system stops collecting further digits and processes digits dialed so far. Matrix Echo Cancellation: Line echo is caused when conversion between 4-wire to 2-wire line is carried out, by telephony hybrid. And delay exceeds 50ms. Acoustic echo is caused when our own voice reverberating the phone-receiver when we talk. Echo Cancellation is a technique that allows for isolation and filtering of unwanted signals caused by echoes. The echo canceller-device puts a signal on the return transmission path, which is equal and opposite of the echo signal. Echo cancellers (digital filters) are used depending on the length of the echo to be compensated, also depends on the distance between gateway and line hybrid. (Between 8-32ms for SOHO products) VAD: Voice Activation Detection (VAD), is a software application that allows a 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. Most conversations include about 50% silence. VAD (also called silence suppression ) can be enabled to monitor signals for voice activity so that when silence is detected for a specified amount of time, the application informs the Packet Voice Protocol and prevents the encoder output from being transported across the network. Click on FXS FXS Ports SETU ATA2S System Manual 37
38 Parameter FXS Port Name FXS Port Number Listening Volume Speaking Volume Values/Default 16 char. Max. Default = Blank 4 digits max. (0,9,*) Default = Blank Voice Volume Control Tones and Rings Call Progress Tones Ring Type Ring Timer -6 to +6 db, in 1- db Steps Default = 0-dB. -6 to +6 db, in 1- db Steps Default = 0-dB. List of countries refer table 2. Default=India List of countries refer table 3. Default = India seconds Default = 60 Description (Enter for FXS1 and FXS2) Enter if name is to be sent with the dialed number. Enter here, the extension number on which, incoming Peer-to-Peer call, when dialed on this number will land. Both phones at FXS1 and FXS2 will ring is this number if not matching. Select to modify audio level being output to the device attached to the FXS. Select to modify audio level entering voice before it is sent over the network to the remote VoIP. Select the country where the ATA is installed. Refer table 2 for the frequency and cadence of various tones supported. Select the country where ATA is installed. Refer table-3 for the cadence of various rings supported Select the time period to play the ring. 38 SETU ATA2S System Manual
39 Calling Party number Presentation Enable Type Caller ID Send Caller ID Polarity Reversal Enable Call Party Control Enable CPC Flash Timer Default = Yes FSK-BellCore Default = FSK- ITU (V.23) Default = Yes Default = No Default = No Matrix Click here, to enable the feature. Select the CLI Type as per the Phones connected at FXS1 and FXS2. Click if CID is to be sent during an OG call. If NO, CID will be Blocked. When called party goes OFF- Hook caller gets 'Polarity Reversal'. Click if CPC feature required. When called party goes ON- Hook, CPC is detected for 500msec. Flash Timer 100 to 900 Millisecond in steps of 100 ms. Default = 600 ms. Select as per the Flash period standards of the terminal or phone. Echo Cancellation Enable Default = Yes Click if the feature required Tail Length 8 msec 16 msec 32 msec Default = 32 msec. Select such that delay is minimum so that Voice quality is improved. SETU ATA2S System Manual 39
40 Forward Error Correction Enable Default = No FEC feature is to reduce error due to the Lost Packet. Click for improved voice quality. Voice Activity Detection Enable Default = No Click if VAD feature required. Number Dialing Timer First Digit Wait Timer Inter Digit Wait Timer Call Waiting 7 sec Fixed for the system. 1-9 seconds Default = 4 Enter if time period is required to change for dialing digits. Call Waiting Default = Yes Click if this feature is required only for next OG call. Outgoing Calls Outgoing calls None SIP Account 1 SIP Account 2 Dial Plan Default = SIP1 for FXS1 and SIP2 for FXS2 Peer-to-Peer Yes/No Enable Default = No Assign SIP Account SIP1, SIP2 or the Account programmed in the Dial Plan that FXS1 and FXS2 will use for OG-Call. Select 'Yes' if direct IP Calling feature is required. Note 2: Polarity Reversal on FXS port is as per following table: 40 SETU ATA2S System Manual
41 Condition Idle Off-hook Remote Party Answers On-hook Battery Polarity Negative Toggle to Positive Toggle to Negative Retain to Negative Table 2: Dial Tone Ring Back Tone Busy Tone Error Tone Country Freq. Cadence Freq. Cadence Freq. Cadence Freq. Cadence Hz second Hz second Hz second Hz second Australia 425*25 cont. 400*25 0.4on 0.2off 0.375on on 2.0off 0.375off on 0.375off Argentina 425 cont on 4.0 off on 0.2off on 0.4off Brazil 425 cont on 4.0 off on 0.25off on 0.25 off China 450 cont on 4.0off on 0.36off on 0.7off Egypt 425*50 cont 425*50 2.0on 1.0off 425*50 1.0on 4.0off on 0.5off France 440 cont on 3.5off on 0.5off on 0.25off Germany 425 cont on 4.0off on 0.48off on 0.24off Greece on 0.3off 0.7on 0.8off on 4.0off on 0.3off on 0.15off India 400*25 cont 400*25 0.4on 0.2off 0.4on 2.0off on 0.75off on 0.25off Indonesia 425 cont on 4.0off on 0.5off on 0.25off Iran 425 cont on 4.0off on 0.5off on 0.25off Israel 400 cont on 3.0off on 0.5off on 0.25off Italy 425 cont on 4.0off on 0.5off on 0.2off Japan 400 cont 400*20 1.0on 2.0off on 0.5off on 0.25off Kenya 425 cont on 3.0off 0.2on 0.6off on 5.0off 0.2on 0.6off on 0.6off Korea cont on 2.0off on 0.5off on 0.2off Malaysia 425 cont on 0.2off 0.4on 2.0off on 0.5off on 0.25off Mexico 425 cont on 4.0off on 0.25off on 0.25off New Zealand400 cont. 0.4on 0.2off on 2.0off on 0.5off on 0.25off Phillippines 425 cont on 4.0off on 0.5off on 0.25off Poland 425 cont on 4.0off on 0.5off on 0.5off Portugal 425 cont on 5.0off on 0.5off on 1.0off Russia 425 cont on 3.2off on 0.4off on 0.25off Saudi Arabia425 cont on 4.6off on 0.5off on 0.25off Singapore 425 cont. 425*24 0.4on 0.2off 0.4on 2.0off on 0.75off on 0.25off South Africa 400*33 cont. 400*33 0.4on 0.2off 0.4on 2.0off on 0.5off on 0.25off Spain 425 cont on 3.0off on 0.2off on 0.25off Thailand 400*50 cont on 4.0off on 0.5off on 0.3off Turkey 450 cont on 4.0off on 0.5off on 0.2off 0.6on 0.2off 0.4on 0.2off 0.375on 0.4on 0.35off UAE cont on 2.0off 0.375off 0.225on 0.525off 0.4on 0.2off 0.375on 0.4on 0.35off UK cont on 2.0off 0.375off 0.225on 0.525off USA/Canada cont on 4.0off on 0.5off on 0.25off SETU ATA2S System Manual 41
42 Remarks: Frequency notations shall be as: f1*f2: f1 is modulated by f2. For All Countries Confirmation Tone Programming Tone CCWT : 400Hz, 0.1 on 0.1 off, : 400Hz, 0.1 on 0.9 off, : 400Hz, 0.3 on 6.0 off Table 3: Code Country Frequency CADENCE (In Seconds) (Hz) ON 1 OFF 1 ON 2 OFF 2 01 Australia Brazil China Egypt France Germany Greece India Israel Italy Japan Korea Malaysia New Zealand Poland Portugal Russia Singapore South Africa Spain Thailand UAE UK USA/Canada SETU ATA2S System Manual
43 Relevant Topics: 1. Peer-to-Peer Calling SIP Accounts Supplementary Services Dial Plan 32 =X=X= SETU ATA2S System Manual 43
44 Jeeves What s this? As, the set up of SOHO and Enterprise is normally equipped with WAN and Router on the Internet line, it is very convenient if the device has the capability to be programmed via HTML-based interface. The Matrix ATA has this provision. WWW is an abbreviation of Internet s World Wide Web. Matrix ATA needs Web browser IE6 with SP2 (Service Pack) patch or Mozilla Fire Fox installed on the PC. Description of following features related to web will help in programming the system. IP Address An IP address (Internet Protocol address) is a unique number, similar in concept to a telephone number, used by network devices (computers, time-servers, FAX machines, some telephones) attached to a network to refer to each other when sending information through a WAN or the Internet for example. This allows devices passing the information onwards on behalf of the sender to know where to send it next, and for the device receiving the information to know that it is the intended destination. An example of IP address is Converting a number address to a more human-readable form called a domain address is done via the Domain Name System. The Internet Protocol (IP) knows each logical host interface by a number, the so-called IP address. On any given network, this number must be unique among all the host interfaces that communicate through this network. Users of the Internet are sometimes given a host name in addition to their numerical IP address by their Internet service provider. For all programs that utilize the TCP/IP protocol, the sender IP address and destination IP address are required in order to establish communication and send data. Sub Net It is a mechanism that is used to split a network into a number of smaller sub networks. It can be used to reduce traffic on each sub network by confining traffic to only the sub networks for which it is 44 SETU ATA2S System Manual
45 intended, thereby eliminating issues of associated congestion on other sub networks and reducing congestion in the network as a whole. Makes entire network more manageable. Each sub network functions as though it was an independent network, keeping local traffic local, and forwarding traffic to another sub network only if the address of the data is external to the sub network. Such decisions are made on the basis of routing-tables contained within the various routers, with each table comprising an IP address table. Subnet is a portion of the network, which may be a physically independent network, which shares a network address with other portions of the network and is distinguished by a subnet number. Default Gateway A default gateway is a node on a computer network that serves as an access point to another network. In enterprises, the gateway is the computer that routes the traffic from a workstation to the outside network that is serving the Web pages. In homes, the gateway is the ITSP that connects the user to the internet. A default gateway is used by a host when an IP packet s destination address belongs to someplace outside the local subnet. Domain Name System (DNS) DNS is a system that stores information about host names and domain names in a type of distributed database on networks, such as the Internet. Of the many types of information that can be stored, most importantly it provides a physical location (IP address) for each domain name, and lists the mail exchange servers accepting for each domain. The DNS provides a vital service on the Internet as it allows the transmission of technical information in a user friendly way. While computers and network hardware work with IP addresses to perform tasks such as addressing and routing, humans generally find it easier to work with hostnames and domain names (such as in URLs and addresses. SETU ATA2S System Manual 45
46 The DNS therefore mediates between the needs and preferences of humans and of software. Dynamic Host Configuration Protocol (DHCP) The DHCP is a client-server networking protocol. A DHCP server provides configuration parameters specific to the DHCP client host requesting, generally, information required by the host to participate on the Internet network. DHCP also provides a mechanism for allocation of IP addresses to hosts. The DHCP protocol provides three methods of IP address allocation: Manual Allocation. Automatic Allocation. Dynamic Allocation. How to configure ATA? To start basic VoIP configuration first enter the default WAN IP address of the ATA, obtained from SE in the Address bar of the Webbrowser screen as explained below: Launch the Web browser as explained below: Double Click on Internet Explorer icon Or Click on Start Programs Internet Explorer. Enter IP Address at Address Bar after confirming from SE. Let us say IP Address is entered at Address Bar on opening page of Microsoft Internet Explorer. Note that Matrix ATA2S default IP Address is ATA displays Welcome page with a field for entering Password. Field Name Value/Default Description Password Up to '4' digits (0-9), 1234 Enter the default password 1234 to access the system Click Login after entering valid Password. If there is any error in typing the Password, the Message page is displayed for entering Password again and click on Submit. After entering the correct password, login status page is displayed. This page contains following table (information) at the centre of the page. 46 SETU ATA2S System Manual
47 MATRIX SETU ATA2S Welcome User Address Login expires if idle for xxx.xxx.xxx.xxx 10 Minutes xxx.xxx.xxx.xxx : Address of the computer from where the user has logged in. Login Expiry Time: This is fixed (maximum) time for which user can keep a page without programming any parameter in it. If this time is exceeded, he cannot submit current page. He has to login again. All the above fields are non-editable. The user can now go to any of the screens by clicking on the links provided at the left of the screen. Main Menu is displayed on the left side of Welcome page (without Password field). The Main Menu List shows all the important parameters that are required to be programmed, to make the ATA work as expected. Click from the Main Menu. Common screen command buttons: (General Note for all the screens) Note that at the end of the every screen, options are provided for really applying these changes to the ATA. This is conveyed by Submit and Default All : Button Meaning Submit Click, to save changes to ATA Click, to set all parameter on the Default All screen to factory default The Main Menu List will be displayed as shown below: Password System Name Time Settings FXS Ports Supplementary Services WAN Port SETU ATA2S System Manual 47
48 SIP Accounts Dial Plan Phone Book Peer-to-Peer Calling Auto Configuration Software Upgrade Status Default the ATA (Resetting the ATA) Screens Summary: LINK FUNCTIONS PASSWORD Use this screen to change password. System Name Use this screen to assign name to ATA. Time Setting Use this screen to configure system s time to synchronise with the time server. FXS Port Use this screen to configure FXS-Port parameters like gain, echo cancellation etc. Supplementary services Use this screen to configure call features like call forward, DND etc. WAN Port Use this screen to configure WAN Port Parameters. SIP Account Use this screen to configure SIP settings, coder, other parameters for SIP1 and SIP2 Ports. Dial Plan Phone book Auto Configuration Software Upgrade Status Use this screen for entering number strings to select SIP Account. This is for programming phone book entries. This is to configure the ATA of specific MAC Address number by ITSP. This is to upgrade any file of the software using FTP feature. This is a brief status of important parameters like WAN address. Peer-to-Peer This is for programming Peer-to-Peer Call entries. Calling Default the ATA This is for downloading default parameters. 48 SETU ATA2S System Manual
49 In order to make required changes in features, click on any concerned feature from the list given on left side of the Main Menu. Example 1: To use Dial Plan feature for outgoing call. 1. After entering suitable IP address in address field the welcome page look, as below: 2. After Login, the welcome page displays the time after which login expires as shown below: SETU ATA2S System Manual 49
50 3. Click on SIP Accounts. Enter SIP ID Numbers and registrar Server Address provided by ITSP and enable SIP accounts as shown below: 50 SETU ATA2S System Manual
51 4. Click on FXS Port and select Dial Plan for FXS1 and select SIP Account2 for FXS2. Thus any number dialed from phone at FXS1 will use the SIP number mentioned in the Dial Plan screen or SIP1 as default. But any number dialed from FXS2 will use SIP Accounts2 only as shown below: 5. Click on Dial Plan. Enter the required number string that will be dialed from FXS1 and use the specific SIP account as shown below: SETU ATA2S System Manual 51
52 Example 2: 1. Click on SIP Accounts and select Route to BOTH for incoming calls field, as shown below. If any IC call comes on SIP1, both phones will ring. 52 SETU ATA2S System Manual
53 Example 3: To program Call Progress Tones and Ring Type for country, click on FXS port and then select the name of country from the list of the combo box of Call Progress Tones and Ring Type, as shown below: Relevant Topics: 1. Password Software Upgrade 65 =X=X= SETU ATA2S System Manual 53
54 Password For system security, it is advisable to change the default System Password after configuring the parameters. Matrix ATA supports this feature. For this, old Password, New Password and Retyping of the new-password are required to be entered. The Password will not be visible. Only * will be seen for every character of the password. The fields are as shown in the Web Page: Click on Password Value/Default Description Password Type the default password or the 4 digits existing password you use to Max/1234 access the system, New Password 4 digits Max Type the new password Retype to confirm 4 digits Max Type the new password again By default, Password is Blank. Only numerals (0 to 9) are allowed up to four digits maximum. This is because same password should be used to enter SE mode from SLT. Relevant Topic: 1. Jeeves 44 =X=X= 54 SETU ATA2S System Manual
55 Peer-to-Peer Calling Matrix What s this? ATA can call another user agent (ATA or Softphone), which is in the same network (figure 2) or which can be in another network, which is connected by virtual LAN (VLAN) provided by ITSP (Figure 1). For example, ITSP provides VLAN services to connect two offices situated in different cities. ATA can call to an extension number of some other ATA or soft phone without going through any proxy. This is known as Peerto-Peer Calling. For this, the extension number (FXS Port number), which is to be called, is programmed in the number field and IP address of the called ATA is programmed in IP Address field of the Peer-to-Peer Call Table. Each entry has two parameter viz. Number and IP Address. For example, if IP address of ATA which is to be called, is and extension number 3005, then at Index 01, 3005 is programmed in Number field and is entered in IP Address field and screen is Submitted. For Peer-to-Peer OG Call, user has to follow following steps: Enable Peer-to-Peer in FXS Port screen. Dial For incoming Peer-to-Peer call (IC Call), the FXS port number on which call is to be received, is programmed in FXS Port number field on the screen FXS Port. For example, if 4001 is programmed in FXS Port screen in FXS1 column then any IC call on 4001 will land on FXS1 port. But, if incoming call doesn t have perfect match and only IP Address of Matrix ATA is called, then call will land on both FXS1 and FXS2. ATA also supports to call the ATA by an IP number. After lifting the handset, user dials digits with * in place of.. This is called Direct IP Calling feature. This feature is very much useful in case, Party s IP Address only is known and not SIP-ID. For example, a PC phone, with its IP Address , as shown in figure 1 can be called by just dialing 192*167*1*21, after lifting the handset. SETU ATA2S System Manual 55
56 Figure 1: Mumbai Office Delhi Office SETU ATA2S SETU ATA2S PC Phone Router Internet Router PC Phone SETU ATA2S VLAN Services PC Phone SETU ATA2S PC Phone Figure 2: First Floor Second Floor SETU ATA2S SETU ATA2S PC Phone Router Router PC Phone SETU ATA2S PC Phone PC Phone SETU ATA2S SETU ATA2S System Manual
57 Example of Peer-to-Peer table programming: Index Number IP Address : : : As shown in the programming table, number 2001 is dialed for ATA bearing IP Address as , to make Peer-to-Peer Call. Click on Peer-to-Peer, on the welcome page. Peer-to-Peer: Parameter Value/Default Description Index 01 to 64 Number at which entry is made. Number Maximum 4 digits (0-9,*) Default = Blank Enter the extension number of the user agent to be called. IP Address Relevant Topic: 1. FXS Port 34 Maximum 15 digits (0-9, '.') Default = Blank Enter IP Address of the user agent to be called. =X=X= SETU ATA2S System Manual 57
58 Phone Book It is the Database for storing Full URI SIP number, name and SIP account number. Maximum 99 numbers can be stored in a phone book. Each entry has three parameters viz. the SIP number, name and SIP Account number. The number is dialed out using the SIP Account assigned to it. The Access Code for dialing Phone book numbers is #8-Index Number. The user shall go off-hook and dial #8 followed by the Phone book index number (from 01 to 99). e.g. #8-64, to dial the number, stored at Index 64. If number is not programmed at index 64, user will get Error Tone. Click on Phone Book. Value/Default Index 01 to 99 Number Name SIP Account SIP number maximum 16 characters. Default = Blank Maximum 16 characters Default = Blank SIP 1 and SIP 2 Description This is the index number at which an entry is done. Enter the numberof person to be called. e.g Enter name of person, whose SIP number is stored in the phone book. Enter SIP Account number on which the OG call will be made, when dialing the specified number. Relevant Topic: 1. SIP Accounts 60 =X=X= 58 SETU ATA2S System Manual
59 Resetting the ATA Matrix What is this? ATA has the provision for changing the password. In case, user has forgotten the Password or he may need to default/reset the system, System Default feature is also provided. The system will loose all previous configuration, using this feature. ATA supports following methods to default: Default only SE Password through Access Code. Default System (Including SE Password) through Web Jeeves or SE command. How it works? For resetting through web page, follow following steps: Click on Default the ATA. User will find the alert message windows as This will assign default values to all the parameters of the ATA. Click on OK or Cancel. If clicked on OK, default values are assigned to the parameters. If clicked on Cancel, all the parameters remain same. Use following access code to set only the login password (SE Password) to default value: #***. It is not required to enter programming mode. Use following SE command to default the system: 21-Reverse SE Password-#*. Relevant Topic: 1. Getting Started 18 =X=X= SETU ATA2S System Manual 59
60 SIP Accounts This chapter describes how to configure basic SIP settings and Advanced VoIP and RTP parameters. Before starting it will be useful to refer some VoIP terms in the VoIP Basics chapter and Glossary. Click on SIP Accounts. Performance Parameters for SIP Accounts: Enter all values for SIP1 and SIP2. Parameters in SIP settings column are explained below: Registrar Settings: Enable SIP Account: Click on enable SIP1, SIP2 or both accounts. If SIP account is disabled, it will not be used to make OG calls even if it gets selected by the OG call routing logic. By default, SIP Account is Disable. SIP ID: Enter here, the user part of the full SIP URI. This can be a number or text. Field size: 24 characters maximum. Allow all ASCII characters. e.g. SIP URL provided by the ITSP is [email protected], enter in this filed. By default, SIP ID is Blank. Registrar Server Address: Enter the SIP registrar server s address. It can be IP address also and it s provided by ITSP. Field size: 40 characters maximum. Allowed all ASCII chars. By default, Registrar Server Address is Blank. Registrar Server Port: Enter here, the listening port for SIP. SE may change this if ITSP has provided the number, not same as default. Valid range is from 1025 to By default, Registrar Server Port is Re-Registration Timer: As a part of normal process, the register server deletes an entry of its client from its database on expiry of a fixed timer. This timer is set by the register server. Hence, in order to be registered always, the ATA should send a 60 SETU ATA2S System Manual
61 registration request before this timer expires. Enter the value of this timer here. It signifies the time after which the ATA should send registration request again to be registered. Valid range is from to By default, Re-Registration Timer is 3600 secs. Registration Retry Timer: Enter the period between retries for registration. On failure of registration, the ATA sends the registration request. If the registration attempts fails, the ATA should send the registration request on expiry of this timer. The ATA should keep sending the registration request till it gets registered and once it gets registered the ATA will start Re-Registration timer. Valid range is from to By default, Registration Retry Timer is 10 seconds. Authentication: User-ID: Enter the user name for registering the SIP account with the SIP register server. This field is relevant when SIP user ID and Authentication User ID are not same. It is provided by ITSP. Valid range is maximum 40 characters. By default, Authentication User ID is Blank. Password: Enter the password associated with the user name above. Valid range is maximum 16 characters. By default, Authentication Password is Blank. Vocoders Preference: Type : Two options for selecting Vocoder are provided: Single List By default, Vocoders Type is List. Single: If single is selected, preferred Vocoder only will be offered. Valid Range is G.711 a-law, G.711 u-law, G.723 and G.729. List : If clicked here, system will negotiate with remote phone from the available Vocoders when call is established. If list is selected, SETU ATA2S System Manual 61
62 62 list of supported Vocoders will be offered with selected Vocoder as preferred Vocoder. Vocoders : ATA supports 4-types of Vocoders: G.711 a-law. G.711 u-law. G.729. G.723. By default, Preferred Vocoder is G.729. DTMF Option: Select the option to decide, how the DTMF digits will be sent over IP, When digit is pressed. Inband, which means DTMF is combined in audio signal. Select outband, if digits are to be sent via RTP, using RFC Valid range is Inband, Outband. By default, DTMF Option is Outband. Outbound Proxy: Enable: Click to enable, if the ITSP service provider has a SIP outbound server to handle voice calls. By default, Enable Outbound Proxy is Blank. Server Address: Enter to specify the outbound proxy servers address. Allow IP address also. Valid range N.N.N.N, Where N= to , Maximum 48 characters. By default, Outbound Proxy Server Address is Blank. Server Port: Enter to specify the outbound proxy server s listening port for SIP. SE may change this field if the ITSP provides an outbound proxy sever port number other than the default. This may be same as SIP server s port address. Valid Range is from 1025 to By default, Outbound Proxy Server Port is STUN: Enable: Click, if there is a NAT router between the ATA and the ITSP s SIP server. SETU ATA2S System Manual
63 By default, Enable STUN is Blank. Server Address: Enter the STUN server s address. Allow IP address also. Allow extended ASCII characters. Valid range is N.N.N.N, Where N=0 to 255 or maximum 40 characters. By default, STUN Server Address is Blank. Server Port: enter the STUN server s listening port for SIP. SE may change this field if the ITSP provides a STUN Server port number other than the default. Valid range is from 1025 to By default, STUN Server Port is NAT Keep Alive: Enable: Click Yes to enable. The ATA s SIP messages will use NAT router s public IP address and SIP port number. The NAT should be also be configured with this port number to forward the traffic with this port number. NAT eliminates the need for STUN. By default, disable. Interval: Assign the time after which the ATA should send SIP notify messages to the SIP server. By default, NAT keep Alive interval is 120 seconds. Incoming Call-Routing: Incoming Calls: Select the port on which you want the incoming calls on SIP Account 1 to be routed and to land. Similarly specify the port for SIP Account 2 calls. Valid range is FXS1, FXS2 or Both. By default, FXS1 for SIP1 and FXS2 for SIP2. QOS (Layer 3): SIP QOS: This field specifies the QOS type viz. TOS (Precedence, also called Priority), or DiffServe for voice traffic. The ATA sends all the voice packets with this QOS setting. Precedence uses 3 bits and DiffServe uses 6-bits. Valid range SETU ATA2S System Manual 63
64 for Precedence is from 0 to 7 and for DiffServe is from 00 to 63. This parameter is same for SIP1 and SIP2. By default, Precedence is 5 and DiffServe is 26. RTP QOS: this field specifies the QOS type viz. TOS (Precedence, also called Priority) Or DiffServe for voice traffic. The ATA sends all the voice packets with this QOS setting. Precedence uses 3 bits and DiffServe uses 6-bits. Valid range for Precedence is from 0 to 7 and for DiffServe is from 00 to 63. This parameter is same for SIP1 and SIP2. By default, Precedence is 5 and DiffServe is 46. Relevant Topics: 1. VoIP Basics Glossary FXS Port Time Settings 82 =X=X= 64 SETU ATA2S System Manual
65 Software Upgrade What s this? ATA support an embedded FTP server which can be used for: Uploading/Downloading System files File Transfer Protocol (FTP), a standard Internet Protocol, is the simplest way to exchange files between computers on the Internet. Like the Hypertext Transfer Protocol (HTTP), which transfers displayable Web pages and related files, and the Simple Mail Transfer Protocol (SMTP), which transfers , FTP is an application protocol that uses the Internet s TCP/IP protocols. FTP is commonly used to download program and other files to your computer from other servers. Pop-up should be enabled from; TOOLS-Popup Blocker on the Internet Explorer. Procedure to upgrade file: To upgrade any file of the ATA software, click on S/W Upgrade. Four file folders are displayed: appli, config, driver, httpd. Modified file from User s Page is copied in the suitable file folder, config, driver or httpd. Old file is automatically replaced. To upgrade an appli file, first old one must be deleted and then new file is copied. Finally ATA has to be restarted for applying new file application. Mozilla-Client will not work for upgrading the Software. Relevant Topics: 1. Password Jeeves 44 =X=X= SETU ATA2S System Manual 65
66 Status What s this? ATA user may require to view remotely for WAN Addresses and registration status. Matrix ATA has this facility. The registration time is provided by the server. User can quickly have a glance for WAN address and can change it from respective page, if required. It also provides Registration Status which is useful to know whether SIP is registered or not. For example, if the Ethernet cable for Internet gets disconnected than after some time, Registration Status is displayed on the Status screen (if updated) and user can take corrective action, just by connecting the cable instead of restarting the system. The MAC Address used for Auto Configuration is programmable only at factory. It cannot be changed by user. Click on STATUS page. Following basic parameters are displayed: STATUS Parameters Display Description System Matrix SETU Name ATA2S This displays the name of system Version/ This displays the current version 'x' VxRy Revision and Reviosn 'y' of software WAN Port: IP Address or default Subnet Mask MAC Address Unique Address Number or Default C2.55.B0.18 This displays the programmed IP Address This displays the programmed subnet mask This displays the MAC Address programmed at factory. Default address will be dis]played only if it is not programmed at factory. It will not be defaulted, if user will try to default ATA. 66 SETU ATA2S System Manual
67 SIP Account1: Registration Status Registration Time Registration Retry Count Reg Last Failed Reason NAT Type Public IP Not Registered/ Registered 0,1,2 Blank display, Unknown, full cone, Restricted, Port Restricted, Symmetric, Symmetric Behind Firewall, Blocked N.N.N.N. N = 001 to 255 Matrix When 'Register' button is presses the ATA sends registration request to the server and the 'Status' is displayed. This time is provided server during registration This shows how many times after registration failure, ATA has sent registration request. This displays the reason for last time failure of registration. For example, Message sent failure. Response timeout. Failed response 486, etc. NAT is configured for option as shown. (1) If STUN is not enabled, it is Blank (No Display) (2) if none of the type of NAT is detected by ATA, then 'Unknown' is displayed. When STUN is enabled 'Public IP' is displayed here. SETU ATA2S System Manual 67
68 SIP Account2: Registration Status Registration Time Registration Retry Count Reg Last Failed Reason NAT Type Public IP Not Registered/ Registered 0,1,2 Blank display, Unknown, full cone, Restricted, Port Restricted, Symmetric, Symmetric Behind Firewall, Blocked N.N.N.N. N = 001 to 255 When 'Register' button is presses the ATA sends registration request to the server and the 'Status' is displayed. This time is provided by server during registration This shows how many times after registration failure, ATA has sent registration request. This displays the reason for last time failure of registration. For example, Message sent failure. Response timeout. Failed response 486, etc. NAT is configured for option as shown. (1) If STUN is not enabled, it is Blank (No Display) (2) if none of the type of NAT is detected by ATA, then 'Unknown' is displayed. When STUN is enabled 'Public IP' is displayed here. 68 SETU ATA2S System Manual
69 Auto Configuration Status Blank Statistics Blank Message displays after Auto Configuration is applied during Power ON: for e.g. 'No tftp Server Specified', 'Auto Configuration Fail', 'Auto Configuration Success'. Message displays after Auto Configuration is applied: for e.g. 'Files Downloded/Configured =X/Y'. Relevant Topics: 1. WAN Port SIP Accounts System Name Auto Configuration 29 =X=X= SETU ATA2S System Manual 69
70 Supplementary Services This chapter contains following sections: Introduction Programming Type Programming of Web Pages Programming using Phones Important Points Introduction: ATA provides following call features based on supplementary services provided by the ITSP: Making Second Call Call Hold Call Waiting Call Toggle Call Transfer-Blind Call Transfer-Attended 3-Party Conference Call Forward-Unconditional Call Forward-On Busy Call Forward-On No Reply DND Caller ID These supplementary services should be supported by the service provider. Making Second Call Telephone user can make second call while talking to first caller by keeping the first caller on hold. The user should keep the first caller on hold and then dial the destination number to which he wants to talk. On completing the talk, ask the called party to hang-up. When the called party hangs up, the user gets connected to the first caller. Also, while in speech with second call, he can talk to first user by using call toggle access code (except type 3 users). The user can also have conference between three parties by dialing conference access code. 70 SETU ATA2S System Manual
71 Call Hold Call Hold allows one party to maintain an idle (non-talking) connection with another party while receiving another call (Call Waiting) or while initiating another call (Call Transfer), or while performing some other call management functions. Call Waiting Call Waiting (CW) notifies an engaged caller of an incoming call and allows them to receive a call from a third party while the party with whom they have been speaking is put on hold. Call Transfer (Blind): Call Transfer allows one party to reconnect the party with whom, they have been speaking; to a third party. The first party is disconnected when the third party becomes connected. Call transfer without consulting to the party on whom call is to be transferred is Blind Transfer. Call Transfer-Attended Telephone user dials Call Hold access code followed by the Transfer target s number to transfer the caller (Transferee) to the Transfer target. In case of, Call transfer with consulting or Call Transfer-attended the user gets ring back tone followed by speech or Busy Tone. He may talk to transfer target and hang up. Call Toggle To attend to the second call, phone user has to use this feature by dialing Call Toggle Access Code. When the user dials the Call Toggle access code when a party is held, user is connected to the waiting caller. If no party is held, then user gets error tone. The user is connected to the current party when he goes ON-Hook, gets ring and again goes OFF-Hook. Type 3 user cannot use this feature. 3-Party Conference Telephone user dials Conference access code to have Conference between the User and the two callers. When the user dials the Conference access code (with one held SETU ATA2S System Manual 71
72 party and one current party): Conference is held between the user and the two callers. While during the Conference if the user dials the access code to Toggle (Not applicable for Type 3 users). User is connected to one of the callers. Other caller gets music. Programming Type: Following is the method of programming the features by phone as well as Jeeves: The following table shows the feature codes for various features to be used in three different types of feature code sets. Type 1 is a set of access codes used by Matrix-ATA. Type 2 is made considering the European access codes. Type 3 is made considering the American access codes. Type 1 (Matrix System) Type 2 (European System) Type 3 (American System) Call Hold Hold Flash Flash Flash Hold Retrieve Flash (within inter-digit timer) /Remind Ring Flash (within inter-digit timer) /Remind Ring Call Waiting Reject Waiting Flash-2 call Flash-0 -- Hold current call & answer Flash-1 Flash-2 Flash waiting call Disconnect current call and Flash-3 answer waiting Flash-1 -- call Flash (within inter-digit timer) /Remind Ring 72 SETU ATA2S System Manual
73 Call Transfer Blind Call Transfer Flash-*1- Number-ON Hook Flash-*98- Number-ON Hook Flash-#90- Number-ON Hook Attended Call Transfer Flash-Number- Flash-Number- Flash-Number- Speech-OnHook Speech-OnHook Speech-OnHook Conference Conduct Conference Flash-0 Flash-3 Flash Conference to Toggle Flash-1 Flash-2 -- Call Toggle Call Toggle Flash-1 Flash-2 -- Programming of Web Pages: Programming through Jeeves is as given below: Click on Supplementary Services Type Selection: Value/Default Description Feature Code Type Type1 Type2 Type3 Default = Type 1 Select Type from Type1, Type2 and Type3 as per the country in which ATA is installed. Select Type1 for Matrix System, Type2 for European System and Type3 for American System. SETU ATA2S System Manual 73
74 Call Forward and DND: Call Forward- Unconditional Call Forward- When Busy Call Forward- When No Reply Value/Default 16 digits max (Digits 0 to 9 and *) (Default: Blank) Note 1 16 digits max (Digits 0 to 9 and *) (Default: Blank) Note 1 16 digits max (Digits 0 to 9 and *) (Default: Blank) Note 1 Description (Programm for SIP1 and SIP2) Enter destination number to set call forward -for call coming on SIP1. Set another number for call coming on SIP2. Enter destination numbers to set call forward -for call coming on SIP1 which is busy on speech. Set another number for call coming on SIP2 (busy). Enter destination number to set call forward -for call coming on SIP1, in case this call is not attended. Set another number for SIP2. Call Forward Timer DND (Do Not Disturb) sec (45) Note 2 Set/Cancel (Cancel) Set the timer for which No Reply condition remains. After this time, system will initiate call forward. This is Do Not Disturb feature. Enable if you do not want to receive any calls. Note 1: The number field shall remain grey till the feature is enabled by checking it in the check box. Note 2: This field will be activated only if Call Forward-When No Reply is selected. Note 3: Call Forward and DND applies to SIP account and not FXS. 74 SETU ATA2S System Manual
75 Procedure of Calling: Call Forward Call Forwarding (Always) Incoming calls may always be forwarded to another designated party, and call forwarding founder can hear the notified rings. Example: 1 A calls B. NO TALK. 2 3 B responds: it is unavailable. So it returns a forwarding destination C. A is automatically forwarded to C. User at A is unaware of this. Talk between A, C. Matrix Call Forwarding (Busy): Incoming calls can be forwarded to another designated party if the phone of the called party is busy. 1 A calls B. NO TALK. 2 B responds, (Busy state). B is in busy state and the new forwarding state is C. 3 A is automatically forwarded to C. Talk between A, C. Call Forwarding (No-Answer): Incoming calls can be forwarded to another designated party when there is no response, from the original called party. B is ringing, A can hear 1 A calls B. ring back tone. 2 3 B responds to A, after 45 secs. (programmable) with no one picking up the phone. A is automatically forwarded to C. No answer and A needs to contact C. Talk between A, C. SETU ATA2S System Manual 75
76 Call Transfer (for Type 1): Call Transfer without consultation (Blind transfer): Assuming that call party A and B are in conversation. A wants to Blind Transfer B to C: To transfer incoming call to another phone: 1 Speech ON. A and B under talk. 2 Dial Flash on SLT-A. Feature tone 3 Dial * 1-Station-#. Ring back tone on B and C is ringing. 4 Note1. B and C talking. Note 1 is: (a) A gets confirmation tone followed by Dial tone if transfer is successful. (b) A gets busy tone for 2 seconds followed by speech, if transfer target is busy. (c) A gets continuous busy tone, if no response comes from transfer target. (d) A gets error tone, if transfer isn t successful due to either timeout or invalid codes have been entered. By default, Matrix Flash key Command. Call transfer with consultation (Attended Transfer) Event 1: When A&B are in communication, A puts B on hold and then dials Number of C plus the pound key #. Event 2: After A and C are in communication, A goes ON-Hook and C gets connected to B. Call Waiting (For Type 1): 1 C calls B, when A and B are B hears waiting tone, C can hear talking. ring back tone. 2 B press Flash 2 or B press Flash 3 or just hang-up B press Flash 1 Reject the second call. Disconnect first call and answer second call. A is on hold and B can answer to C. 76 SETU ATA2S System Manual
77 3-Party Call Conference (For Type 1): To make 3 party call conference. 1 Speech on. 2 Dial Flash. Feature tone (Second dial tone B on hold). 3 Dial Station-#. Ring Back Tone on B. C is ringing. 4 A presses Flash-0. A and B and C are talking. 5 A presses Flash-1. For conference toggle only A and B are takling. Event 1: When A and B are in communication, A puts B on hold then dials C s number plus the pound key #. Event 2: After A and C are in communication, press Flash-0 button again to initiate the 3 party call conference. Event 3: During conference if A goes ON-Hook, B gets transferred to C. Event 4: Press Flash-1 to separate activated 3 party call conference into two individual connections (One is online the other is on hold). Call Hold (For Type 1): Caller can use hook flash key to hold the call. 1 A calls B Talk is ON. 2 A presses Flash. A, hears second dial tone, B hears nothing. B is on Hold. 3 A can press again Talk between A, B. Flash. Programming using phone: Call Forward and DND: Use following commands: SETU ATA2S System Manual 77
78 Set Call Forward- Unconditional Cancel Call Forward- Unconditional Set Call Forward- When Busy Cancel Call Forward-When Busy Set Call Forward- When No Reply Feature Command 51-SIP Account- Destination Number-#* 52-SIP Account-#* 53-SIP Account- Destination Number-#* 54-SIP Account-#* 55-SIP Account- Destination Number-#* Cancel Call Forward-When No 56-SIP Account-#* Reply Set Call Forward Ring Timer Set DND Cancel DND 57-SIP Account-Timer (Time in seconds is from 00-99) 61-SIP Account-#* 62-SIP Account-#* Description Enter SIP Account number 1 or 2 for SIP Account. After issuing the command user gets Confirmation tone. -Do- -Do- -Do- -Do- -Do- Enter value in seconds for XX, for how long Ring should continue. Default=60 secs. Use command if DND feature required on a SIP Account Use command if DND feature is not required on a SIP Account. 1 for SIP1 and 2 for SIP2. 78 SETU ATA2S System Manual
79 Call Waiting Use following commands to enable/disable Call Waiting beeps on FXS: Enable Call Waiting 72-FXS-#* Disable Call Waiting 73-FXS-#* Where, FXS can be FXS1 or FXS 2 and 1 for FXS1 and 2 for FXS2. Note: Matrix ATA supports display of Caller ID of the third party s number when he is trying to call to a Matrix ATA, which is busy with speech. This feature requires that the phone connected to ATA, which supports Call Waiting, should be programmed for offline CLIP. Caller ID Caller ID of the user who is making call can be blocked or sent using Jeeves as well as Phone. Using Jeeves: Refer the page FXS settings for programming Caller ID through Jeeves. Using Phone: Use following commands to send/block Caller ID: FXS can be1 or 2 as per FXS1 Send Caller ID 66-FXS-#* or FXS2 on which this feature is required. Block Caller ID 67-FXS-#* FXS can be1 or 2 as per FXS1 or FXS2 on which this feature is required. Cancel all the features Many times the user does not get calls and he gets confused whether the calls are not coming due to network problem or the setting of some feature on the ATA. In such case, it is convenient to use some command for resetting any feature setting. Use the following command to cancel all the features: 99-#* SETU ATA2S System Manual 79
80 This command cancels all the features set for the SIP account or the Phone (FXS). Following features get reset: Call Forward-All types get canceled DND gets canceled. Important Points: When one SIP account is routed to both FXS ports, for incoming call: If one port is IDLE and other in speech, only port which is IDLE rings, even when other port has CW enabled. If both ports are in speech with CW enable, then both will receive CW tone for IC call. In case any of the FXS ports rejects the waiting call, other port doesn t get CW tone. Call Forward-always has priority over DND. Relevant Topics: 1. Time Settings FXS Port SIP Accounts 60 =X=X= 80 SETU ATA2S System Manual
81 System Name The System Name is for identification purpose. This screen allows user to change the system name. This feature helps the user identify the system. This is required when multiple ATA systems are connected in the same network. Now when accessing the ATA for configuring it, the name programmed in this field should appear as the system name. This shall help SE to confirm that he is programming the correct system. Click on System Name. Value/Default System Name Relevant Topic: 1. FXS Port char. Max. may be ASCII. Default = Matrix SETU ATA2S =X=X= Description Enter the name to identify ATA SETU ATA2S System Manual 81
82 Time Settings Matrix ATA can be configured for correct time based on the Simple Network Time Protocol (SNTP). SNTP is a protocol built on top of TCP that ensures accurate local time-keeping with reference to radio and atomic clocks located on the Internet. The ATA gets synchronized with the specified NTP server (IP Address), which is a global server and gets the date and global time. Based on the time zone, settings, Matrix ATA shows correct time for that country. For example, if time shown by ntp1.cs.wisc.edu server for India (Culcutta, Chennai, Mumbai, New Delhi) is 10:44:50AM, then current time shown for Kabul is 09:44:50AM. Because for India GMT is GMT but for Kabul it is GMT (1 hour behind India). The date is displayed as 17May2006. Click on Time Settings Field Name Value/Default Description Current Date DD-Month-YYYY ATA displays current system Date Current Time HH:MM:SS ATA displays current system Time ntpl.cd.wise.edu Time Server Select the suitable time.windows.com Address Time server time.nist.gov Time Zone List of places as per windows software of PC. Default = Calcutta Relevant Topics: 1. FXS Port Supplementary Services 70 =X=X= Select the place name, based on country where ATA is installed 82 SETU ATA2S System Manual
83 VoIP Basics Matrix VoIP is the sending of voice signal over Internet Protocol. Calling on VoIP allows user to make phone calls and send faxes over the internet at a negligible cost compared to the normal circuit switched Telephone network. The main components of VoIP are: SIP Network Servers RTP NAT and STUN Types of NAT STUN Outbound Proxy SIP ALG SIP Call Flow Important Voice Parameters: Jitter Buffer, Voice Compression and QoS. SIP: The signaling protocol which is used to establish an IP call is called SIP-Session Initiation Protocol. It is an application layer protocol that handles the setting up, altering and tearing down of voice and multimedia sessions over the internet. Refer RFC 3261 for more information. SIP handles telephone calls and can interfere with PSTN network. A SIP account uses complete identity called, URI (Uniform Resource Identifier) or SIP Address, like address and is given by: SIP-Number@SIP service Domain. SIP number can use letters or numbers like vrajeshp@itsp Provider.com or 73121@ITSP Provider.com. A SIP service domain is the domain name in the SIP URL. For example, if SIP address is 73121@VoIP Provider.com, then VoIP Provider.com is the SIP service domain. In general, SIP follows the Client-Server Architecture. To support this, there are two main entities-user agent and network servers. The peers in a session are called User Agents (UAS). A user agent can function in one of the following roles: User agent client (UAC)-A client application that initiates the SIP request. SETU ATA2S System Manual 83
84 User agent server (UAS)-A server application that contacts the user when a SIP request is received and that returns a response subject to user s input. Network Servers: There are three types of network servers in a SIP network: proxy server, redirect server and registrar server. Proxy Server: A SIP proxy receives a request, makes a determination about the next server to send it to, and forwards the request, possibly after modifying some of the header fields. As such, SIP requests can traverse many servers on their way from UAC to UAS. Responses to a request always travel along the same set of servers the request followed, but in reverse order. Redirect Server: The redirect server does not forward requests to the next server. Instead it sends a redirect response back to the client containing the address of the next server to contact. SIP Registrar: SIP Registrar is an entity where SIP users can get themselves registered. Registrar imparts mobility to the SIP users. A SIP user can register himself with a registrar. If the user changes his location, he has to register again with the registrar stating his latest contact information. Whenever a call is to be delivered to that user, Registrar can provide the information about the location where the user was active recently. RTP: After a VoIP call is established using SIP, the RTP (Real Time Protocol) is used to handle voice data transfer in the packets form. Refer RFC 1889 for more details. NAT and STUN: NAT (Network Address Translation-RFC 1631), is the translation of IP address of a host in a packet. For example, the source address of an outgoing packet, used within one network is changed to a different IP address known within another network. 84 SETU ATA2S System Manual
85 For using this feature, two terms are explained as below: a. NAT b. STUN One network is designated the inside network and the other is the outside. Typically, a company maps its local inside network addresses to one or more global outside IP addresses and unmaps the global IP addresses on incoming packets back into local IP addresses. This helps ensure security since each outgoing or incoming request must go through a translation process that also offers the opportunity to qualify or authenticate the request or match it to a previous request. NAT also conserves on the number of global IP addresses that a company needs and it lets the company use a single IP address in its communication with the world. Types of NAT: Four types of NAT s can be used: Full Cone NAT-This type of NAT maps all requests from the same internal IP address and port to the same external IP address and port. Any external host can send a packet to the internal host only if the internal host had previously sent a packet through the NAT. Restricted Cone NAT-This type of NAT maps all requests from the same internal IP address and port to the same external IP address and port. An external host (with IP address X) can send a packet to the internal host only if the internal host had previously sent a packet to IP address X. Port Restricted Cone NAT-This type of NAT maps all requests from the same internal IP address and port to the same external IP address and port. An external host (with IP address X and source port P) can send a packet to the internal host only if the internal host had previously sent a packet to IP address X and port P. Symmetric NAT-In a symmetric NAT, all requests from the same internal IP address and port to a specific destination IP address and port are mapped to the same external IP address and port. If the same host sends a packet with the same source address and port, but to a different destination, a different mapping is used. Furthermore, only the external host that receives a packet can send a UDP packet back to the internal host. SETU ATA2S System Manual 85
86 STUN: (RFC 3489) Note: STUN is used if there is a NAT router between ATA and SIP Server (from ITSP). STUN stands for Simple Traversal of UDP over NAT. It is a protocol, which enables an IP phone to detect the presence and type of NAT behind which the phone is placed. An IP phone that supports STUN can intelligently modify the private IP address and port in its SIP/SDP message by using the NAT mapped public IP address and port through a series of STUN queries against a STUN server located on the public Internet. This will allow SIP signaling and RTP media to successfully traverse a NAT without requiring any configuration changes on the NAT. STUN presents a working solution for most NATs that are not symmetric NAT, e.g., most of the SOHO routers have non-symmetric NAT and in this case, it is OK to use STUN. However, STUN does NOT work with symmetric NAT and if your routers have built-in symmetric NAT, do not use STUN. A STUN server can help facilitate traversing through most NATs, except for symmetric NATs. Outbound Proxy VoIP service provider can host a SIP outbound proxy server to handle all of the ATA s VoIP traffic. User has to enable this feature if VoIP service provider has a SIP outbound server to handle voice calls. This allows the ATA to work with any type of NAT router and eliminates the need of STUN. SIP ALG Some NAT routers includes the gateway which allows SIP calls to pass through NAT by checking and translating IP addresses embedded in the data stream to Public IP addresses. This type of NAT is known as SIP ALG, SIP Application Layer Gateways. If ATA is behind ALG, it eliminates need to use STUN or Outbound Proxy for NAT traversal. SIP Call flow: A simple example of how SIP requests and responses are sent will help understanding SIP and RTP. 86 SETU ATA2S System Manual
87 When A calls B, the steps for the set up and tear down of a SIP call is explained as below: A B 1. INVITE 2. Ringing 3. OK 4. ACK 5. Voice Data-RTP 6. BYE 7. OK 1. A sends SIP INVITE request to B. This message is an invitation to B to participate in a SIP telephone call. 2. B sends a response that it is ringing. 3. A answers the call. B responds it by OK. 4. B also answers the call and A sends acknowledgment by sending ACK message. 5. Now, voice session starts between A and B in form of RTP (Real Time Protocol). 6. After the talk, A hangs up and sends a BYE request. 7. Finally, B sends the reply by sending OK response and the call is terminated. Important Voice Parameters: Jitter Buffer: The clarity of voice heard from the called party is also distorted due to non-uniform delay introduced by the Telephony network known as Jitter. In order to reduce the effect of jitter, some sort of filters are used in the voice-channel called Jitter Buffer. The ATA has built-in adaptive buffer that helps to smooth out variations in delay (jitter) for voice traffic. This will ensure good voice quality. It is not programmable. Voice Compression: Voice compression is a technique to reduce the band width of voice channel, using suitable codec or Coder-Decoder. The codec SETU ATA2S System Manual 87
88 converts voice signals from analog form to digital signals and vice versa. It uses the compression alongwith, CELP. Compression-decompression is a voice compressiondecompression algorithm that defines the rate of speech compression, quality of decompressed speech and processing power requirements. It is called companding. Codec of suitable bandwidth is selected during the speech reception or transmission, to minimize the digital voice channel bandwidth. QoS: Quality of Service, QoS is the capability of a network to provide better service to selected network traffic over various technologies, including Ethernet and networks, and IP-routed networks. ATA supports Type of Service tagging and Differentiated Services tagging as mentioned below: (This allows ATA to tag voice frames so they can be prioritized over the network). 1. Type of Service (ToS) 2. DiffServe ToS: Network traffic can be classified by setting the ToS values at the data source (at the ATA) so that a server can decide best method of delivery, such as the least cost, fastest route etc. This parameter allows you to configure Type of Service (ToS) bits by specifying the precedence and delay of audio and signaling IP packets. DiffServe: DiffServe is differentiated Services, which defines a Class of Service model that makes packets so that they receive per hope treatment at DiffServe-compliant network services along the route, based on the application types and traffic flow. Packets are marked with DiffServe Code Point (DSCP) indicating the level of service desired. Relevant Topic: 1. Glossary 104 =X=X= 88 SETU ATA2S System Manual
89 WAN Explanation What is this? This chapter explains programming aspects of WAN ports, in detail. If you have already started programming using chapter Getting Started, do not use this information. ATA has facility to program WAN port by normal phone as well as the computer. For this, DHCP server used in the LAN is also discussed. For more details on IP and DHCP refer Glossary and Jeeves topics. How to program? Configuring the ATA ATA has a WAN port whose IP address is (default). The WAN port of ATA can be connected to one of the ports of the LAN Switch/Hub/Router (in Corporate Offices) OR it can be connected to the Ethernet port of the Broadband modem (in SOHO). When connected to LAN of the organization, the IP address of the WAN port should be changed according to the LAN addressing scheme. The LAN might be using static addresses or dynamic addresses. If static addresses are used, the SE should take the static address of the WAN port from the LAN administrator and program it as the IP address for the WAN port of ATA. If DHCP server is used in the LAN, then DHCP client should be enabled in the product using the embedded Web server. Doing so, the DHCP server in the LAN shall assign an IP address dynamically to WAN port of the ATA whenever ATA is restarted. The IP address of the WAN port can be configured using a computer as well as conventional instrument. (Please note that, only the IP address of WAN port can be programmed from conventional Phone and no other parameter/feature of the WAN port can be programmed from conventional Phone.) Also WAN IP address is requested from server if PPPoE server is enabled and User Name and Password are entered. Programming are explained with two options as mentioned below: Configuring IP Address of WAN port of ATA using computer Configuring IP Address of WAN port of ATA using conventional phone instrument. SETU ATA2S System Manual 89
90 Configuring the IP address of WAN port of ATA using Computer Make the connections as shown in the figure given below: WAN Port FXS1 Computer FXS2 SETU ATA2S Get the IP address for the WAN port of ATA from the LAN administrator or ITSP. Open Web browser (Internet Explorer or Netscape) on the computer connected to the WAN port of the ATA. Enter the default WAN port address ( ) in the URL field and press Enter. Embedded Web server takes the System Engineer (SE) to the Home page. The SE has to Log in into the ATA using password (default password = 1234). The SE will have to go to WAN port settings link and change the IP address of the WAN port from default to that given by the LAN administrator. The SE may change the Subnet Mask, Gateway address and DNS address if required. Once the new setting take effect the SE will have to re-enter the web server with new URL. Note: Ensure that the subnet of WAN and PC is same. 90 SETU ATA2S System Manual
91 Configuring the IP address of WAN port of ATA using Conventional Phone Instrument Make the connections as shown in the figure given below: WAN Port FXS1 FXS2 SETU ATA2S Get the IP address for the WAN port of ATA from the LAN administrator or ITSP. Pick up the Handset of the Conventional Phone. You get error tone, since you are not registered. If you are registered, you shall get dial tone. (This is less likely when configuring the ATA for the first time) Enter the Programming Access Code #19 followed by password You get programming tone. Enter the WAN IP address using following command: 11-WAN IP Address-#* Where, WAN IP Address is of 12 digits. Each octet is of three digits. If only single digit is used then it should be converted to three digits and then entered. For e.g. To enter IP address , enter command #*. You get confirmation tone for 5 secs. followed by programming tone. SETU ATA2S System Manual 91
92 Enter 0 to get out of programming mode. The SE can dial any programming command during confirmation tone. Similarly, if ATA is installed in Residences where a HUB or a switch is not used, the static IP address given by the Service Provider should be programmed as the WAN port address. However, if the Internet Telephony Service Provider provides dynamic IP address then also ATA will work and it will be possible to make calls and receive calls through IP network. In this case, the ATA will read the dynamic IP address assigned by the Service Provider s Router and use this address to make the IP calls. It is assumed that the connection provided by the Service Provider will be always available. Important Point: When SIP is not registered or Ethernet cable is not connected, error tone is coming on both FXS ports, but programming by phone is still possible. Relevant Topics: 1. Jeeves Glossary Supplementary Services WAN Port 93 =X=X= 92 SETU ATA2S System Manual
93 WAN Port Matrix ATA is receiving and transmitting data to the Internet Network through Wide Area Network (WAN) port. There are three ways to obtain WAN IP-address as mentioned below. Depending upon type of Internet connection, one of this is used. e.g. if Internet connection is through xdsl (Digital Subscriber Loop), PPPoE is selected. By manually entering a static IP address. By configuring the ATA in DHCP mode. By configuring the ATA in PPPoE mode. Point-to-Point Protocol over Ethernet (PPPoE) defined in RFC 2516, is a means of connecting one premise to Internet Service Provider. PPPoE is a method of encapsulating data for transmission to a far point. Its advantage is that, it eliminates need for the ITSP to manage the allocation of IP address. If DHCP or PPPoE is enabled, IP address is allocated dynamically by the respective server and static IP address need not be entered. Matrix provides unique MAC address for each ATA, which is programmed at factory. User cannot change it. It can be in the range of C2.55.B0.18 to C2.55.B0.1F (HEX Format). The registered series of MAC address can have a maximum of 8 different MAC addresses. Hence maximum of 8-ATA units can be connected in a single LAN. The MAC address is used by ITSP to maintain the files of individual ATA in the TFTP Server. Click on WAN. WAN Port: Parameters Value/Default Description Enable DHCP Default = Disable Click, if DHCP is supported by the ITSP Enable PPPoE Default = Disable Click, if PPPoE is supported by the ITSP SETU ATA2S System Manual 93
94 Maximum 16 User Name characters. Allow ASCII characters Maximum 16 Password characters. Allow ASCII characters Enable Static IP Default = Enabled Address IP Address IP Mask Gateway DNS: IP Address Domain Name N.N.N.N Default = N.N.N.N. Default = N.N.N.N Default = Blank Relevant Topics: 1. Glossary VoIP Basics Supplementary Services 70 Enter Name e.g. Matrix Enter Password Selete if static address available from ITSP Ener IP address provided by SE or ITSP while Network installation at your site. Do not enter if DHCP is enabled. Not required to change if other IP addresses are with same subnet Provided by ITSP N.N.N.N = To Provided by ITSP Default = Blank. Blank (40 char.). Allow Extended ASCII Provided by ITSP set. =X=X= 94 SETU ATA2S System Manual
95 Section 3: Appendices
96
97 Appendix A: Troubleshooting This section describes trouble shooting procedures for the Matrix ATA. Some most common Problems and Corrective Action are described below: Symptom The Power LED does not light up After Power ON, SIP1, SIP2 LEDs are fast blinking After Power ON, SIP1, SIP2 LEDs are slow blinking Jeeves: Matrix ATA cannot be accessed from the WAN Port using Jeeves Probable Corrective Action Go through this checklist : Unplug the power adapter from the Phone Adapter. Wait for five seconds. Then plug the power adapter into the Phone Adapter again. Make sure the power adapter you are using is the one included with the ATA. Make sure that power source is turned ON. Check that SIP Account is enabled and Registration with SIP Server is successful Check that on SIP Account page, SIP1 and SIP2 are not disabled Make sure that computer s IP address is in the same subnet as ATA s IP address. Ping the ATA. Click Start Programs Accessories Command Prompt. Type ping in the command prompt followed by ATA s IP address and Enter. ATA should respond. Otherwise make sure that computer s Ethernet adaptor is ok. Make sure that Password is correctly entered. Otherwise default the system. SETU ATA2S System Manual 97
98 Password was changed but now it is forgotten Cannot change the Password Telephone port: No dial tone, Error tone is heard. Only Error Tone is heard when user of any phone connected at FXS1 or FXS2, goes OFF-Hook No other way, except to restore the default configuration file by resetting ATA. Ensure that retyping of new Password is correct. Error tone may be due to SIP Account not registered. Check Phone port configuration. Ensure that it is the same phone port which is assigned the SIP Account number from the ISP. Make sure the telephone is plugged in appropriate port, Phone1 or Phone2. Verify the SIP ID number, entered is correct and active/enabled. Check that, on SIP Account page, SIP1 and SIP2 are enabled and on FXS port page for OG Calls, SIP1 assigned to FXS1 and SIP2 is assigned to FXS2 =X=X= 98 SETU ATA2S System Manual
99 Appendix B: Applications Typical deployment of SETU ATA2S is explained for following applications: 1. Making normal VoIP calls. 2. Making calls via PBX and connecting multiple ATA. Normal VoIP calls: Today the Internet Connection from the service provider is available at most of the places. Hence, Matrix ATA can be used widely in a home or small offices, for making and receiving VoIP calls through the ITSP. SETU ATA2S Telephone or Fax Machine IP Network Access (Modem or Router) Service Provider Network Operations IP Network IP Network Access (Modem or Router) SETU ATA2S PSTN VoIP Gateway Telephone or Fax Machine The above figure shows how the IP call is routed to the destination. The analog phone and the ATA convert the call into VoIP and sends to the Internet and ITSP s SIP server through the Modem or router. The server forwards it to the IP phones through the Internet or to analog phones through the VoIP Gateway and PSTN. SETU ATA2S System Manual 99
100 Calling via PBX: Not only SOHO applications, but Enterprises also have wide applications of Matrix-ATA. This is by way of providing facility to call PSTN and IP network through the PBX. Refer following figures: PABX Telephone PSTN Telephone FXO1 FXO2 Telephone Telephone Matrix ATA-1 LAN Matrix ATA-2 Router Internet Router LAN SIP Proxy Matrix ATA-3 Telephone Telephone =X=X= 100 SETU ATA2S System Manual
101 Appendix C: Product Specifications Configuration and Capacity: Maximum FXS Port WAN Port (Ethernet) DC Jack Application For connecting Analog Phone or Fax Machine To connect Ethernet cable router provided by the service provider To connect the power adapter for DC supply voltage No. of Ports Connector Type 2 RJ RJ 45, 10/100 Base T VoIP and Networking Protocol: MAC Address (IEEE802.3) IPv4-Internet Protocol Version 4 (RFC791) Upgradable to V6 (RFC1883) ARP-Address Resolution Protocol DNS-A record (RFC1706), SRV Record (RFC 2782) DHCP Client-(RFC 2131) ICMP-Internet Control Message Protocol (RFC729) TCP-Transmission Control Protocol (RFC793) UDP-User Datagram Protocol (RFC768) RTP-Real Time Protocol (RFC1889) (RFC1890) RTCP-Real Time Control Protocol (RFC1889) DiffServe (RFC2475), Type Of Service TOS (RFC791/1349) SNTP-Simple Network Time Protocol (RFC 2030) Voice Functionalities: SIP-Version 2 Two SIP Accounts can be used simultaneously DTMF: Inband and Outband (RFC2833) Call Progress Tone Generation: Select as per the country QoS: RTP QoS and SIP QoS using DiffServ and ToS (Precedence) Full Duplex Audio SETU ATA2S System Manual 101
102 Echo Cancellation: G.168, with programmable tail length of 8, 16 and 32 msecs. Forward error correction (FEC) Voice Activity Detection (VAD) Fax using T.38 and Pass Through Speech Volume Setting-Transmit and Receive Flash Timer Programming Voice Codecs: G.711 (a-law and u-law), G.729, G.723 Telephony features: Voice Calls using SIP proxy and Voice calls without using SIP proxy (Peer-to-Peer Calling) Call Waiting and Cancel Call Waiting Caller ID Blocking Call Forwarding 3-Party Conference Call Transfer Caller ID: Generation and Display Calling Party Control (CPC) Speed Dialing with Phone Book feature Time Settings: Synchronizing with specific Time Server Firewall Features; Outbound Proxy STUN NAT Keep Alive Provisioning, Administration and Maintenance: Auto Configuration Programming Using Web Page Software Upgrade using FTP Phone Programming of some parameters like: WAN IP Address DND 102 SETU ATA2S System Manual
103 LED Indication: 1 LED for Power 1 LED for each FXS Port 1 LED for each SIP Account Security : Password Protected Administrator Dimension (W x H x D) : 80x104x27mm (3.15x4.09x1.03Inch) Power Supply: External Adaptor Power Consumption Environmental: Operating Temperature Operating Humidity Storage Temperature Storage Humidity : [email protected]. : 5W (Typical) : -10 C to 50 C (140 to 122 F) : 5-95% RH, Non-Condensing : -40 C to 85 C (-40 F to 185 F) : 0-95% RH, Non-Condensing =X=X= SETU ATA2S System Manual 103
104 Appendix D: Glossary 10BaseT 10-Mbps base band Ethernet specification using two pairs of twistedpair cabling (Categories 3, 4, or 5): one pair for transmitting data and the other for receiving data. 10BASET, which is part of the IEEE specification, has a distance limit of approximately 328 feet (100 meters) per segment. A-law It is an ITU-T companding standard, used in the conversion between analog and digital signals in PCM systems. A-law is used primarily in European telephone networks and is similar to the North American μ- law standard. See also companding and μ-law. CELP It is the Code Excited Linear Prediction compression. It is the compression algorithm used in low bit-rate voice encoding. Used in ITU-T Recommendations G.728, G.729, G.723. Codec It is Coder decoder. In Voice over IP, Voice over Frame Relay, and Voice over ATM, a DSP software algorithm used to compress/ decompress speech or audio signals. Companding Contraction derived from the opposite processes of compression and expansion. Part of the PCM process whereby analog signal values are rounded logically to discrete scale-step values on a nonlinear scale. The decimal step number then is coded in its binary equivalent prior to transmission. The process is reversed at the receiving terminal using the same nonlinear scale. Compare with compression and expansion. See also a-law and μ-law. Compression The running of a data set through an algorithm that reduces the space required to store or the bandwidth required to transmit the data set. Compare with companding and expansion. 104 SETU ATA2S System Manual
105 DHCP It is the Dynamic Host Configuration Protocol. It provides a mechanism for allocating IP addresses dynamically so that addresses can be reused when hosts no longer need them. DNS It is the Domain Name System. System used on the Internet for translating names of network nodes into addresses. Firewall Router or access server or several routers or access servers, designated as a buffer between any connected public networks and a private network. A firewall router uses access lists and other methods to ensure the security of the private network. FTP File Transfer Protocol-A protocol used to transfer files over a TCP/IP network. FXS It is Foreign Exchange Station. An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone. Matrix- ATA s FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment and PBXs. G.711 Describes the 64-kbps PCM voice coding technique. In G.711, encoded voice is already in the correct format for digital voice delivery in the PSTN or through PBXs. Described in the ITU-T standard in its G-series recommendations. G.723 It is an ITU standard for speech codecs that uses the ADPCM method and provides toll quality audio of 20 and 40 Kbps. G.729 Describes CELP compression where voice is coded into 8-kbps streams. There are two variations of this standard (G.729 and G.729 Annex A) that differ mainly in computational complexity; both provide SETU ATA2S System Manual 105
106 speech quality similar to 32-kbps ADPCM. It is described in the ITU-T standard in its G-series recommendations. HTTP Hyper Text Transport Protocol-The communications protocol used to connect to servers on the World Wide Web. IP It is the Internet Protocol, Network layer protocol in the TCP/IP stack offering a connectionless internetwork-service. IP provides features for addressing, type-of-service specification, fragmentation and reassembly and security. It is defined in RFC 791. MAC Media Access Control Address - The unique address that a manufacturer assigns to each networking device. MAC is Media Access Control address in the form of 48-bit number, which is unique to the LAN NIC (network interface card). It is programmed into the card at the time of manufacture. IEEE registration authority administers MAC address scheme for all LANs which conform to IEEE, 802stds. Including both Ethernet and token ring. Consists of two parts: 24-bit company id (manufacturer ID0 and 24-bit Extension id (board id) Destination and source MAC names are contained in the header of the LAN packet and are used by various devices like hubs, bridges. A VoIP Service Provider will typically have its subscribers register with the Service Provider s VoIP service server before starting a subscriber s service. Often, an ITSP will require the registration of the MAC addresses of any devices directly connected to their network. NAT It is the Network Address Translation. Mechanism for reducing the need for globally unique IP addresses. NAT allows an organization with addresses that are not globally unique to connect to the internet by translating those addresses into globally routable address spaces, also known as Network Address Translator. 106 SETU ATA2S System Manual
107 NAT-Traversal It is a method of enabling specialized applications, such as Internet phone calls, video and audio, to travel between your local network and the Internet. STUN is a specific type of NAT traversal. NAT Server It is the Network Address Translation, an Internet standard that enables a local-area network (LAN) to use one set of IP addresses for internal traffic and a second set of addresses for external traffic. NTP It is the Network Time Protocol. Protocol built on top of TCP that assures accurate local time-keeping with reference to radio and atomic clocks located on the Internet. This protocol is capable of synchronizing distributed clocks within milliseconds over long time periods. Ping Packet Internet Groper-An Internet utility used to determine whether a particular IP address is online. Proxy Server It is an intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets, and if necessary, rewrites a request message before forwarding it. QoS It is the Quality of Service, the capability of a network to provide better service to selected network traffic over various technologies, including Frame Relay, Asynchronous Transfer Mode (ATM), Ethernet and networks, SONET, and IP-routed networks that may use any or all of these underlying technologies. The primary goal of QoS is to provide priority including dedicated bandwidth, controlled jitter and latency (required by some real-time and interactive traffic), and improved loss characteristics. SETU ATA2S System Manual 107
108 RTP It is the Real-Time Transport Protocol, one of the IPv6 protocols. RTP is designed to provide end-to-end network transport functions for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or unicast network services. RTP provides services such as payload type identification, sequence numbering, time stamping and delivery monitoring to real-time applications. SDP It is the Session Definition Protocol, an IETF protocol for the definition of Multimedia Services. SDP messages can be part of SGCP and MGCP messages. SIP It is the Session Initiation Protocol, a protocol developed by the IETF MUSIC Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March SIP equips platforms to signal the setup of voice and multimedia calls over IP networks. TCP It is the Transmission Control Protocol, a connection-oriented transport layer protocol that provides reliable full-duplex data transmission. TCP is part of the TCP/IP protocol stack. UAS It is the User agent server (or user agent), a server application that contacts the user when a SIP request is received and then returns a response on behalf of the user. The response accepts, rejects or redirects the request. UDP User Datagram Protocol-A network protocol for transmitting data that does not require acknowledgment from the recipient of the data that is sent. 108 SETU ATA2S System Manual
109 VAD It is the Voice activity detection, when enabled on a voice port or a dial peer, silence is not transmitted over the network, only audible speech. When VAD is enabled, the sound quality is slightly degraded but the connection monopolizes much less bandwidth. VoIP It is the Voice over IP, the capability to carry normal telephony-style voice over an IP-based Internet with POTS-like functionality, reliability, and voice quality. VoIP enables a router to carry voice traffic (for example, telephone calls and faxes) over an IP network. In VoIP, the DSP segments the voice signal into frames, which then are coupled in groups of two and stored in voice packets. =X=X= SETU ATA2S System Manual 109
110 Appendix E: Features at a Glance Description Feature Code To enter SE Mode. #19-SE Password To exit SE Mode. 0 To dial a phone book number. #8-Index Number To default SE Password #*** =X=X= 110 SETU ATA2S System Manual
111 Appendix F: System Commands Meaning To program the WAN IP Address To default ATA Set Call Forward-Unconditional Cancel Call Forward-Unconditional Set Call Forward-When Busy Cancel Call Forward-When Busy Set Call Forward-When No Reply Command 11-WAN IP Address-#* 21-Reverse SE Password-#* 51-SIP Account-Destination Number-#* 52-SIP Account-#* 53-SIP Account-Destination Number-#* 54-SIP Account-#* 55-SIP Account-Destination Number-#* Cancel Call Forward-When No Reply 56-SIP Account-#* Set Call Forward Ring Timer 57-SIP Account-Timer Set DND 61-SIP Account-#* Cancel DND 62-SIP Account-#* Send Caller ID 66-FXS-#* Block Caller ID 67-FXS-#* Enable Call Waiting beeps 72-FXS-#* Disable Call Waiting beeps 73-FXS-#* Cancel all features 99-#* =X=X= SETU ATA2S System Manual 111
112 Index A Applications-Refer Appendix B: Applications 99 Authentication-Refer SIP Accounts 60 Auto Configuration 29 B C Call Progress Tones-Refer FXS Port 34 Caller ID-Refer FXS Port 34 Call Party Control-Refer FXS Port 34 Calling Party Number Presentation-Refer FXS Port 34 Call-Routing-Refer SIP Accounts 60 Call Waiting-Refer Supplementary Services 70 Call Hold-Refer Supplementary Services 70 Call Toggle-Refer Supplementary Services 70 Call Transfer-Refer Supplementary Services 70 Call Forward-Refer Supplementary Services 70 Conference-Refer Supplementary Services 70 D Dial Plan 32 Default-Refer Resetting the ATA 59 Do Not Disturb-Refer Supplementary Services 70 DTMF Option-Refer SIP Accounts 60 Direct IP Calling-Refer Peer-to-Peer Calling 55 E Echo Cancellation-Refer FXS Port 34 F Flash Timer-Refer FXS Port 34 Forward Error Correction-Refer FXS Port 34 FXS Port Name-Refer FXS Port 34 FXS Port SETU ATA2S System Manual
113 G H I Incoming Call-Refer FXS Port 34 J Jeeves 44 K L M MAC Address-Refer WAN Port 93 N Number Dialing Timer-Refer FXS Port 34 New Password-Refer Password 54 NAT Keep Alive-Refer SIP Accounts 60 O Outgoing Calls-Refer FXS Port 34 Outbound Proxy-Refer SIP Accounts 60 P Password 54 Peer-to-Peer Calling 55 Phone Book 58 Polarity Reversal-Refer FXS Port 34 Precedence-Refer SIP Accounts 60 Q QoS-Refer SIP Accounts 60 R Ring Types-Refer FXS Port 34 Registrar Server-Refer SIP Accounts 60 SETU ATA2S System Manual 113
114 Re-Registration Timer-Refer SIP Accounts 60 Resetting the ATA 59 S SIP Accounts 60 SIP ID-Refer SIP Accounts 60 SIP Protocol-Refer Introducing the System 15 Software Upgrade 65 Status 66 STUN-Refer SIP Accounts 60 Supplementary Services 70 System Name 81 T Tones and Rings-Refer FXS Port 34 Time Settings 82 ToS-Refer SIP Accounts 60 U Upgrade-Refer Software Upgrade 65 V VAD-Refer Glossary 104 VoIP Basics 83 Volume Control-Refer FXS Port 34 Vocoder-Refer SIP Accounts 60 W WAN Explanation 89 WAN Port 93 X Y Z =X=X= 114 SETU ATA2S System Manual
115 Notes SETU ATA2S System Manual 115
116 Notes 116 SETU ATA2S System Manual
117 Programming Register S.N. Date Major Programming Changes made Register of Changes SETU ATA2S System Manual 117
118 Programming Register S.N. Date Major Programming Changes made Register of Changes 118 SETU ATA2S System Manual
ITC-BTTN Cellular Bluetooth Gateway. Owner s Manual 1
ITC-BTTN Cellular Bluetooth Gateway Owner s Manual 1 2 Table of Contents Introduction...3 Package Contents...3 XLink Connections Diagram...4 Setup...5 Pairing your Bluetooth Cell Phone to the XLink...6
User Manual 821121-ATA-PAK
User Manual 821121-ATA-PAK IMPORTANT SAFETY INSTRUCTIONS When using your telephone equipment, basic safety precautions should always be followed to reduce the risk of fire, electric shock and injury to
SIP Proxy Server. Administrator Installation and Configuration Guide. V2.31b. 09SIPXM.SY2.31b.EN3
SIP Proxy Server Administrator Installation and Configuration Guide V2.31b 09SIPXM.SY2.31b.EN3 DSG, DSG logo, InterPBX, InterServer, Blaze Series, VG5000, VG7000, IP590, IP580, IP500, IP510, InterConsole,
V101 SIP VoIP Telephone Adaptor User Manual V1.1m
V101 SIP VoIP Telephone Adaptor User Manual V1.1m Quick Guide Step 1: Broadband (ADSL/Cable Modem) Connections for V101 A. Connect V101 LAN port to ADSL NAT Router as the following connection. B. Connect
CRA 210 Analog Telephone Adapter 3 Ethernet Port + 2 VoIP Line + 1 PSTN Line
CRA 210 Analog Telephone Adapter 3 Ethernet Port + 2 VoIP Line + 1 PSTN Line Getting Started Guide Page: 1 of 30 Table of Contents 1. WELCOME - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
IMPORTANT NOTICE CONCERNING EMERGENCY 911 SERVICES
IMPORTANT NOTICE CONCERNING EMERGENCY 911 SERVICES Your service provider, not the manufacturer of the equipment, is responsible for the provision of phone services through this equipment. Any services
Wireless Router Setup Manual
Wireless Router Setup Manual NETGEAR, Inc. 4500 Great America Parkway Santa Clara, CA 95054 USA 208-10082-02 2006-04 2006 by NETGEAR, Inc. All rights reserved. Trademarks NETGEAR is a trademark of Netgear,
1. Installation Requirements
1. Installation Requirements 1.1. Package Contents Analog Telephone Adapter (CRA-210) Standard Telephone Cable (RJ11) Ethernet Cable (RJ45) Power Adapter 1.2. You will also need the following: 1.2.1. A
Prestige 2002 Series. VoIP Analog Telephone Adaptor. Quick Start Guide
VoIP Analog Telephone Adaptor Quick Start Guide Version 3.60 5/2005 Overview The Prestige allows you to use an analog telephone to make phone calls over the Internet (Voice over IP or VoIP). It uses SIP
SETU VP236 System Manual
SETU VP236 System Manual 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,
H.323 / SIP VoIP Gateway VIP GW. Quick Installation Guide
H.323 / SIP VoIP Gateway VIP GW Quick Installation Guide Overview This quick installation guide describes the objectives; organization and basic installation of the PLANET VIP-281/VIP-480/VIP-880/VIP-1680/VIP-2480
IMPORTANT NOTICE CONCERNING EMERGENCY 911 SERVICES
IMPORTANT NOTICE CONCERNING EMERGENCY 911 SERVICES Your service provider, not the manufacturer of the equipment, is responsible for the provision of phone services through this equipment. Any services
GW400 VoIP Gateway. User s Guide
GW400 VoIP Gateway User s Guide P/N: 956YD30001 Copyright 2006. All Rights Reserved. Document Version: 1.0 All trademarks and trade names are the properties of their respective owners. i Table of Contents
XPanel V2. Remote Control Panel. User Manual. XILICA Audio Design
XPanel V2 Remote Control Panel User Manual XILICA Audio Design Important Safety Instructions 1. READ THESE INSTRUCTIONS All the safety and operating instructions should be read before the product is operated.
DPH-140S SIP Phone Quick User Guide
DPH-140S SIP Phone Quick User Guide Version 1.0 TABLE OF CONTENTS 1.0 INTRODUCTION... 1 2.0 PACKAGE CONTENT... 1 3.0 LIST OF FIGURES... 2 4.0 SUMMARY OF KEY FUNCTIONS... 3 5.0 CONNECTING THE IP PHONE...
Voice Internet Phone Gateway
Voice Internet Phone Gateway Quick Installation Guide IPC 1000 Series ARTDio Company Inc. Edition 1.0 Note: For more detailed hardware installation instructions, please refer to the IPC 1000 series User
ReadyNet Easy Jack 2 Voice/Data and Data Only Owner s Manual PX-211d and PX-211v
ReadyNet Easy Jack 2 Voice/Data and Data Only Owner s Manual PX-211d and PX-211v Phonex Broadband Corporation dba ReadyNet 6952 High Tech Drive Midvale, Utah 84047 801.566.0100 Phone 801.566.0880 Fax www.readynetsolutions.com
SIMADO GFX44 System Manual
SIMADO GFX44 System Manual 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,
P160S SIP Phone Quick User Guide
P160S SIP Phone Quick User Guide Version 2.2 TABLE OF CONTENTS 1.0 INTRODUCTION... 1 2.0 PACKAGE CONTENT... 1 3.0 LIST OF FIGURES... 2 4.0 SUMMARY OF KEY FUNCTIONS... 3 5.0 CONNECTING THE IP PHONE... 4
FUTURE CALL PICTURE CARE PHONE MODEL: FC-1007 USER MANUAL
FUTURE CALL PICTURE CARE PHONE MODEL: FC-1007 USER MANUAL Please follow instructions for repairing if any otherwise do not alter or repair any parts of device except specified. IMPORTANT SAFETY INSTRUCTIONS
AudioCodes. MP-20x Telephone Adapter. Frequently Asked Questions (FAQs)
AudioCodes MP-20x Telephone Adapter Frequently Asked Questions (FAQs) Page 2 AudioCodes Customer Support Table of Contents Introduction... 6 Frequently Asked Questions... 7 Web Access... 7 Q1: How must
Phone Adapter. with 2 Ports for Voice-over-IP. Installation and Troubleshooting Guide. Model No. PAP2 Ver. 2. Voice
Phone Adapter with 2 Ports for Voice-over-IP Voice Installation and Troubleshooting Guide Model No. PAP2 Ver. 2 Copyright and Trademarks Specifications are subject to change without notice. Linksys is
DVG-2101SP VoIP Telephone Adapter
This product can be set up using any current web browser, i.e., Internet Explorer 6 or Netscape Navigator 6.2.3. DVG-2101SP VoIP Telephone Adapter Before You Begin 1. If you purchased this VoIP Telephone
VP100 QUICK INSTALLATION GUIDE. VoIP Phone
VP100 VoIP Phone For more exciting new products please visit our website: Australia: www.uniden.com.au New Zealand: www.uniden.co.nz QUICK INSTALLATION GUIDE IMPORTANT SAFETY INSTRUCTIONS When using your
NeoGate TA Series Quick Installation Guide
NeoGate TA Series Quick Installation Guide Version: V1.1 Yeastar Technology Co., Ltd. Date: November 18, 2014 http://www.yeastar.com 1/15 Contents NeoGate TA Series Quick Installation Guide 1. Preparation
Connecting the DG-102S VoIP Gateway to your network
Contents of Package: DG-102S VoIP Station Gateway Power adapter CD-ROM, including User s Manual Quick Install Guide Requirements: RS-232 Console Cable Two RJ-45 CAT-5 Straight-Through Cables For more information
User Manual. Page 2 of 38
DSL1215FUN(L) Page 2 of 38 Contents About the Device...4 Minimum System Requirements...5 Package Contents...5 Device Overview...6 Front Panel...6 Side Panel...6 Back Panel...7 Hardware Setup Diagram...8
Linksys Gateway SPA2100-SU Manual
Linksys Gateway SPA2100-SU Manual Manuel de l'utilisateur Table of Contents Looking for Basic Setup Instructions?... 3 Most Recent Version of this Manual... 3 Advanced Setup Instructions... 4 Wiring Your
6002TA Analog Port Terminal Adapter User Manual
6002TA Analog Port Terminal Adapter User Manual Contents Introduction... 1 Operation... 3 Placing a Call... 3 Answering a Call... 3 Switching a Call Between the POTS Port and Speakerphone or Handset...
Broadband Phone Gateway BPG510 Technical Users Guide
Broadband Phone Gateway BPG510 Technical Users Guide (Firmware version 0.14.1 and later) Revision 1.0 2006, 8x8 Inc. Table of Contents About your Broadband Phone Gateway (BPG510)... 4 Opening the BPG510's
User Manual. SIP Analog Telephone Adaptor SIP-GW2. Sedna Advanced Electronics Ltd. www.sednacomputer.com
User Manual SIP-GW2 SIP Analog Telephone Adaptor Sedna Advanced Electronics Ltd. www.sednacomputer.com Table of Contents 1. WELCOME... 3 2. INSTALLATION... 3 3. WHAT IS INCLUDED IN THE PACKAGE... 5 3.1
In addition to our VoiceDirector hardware products, the following SIP broadband devices are also compatible with VoiceDirector:
Device Compatibility Along with the full range of VoiceDirector devices we offer, a number of other SIP telephony products are compatible with the VoiceDirector corporate calling solution In addition to
Welcome. Unleash Your Phone
User Manual Welcome Unleash Your Phone For assistance with installation or troubleshooting common problems, please refer to this User Manual or Quick Installation Guide. Please visit www.vonage.com/vta
Quick Installation and Configuration Guide OX10
Quick Installation and Configuration Guide OX10 Hybrid Office Business Telephone System with Built-In Analog Phone Ports, CO Line Ports and SIP Adaptors Table of Contents 1. INTRODUCTION... 3 2. GETTING
Voice Gateway with Router
Voice User Guide Model No. SPA3102 Copyright and Trademarks Specifications are subject to change without notice. Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates
LifeSize Networker Installation Guide
LifeSize Networker Installation Guide November 2008 Copyright Notice 2006-2008 LifeSize Communications Inc, and its licensors. All rights reserved. LifeSize Communications has made every effort to ensure
Note: these functions are available if service provider supports them.
Key Feature New Feature Remote Maintenance: phone can be diagnosed and configured by remote. Zero Config: automated provisioning and software upgrading even through firewall/nat. Centralized Management:
Chapter 1 Installing the Gateway
Chapter 1 Installing the Gateway This chapter describes how to set up the wireless voice gateway on your Local Area Network (LAN), connect to the Internet, and perform basic configuration. For information
P-2024. Quick Start Guide. VoIP Analog Telephone Adaptor DEFAULT LOGIN. IP Address http://192.168.5.1 Password 1234. Version 3.60 7/2007 Edition 1
P-2024 VoIP Analog Telephone Adaptor Quick Start Guide Version 3.60 7/2007 Edition 1 DEFAULT LOGIN IP Address http://192.168.5.1 Password 1234 Copyright 2007. All rights reserved. Overview Use your P-2024
Adapter GL386. User Manual is available in other languages at
Adapter GL386 User Manual is available in other languages at www.glipfone.com GL386 User Manual Contents: Chapter 1 Introduction ---------------------------------------------------------------- 1 Chapter
VoIP Router TA G81022MS User Guide
VoIP Router TA G81022MS User Guide V. 1.0 TABLE OF CONTENTS TABLE OF CONTENTS...2 1.0 INTRODUCTION...1 2.0 PACKAGE CONTENT...1 3.0 SUMMARY OF LED & CONNECTOR DESCRIPTION...2 3.1 THE FRONT LEDS...2 3.2
CPEi 800/825 Series. User Manual. * Please see the Introduction Section
CPEi 800/825 Series User Manual * Please see the Introduction Section Contents Introduction...iii Chapter 1: CPEi 800/825 User Guide Overview... 1-1 Powerful Features in a Single Unit... 1-2 Front of the
Internet Telephony PBX System. (30/100 SIP Users registrations) IPX-330 / IPX-2100. Quick Installation Guide
Internet Telephony PBX System (30/100 SIP Users registrations) IPX-330 / IPX-2100 Quick Installation Guide Table of Contents 1. IPX-330... 3 1.1. Package Contents... 3 1.2. Hardware Installation... 3 1.2.1
your Gateway Windows network installationguide 802.11b wireless series Router model WBR-100 Configuring Installing
your Gateway Windows network installationguide 802.11b wireless series Router model WBR-100 Installing Configuring Contents 1 Introduction...................................................... 1 Features...........................................................
Vertex VoIP Caller ID (Version 1.5)
Vertex VoIP Caller ID (Version 1.5) Introduction The Vertex unit is designed to capture Caller ID and other telephony signaling on VoIP phone calls and send this information to computers. Depending on
WEB CONFIGURATION. Configuring and monitoring your VIP-101T from web browser. PLANET VIP-101T Web Configuration Guide
WEB CONFIGURATION Configuring and monitoring your VIP-101T from web browser The VIP-101T integrates a web-based graphical user interface that can cover most configurations and machine status monitoring.
IP Telephony. User Guide. System SPA9000. Model No. Voice
IP Telephony System User Guide Voice Model No. SPA9000 Copyright and Trademarks Specifications are subject to change without notice. Linksys is a registered trademark or trademark of Cisco Systems, Inc.
P-2302HWUDL-P1. Quick Start Guide. 802.11g Wireless VoIP Station Gateway. with Built-in DECT Base Station
P-2302HWUDL-P1 802.11g Wireless VoIP Station Gateway with Built-in DECT Base Station Quick Start Guide Version 3.60 Edition 1 3/2007 Overview The P-2302HWUDL-P1 model is a router with IEEE 802.11g wireless
Product Manual. Precision Inbound Call Routing Fast Outbound Line Hunting Streamlined Telecommunications
MJNOVIS Expanding Communications Product Manual Precision Inbound Call Routing Fast Outbound Line Hunting Streamlined Telecommunications 1 LINE 2 3 4 1 2 3 4 5 6 7 8 9 10 11 12 LINE 2 LINE 3 LINE 4 MULTI-LINK
Prestige 202H Plus. Quick Start Guide. ISDN Internet Access Router. Version 3.40 12/2004
Prestige 202H Plus ISDN Internet Access Router Quick Start Guide Version 3.40 12/2004 Table of Contents 1 Introducing the Prestige...3 2 Hardware Installation...4 2.1 Rear Panel...4 2.2 The Front Panel
DEX28-VR3P Quick Installation Guide
DEX28-VR3P Quick Installation Guide DEX28-VR3P PBX DTSs 1 3 5 2 4 6 Procedure to start recording: 1. Connect ODD ports (1, 3 or 5) to PBX (MD110 ELU28 or MX-ONE ELU33 board) 2. Connect EVEN ports (2, 4
2100 Series VoIP Phone
2100 Series VoIP Phone Installation and Operations Manual Made in the USA 3 Year Warranty N56 W24720 N. Corporate Circle Sussex, WI 53089 RP8500SIP 800-451-1460 262-246-4828 (fax) Ver. 4 www.rathmicrotech.com
Prestige 2302R Series
VoIP Station Gateway Quick Start Guide Version 3.60 6/2005 Overview This Quick Start Guide covers the Prestige (P2302R) and (P2302RL) models. It explains how to use your Prestige to make phone calls through
PLANET is a registered trademark of PLANET Technology Corp. All other trademarks belong to their respective owners.
Trademarks Copyright PLANET Technology Corp. 2004 Contents subject to revise without prior notice. PLANET is a registered trademark of PLANET Technology Corp. All other trademarks belong to their respective
How To Set Up An Andsl Modem Router For Internet Access
ADSL Modem Router Setup Manual NETGEAR, Inc. 4500 Great America Parkway Santa Clara, CA 95054 USA 208-10026-01 2006-2 2006 by NETGEAR, Inc. All rights reserved. Trademarks NETGEAR is a trademark of Netgear,
Frontier DSL SelfConnect Guide
Frontier DSL SelfConnect Guide Frontier DSL Self-Installation Guide Table of Contents Getting Started...2 Customer and Computer Requirements...2...3 STEP 1: Install Microfilters...3 STEP 2: Install Your
SOYO G668 VOIP IP PHONE USER MANUAL
SOYO G668 VOIP IP PHONE USER MANUAL Inglos Networks Industrial Global Solutions Teléfono: +1 (585) 217-9864, Fax: + 1 (585) 872-9627, Email: [email protected] Table of Content SAFETY INFORMATION... 1 INTRODUCTION...
ZyXEL IP PBX Support Note. ZyXEL IP PBX (X2002) VoIP. Support Notes
ZyXEL IP PBX (X2002) VoIP Support Notes Version 1.00 October 2008 1 Contents Overview ZyXEL IP PBX Support Note 1. How to manage and maintain your IPPBX?...3 1.1 Firmware Upgrade..3 1.2 Backing up your
Box Camera Series Hardware Manual
Encoder Firmware V4.06.09 User s Manual Box Camera Series Hardware Manual D21 (D21F / D21V) D22 (D22F / D22V) E21 (E21F / E21V) E22 (E22F / E22V) E23, E24, E25 2013/08/27 Table of Contents Precautions...
Cisco ATA 187 Analog Telephone Adaptor
Cisco ATA 187 Analog Telephone Adaptor Product Overview The Cisco ATA 187 Analog Telephone Adaptor is a handset-to-ethernet adaptor that turns traditional telephone devices into IP devices. Customers can
TCP/IP MODULE CA-ETHR-A INSTALLATION MANUAL
TCP/IP MODULE CA-ETHR-A INSTALLATION MANUAL w w w. c d v g r o u p. c o m CA-ETHR-A: TCP/IP Module Installation Manual Page Table of Contents Introduction...5 Hardware Components... 6 Technical Specifications...
ICE 008 IP PBX. 1. Product Information. 1.1. New Mini PBX. 1.2. Features 1.2.1. System Features
1. Product Information 1.1. New Mini PBX ICE 008 IP PBX ICE008 is new generation office communication equipment that delivers traditional PBX (private branch exchange) functions and more with advanced
JKW-IP. IP Video Entry System. QuikStart Guide
1210 JKW-IP IP Video Entry System QuikStart Guide This is an abbreviated instruction manual for installation purposes. Please see the JKW-IP Installation Manual and JKW-IP Operation Manual for complete
CallFinder. Model CF220 DID Adapter Quick Start Guide
CallFinder Model CF220 DID Adapter Quick Start Guide CallFinder Model CF220 DID Enabler Quick Start Guide P/N 82000160, Revision A Copyright 2004 by Multi-Tech Systems, Inc. All rights reserved. This publication
RouteFinder SOHO. Quick Start Guide. SOHO Security Appliance. EDGE Models RF825-E, RF825-E-AP CDMA Models RF825-C-Nx, RF825-C-Nx-AP
RouteFinder SOHO SOHO Security Appliance EDGE Models RF825-E, RF825-E-AP CDMA Models RF825-C-Nx, RF825-C-Nx-AP Quick Start Guide RouteFinder RF825 Series Quick Start Guide RouteFinder SOHO Security Appliance
3 Residential VoIP Service
User Guide 3 Residential VoIP Service Content 1.0 About VoIP Service 1.1 System Requirement 1.2 Enquiry and Support 2.0 VoIP Access Device 2.1 Hardware Description 2.2 Connection Map 2.3 Connection Steps
Setting Up the Cisco Unified IP Phone
CHAPTER 3 This chapter includes the following topics, which help you install the Cisco Unified IP Phone on an IP telephony network: Before You Begin, page 3-1 Understanding the Cisco Unified IP Phone 7962G
CelluLine CGW-TS GSM Cellular Gateway. Installation and Programming Manual
CelluLine CGW-TS GSM Cellular Gateway Installation and Programming Manual CelluLine CGW-TS GSM Cellular Gateway Installation and Programming Manual CGWTS-M001A Version 1, Release 1, December 2004 NOTICE
MyPBX U100 & U200 Installation Guide
MyPBX U100 & U200 Installation Guide Version: V1.0 Yeastar Technology Co., Ltd Date: 25 th February, 2013 http://www.yeastar.com 1/14 Contents MyPBX U100 & U200 Installation Guide 1. PREPARATION BEFORE
LevelOne VOI-9000. H.323 VoIP Gatekeeper. User Manual
LevelOne VOI-9000 H.323 VoIP Gatekeeper User Manual Quick Guide Step 1: Broadband (ADSL/Cable Modem) Connections For VOI-9000 A. Connect VOI-9000 RJ45 LAN port to Router/ADSL as one of the following connections.
Quick Start Guide. Cisco SPA232D Mobility Enhanced ATA
Quick Start Guide Cisco SPA232D Mobility Enhanced ATA Package Contents Analog Telephone Adapter Ethernet Cable Phone Cable Power Adapter Quick Start Guide Product CD-ROM Welcome Thank you for choosing
VOI-7000 VOI-7100 SIP IP Telephone
VOI-7000 VOI-7100 SIP IP Telephone User Manual 1 Ver 2.01-0609 Table of Contents 1. INTRODUCTIONS... 1 1.1. FEATURES... 1 1.2. PACKING CONTENTS... 2 1.3. LCD DISPLAY AND KEYPADS... 2 2. INSTALLATIONS &
Linksys SPA2102 Router Configuration Guide
Linksys SPA2102 Router Configuration Guide Dear 8x8 Virtual Office Customer, This Linksys guide provides instructions on how to configure the Linksys SPA2102 as a router. You only need to configure your
MyPBX U510 Installation Guide
MyPBX U510 Installation Guide Version 1.1 Date: 9th, Aug, 2013 Yeastar Information Technology Co. Ltd Contents 1. PREPARATION BEFORE INSTALLATION... 4 2. HARDWARE SPECIFICATIONS... 5 2.1 Overview... 5
Quick set-up instructions for. The Avois AV-3500 IP Phone
Solwise Ltd. Quick set-up instructions for The Avois AV-3500 IP Phone www.solwiseforum.co.uk The Solwise Forum is designed to be the first port-of-call for technical support and sales advice for the whole
IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online
1 IP PBX SD Card Slot FXO Ports PBX LAN port PBX WAN port FXO Ports LED, RED means online 2 Connect the IP PBX to Your LAN Internet PSTN Router Ethernet Switch FXO Ports 3 Access the PBX s WEB GUI The
Crow Limited Warranty. Print Version 017
Crow Limited Warranty (Crow) warrants this product to be free from defects in materials and workmanship under normal use and service for a period of one year from the last day of the week and year whose
AudioCodes Mediant 1000 Configuration Guide
AudioCodes Mediant 1000 Configuration Guide 2010 FaxBack, Inc. All Rights Reserved. NET SatisFAXtion and other FaxBack products, brands and trademarks are property of FaxBack, Inc. Other products, brands
ON HOLD ANNOUNCER. Once you receive your audio announcer, check the packaging to ensure that all of the following items are enclosed:
ON HOLD ANNOUNCER The is a high quality digital on-hold announcer. It is designed to be attached to a 100BASE-T Ethernet network to receive audio production updates via the Internet. These instructions
English version. LW320/LW321 Sweex Wireless 300N Router. Package Contents. Terminology list
LW320/LW321 Sweex Wireless 300N Router Do not expose the Sweex Wireless 300N Router to extreme temperatures. Do not place the device in direct sunlight or in the direct vicinity of heating elements. Do
Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2
Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2 Updated: February 2009 Microsoft Response Point is a small-business phone solution that is designed to be easy to use and
How To Program A Talkswitch Phone On A Cell Phone On An Ip Phone On Your Ip Phone (For A Sim Sim) On A Pc Or Ip Phone For A Sim Phone On Iphone Or Ipro (For An Ipro) On
TALKSWITCH DOCUMENTATION ADDING IP PHONES TO TALKSWITCH RELEASE 6.50 CT.TS005.008104 ANSWERS WITH INTELLIGENCE COPYRIGHT INFORMATION Copyright 2011 Fortinet, Inc. All rights reserved. Fortinet, FortiGate,
WLAN600 Wireless IP Phone Administrator s Guide
WLAN600 Wireless IP Phone Administrator s Guide Trademark Acknowledgement All brand names are trademarks or registered trademarks of their respective companies. Disclaimer This document is supplied by
Broadband Router ESG-103. User s Guide
Broadband Router ESG-103 User s Guide FCC Warning This equipment has been tested and found to comply with the limits for Class A & Class B digital device, pursuant to Part 15 of the FCC rules. These limits
USER PRECAUTION Please read the instruction carefully to protect yourself and others from personal injury or damage to property.
Dexter IP330S PRECAUTION USER PRECAUTION Please read the instruction carefully to protect yourself and others from personal injury or damage to property. To use the phone correctly and safely and prevent
VoIP Network Configuration Guide
The owner friendly phone system for small business VoIP Network Configuration Guide Release 7.10 Copyright 2011 Fortinet, Inc. All rights reserved. Fortinet, FortiGate, FortiGuard, FortiCare, FortiManager,
Configuration Notes 0217
PBX Remote Line Extension using Mediatrix 1104 and 1204 Introduction... 2 Application Scenario... 2 Running the Unit Manager Network (UMN) Software... 3 Configuring the Mediatrix 1104... 6 Configuring
SIP Print Administrator Guide
Version 1.25 Published 2/6/09 Table of Contents Unpack, Install and Power Up SIP Print...3 Port Mirroring Discussion...4 Connect SIP Print to the Network...6 Main Screen...7 Manage User Access...8 Manage
Panasonic. Proprietary Telephone for Electronic Modular Switching System MODEL NO. KX-17030. Illustrated Model: White 1
Panasonic Proprietary Telephone for Electronic Modular Switching System MODEL NO. KX-17030 Illustrated Model: White 1 KX-T7030 is compatible with all of the Panasonic Electronic Modular Switching Systems
VOIP Business Phone User Guide
VOIP Business Phone User Guide Model 25630/25600 MGCP Please read this manual before operating the product for the first time. Interference Information This device complies with Part 15 of the FCC Rules.
FortiVoice. Version 7.00 User Guide
FortiVoice Version 7.00 User Guide FortiVoice Version 7.00 User Guide Revision 2 28 October 2011 Copyright 2011 Fortinet, Inc. All rights reserved. Contents and terms are subject to change by Fortinet
Internet Telephony PBX system IPX-1980
Internet Telephony PBX system IPX-1980 Quick Installation Guide Table of Contents 1. Package Contents... 3 2. Hardware Installation... 4 2.1 Safety Instruction... 4 2.2 Front panel... 4 2.3 LED & Button
Conference Phone UserÕs Manual. Part No. 54-2070-01R1 Printed in Korea. 2002 Bogen Communications, Inc.
Part No. 54-2070-01R1 Printed in Korea. 2002 Bogen Communications, Inc. UserÕs Manual Notice Every effort was made to ensure that the information in this guide was complete and accurate at the time of
VoIP 110R/200R/422R/404R/440R. User s Guide
VoIP 110R/200R/422R/404R/440R User s Guide Trademarks Contents are subject to revise without prior notice. All trademarks belong to their respective owners. FCC Warning This equipment has been tested and
LifeSize Phone User Guide
LifeSize Phone User Guide April 2008 Copyright Notice 2005-2008 LifeSize Communications Inc, and its licensors. All rights reserved. LifeSize Communications has made every effort to ensure that the information
DSL-2600U. User Manual V 1.0
DSL-2600U User Manual V 1.0 CONTENTS 1. OVERVIEW...3 1.1 ABOUT ADSL...3 1.2 ABOUT ADSL2/2+...3 1.3 FEATURES...3 2 SPECIFICATION...4 2.1 INDICATOR AND INTERFACE...4 2.2 HARDWARE CONNECTION...4 2.3 LED STATUS
ADSL MODEM. User Manual V1.0
ADSL MODEM User Manual V1.0 CONTENTS 1.OVERVIEW... 3 1.1 ABOUT ADSL... 3 1.2 ABOUT ADSL2/2+... 3 1.3 FEATURES... 3 2 SPECIFICATION... 4 2.1 INTERFACE INTRODUCTION... 4 2.1.1 INDICATOR AND INTERFACE...
IP DSLAM IDL-2402. Quick Installation Guide
IP DSLAM IDL-2402 Quick Installation Guide Table of Contents Package Contents... 3 Overview... 4 Setup the IDL series IP DSLAM... 5 Safety Instruction... 5 Hardware Installation... 6 WEB Configuration...
CABLE MODEM QUICK START
CABLE MODEM QUICK START This Quick Start describes how to connect your Zoom cable modem to a cable modem service. This lets your cable modem provide Internet access to a computer or other device connected
FortiVoice. Version 7.00 VoIP Configuration Guide
FortiVoice Version 7.00 VoIP Configuration Guide FortiVoice Version 7.00 VoIP Configuration Guide Revision 2 14 October 2011 Copyright 2011 Fortinet, Inc. All rights reserved. Contents and terms are subject
