ATA Configuration 1. Features ..................................5 1.1 VOIP SIP Protocols (RFC3261)/Interfaces...................... 5 1.2 LAN Protocols /Interfaces..........................6 1.3 Soft Switch Interoperability ..........................6 2. ATA Overview ................................7 2.1 Ports and Buttons ............................. 7 2.2 LED Description .............................. 8 3. Installing ATA................................9 3.1 Configure the Obtain and IP Address automatically for LAN Card ..............
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4.2.4.2 Default Dial Plan Rule Nx.5t8xt2>#..................85 4.2.4.3 Default Dial Plan Rule 1Nx.2Nx.5tfxt2>#................86 4.2.4.4 Default Dial Plan Rule 011x>#x.et8xt2 ................... 86 4.2.4.5 Default Dial Plan Rule 1:*72;>#x.etfxt2 ................. 87 4.2.4.6 Default Dial Plan Rule 3:*74;>#x.etfxt2 ................. 88 4.2.4.7 Default Dial Plan Rule 4:*75;>#x.etfxt2 ................. 89 4.2.4.8 Default Dial Plan Rule 2:*73;>#t4...................
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C.1 vophwcfg.ini - Configuring PSTN Backup Support ................... 118 C.1.1 Configuration for CX82310 Based System ..................118 C.1.2 Configuration for 1-Line Reference Hardware ................119 C.1.3 Configuration (2FXS-1FXO) Based Reference Hardware............... 120 Appendix D Ring Tone Descriptions ......................... 121 For Hungarian user manual please visit our website: www.concorde.hu/drvpdf/02-05-405.pdf...
1. Features 1.1 VOIP SIP Protocols (RFC3261)/Interfaces · Single voice port that supports legacy (analog) touch-tone telephones · Connects legacy telephones to IP-based networks · Advanced pre-processing to optimize full-duplex voice compression · High performance line-echo cancellation eliminates noise and echo ·...
2. ATA Overview ATA has many ports, switches and LEDs. ATA may have some or all of the features listed below 2.1 Ports and Buttons 1WAN + 4 LAN + 2 FXS+1 FXO POWER: Connect the power adapter that came with the ATA. Using a power supply with a different voltage rating will damage this product.
2.2 LED Description Power LED: The LED stays lighted to indicate the system is power on properly. SIP1/SIP2 LED: This LED is lighted when the ATA is REGISTERED successfully to the SIP Server. WAN LED: The LED is lighted when a connection is established to WAN port and flashes when WAN port is sending/receiving data.
3. Installing ATA 1. Locate an optimum location for the ATA. 2. For connections to the Ethernet interfaces, refer to figure below. 3. Connect the AC Power Adapter. Depending upon the type of network, you may want to put the power supply on an uninterruptible supply. Only use the power adapter supplied with the ATA.
3.2 Easy Setup For easy advanced configuration, insert the CD into your CD-ROM drive. The CD should auto-start and then click “Easy Setup”. If it does not start, click on Start -> Run and type in CD:\Easysetup\vbpES.exe (where CD is the drive letter of your CD-ROM drive.) There are Three options of Protocol Modes: DHCP Client, PPPoE Client and Static IP Setting Mode.
3.2.1 DHCP Client Mode 1. After selecting the Protocol mode: DHCP Client Mode 2. Enter Registrar Address and Port / Proxy Address and Port / Outbound Proxy Address and Port / Auth User ID / Password in VoIP Line 0 and VoIP Line 1 which was given by Telecom or by your Internet Service Provider (ISP).
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REGISTERED successfully to the SIP Server. If not, please press reset button and reconfigure configuration again. 7. Now you are ready to use the ATA !!!
3.2.2 PPPoE Client Mode 1. After selecting the Protocol mode: PPPoE Client Mode 2. Enter Username / Password / Registrar Address and Port / Proxy Address and Port / Outbound Proxy Address and Port / Auth User ID / Password in VoIP Line 0 and VoIP Line 1 which was given by Telecom or by your Internet Service Provider (ISP).
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REGISTERED successfully to the SIP Server. If not, please press reset button and reconfigure again. 7. Now you are ready to use the ATA !!!
3.2.3 Static IP Settings Mode 1. After selecting the Protocol mode: Static IP Settings Mode 2. Enter IP Address / Subnet Mask / Gateway / DNS Server Url Name / DNS Server IP / Registrar Address and Port / Proxy Address and Port / Outbound Proxy Address and Port / Auth User ID / Password in VoIP Line 0 and VoIP Line 1 which was given by Telecom or by your Internet Service Provider (ISP).
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REGISTERED successfully to the SIP Server. If not, please press reset button and reconfigure again. 7. Now you are ready to use the ATA !!!
3.3 Basic VoIP Configuration 3.3.1 Access to the web configuration of ATA Step 1: 1. Launch the Web browser (Internet Explorer, Netscape, etc.). 2. Enter the LAN port default IP address (default gateway) http://12.0.0.2 in the address bar. Entry of the username and password will be prompted. Enter the default login User Name and Password: The default login User Name of the administrator is admin, and the default login Password is epicrouter.
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Step 2: Now you could configure the ATA in detail.
3.3.2 VoIP Configuration Step 1: Click " VoIP " click Line to Confiure (if you have this option) and then click " Update Service Provider "...
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Step 2: Enter the information of "New Service Provider / Registrar Address / Registrar Port / Proxy Address / Proxy Port / OutboundProxy Address / OutboundProxy Port " , select the Service Provider Action to "ADD NEW SP" and then click "Submit Changes -> VoIP ".
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Step 3: Select the Service Provider which you configured and then click " Update User Login Account".
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Step 4: Enter the "New Account Name / User ID / Password / Auth User ID / Display Name" which you applied to the SIP Server, select the Login Action to "ADD " User and then click "Submit Changes -> VoIP ".
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Step 5: Select the Login Account which you configured and then click " Submit Changes -> Save Configuration".
3.3.3 WAN Configuration 3.3.3.1 Static IP Configuration Step 1: Click " WAN " and then enter the " IP Address / Subnet Mask / Gateway " in Static IP Settings Mode and then click " Submit "...
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Step 2: Click " DNS " and then check User Configuration, enter " DNS Server -> Add / DNS Server -> Enabled / Url Name / Host IP -> Add -> Apply "...
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Step 5: Your settings are being saved and the modem being rebooted. Save-reboot in progress, please wait…...
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Step 6: Your settings are being saved and the modem being rebooted. Done. Please check the SIP LED is lighted or not. If the SIP LED is lighted, the ATA is REGISTERED successfully to the SIP Server. If not, please press reset button and reconfigure configuration again.
3.3.3.2 DHCP Client Mode Configuration Step 1: 1. Launch the Web browser (Internet Explorer, Netscape, etc.). 2. Enter the LAN port default IP address (default gateway) http://12.0.0.2 in the address bar. Entry of the username and password will be prompted. Enter the default login User Name and Password: The default login User Name of the administrator is admin, and the default login Password is epicrouter.
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Step 2: Click " WAN " and then Select " DHCP Client Enable " in DHCP Client Mode and then click " Submit -> Save Configuration"...
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Step 4: Your settings are being saved and the modem being rebooted. Save-reboot in progress, please wait…...
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Step 5: Your settings are being saved and the modem being rebooted. Done. Please check the SIP LED is lighted or not. If the SIP LED is lighted, the ATA is REGISTERED successfully to the SIP Server. If not, please press reset button and reconfigure configuration again.
3.3.3.3 PPPoE Client Mode Configuration Step 1: 1. Launch the Web browser (Internet Explorer, Netscape, etc.). 2. Enter the LAN port default IP address (default gateway) http://12.0.0.2 in the address bar. Entry of the username and password will be prompted. Enter the default login User Name and Password: The default login User Name of the administrator is admin, and the default login Password is epicrouter.
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Step 2: Click " WAN ", check Enable, enter Username and Password, check Automatic Reconnect in PPPoE Client Mode and then click " Submit -> Save Configuration"...
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Step 4: Your settings are being saved and the modem being rebooted. Save-reboot in progress, please wait…...
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Step 5: Your settings are being saved and the modem being rebooted. Done. Please check the SIP LED is lighted or not. If the SIP LED is lighted, the ATA is REGISTERED successfully to the SIP Server. If not, please press reset button and reconfigure configuration again.
4. Advanced VoIP Configuration The ATA is configured using the web interface. The ATA Configuration page can be reached as follows: 1. Launch the Web browser (Internet Explorer, Netscape, etc.). 2. Enter the LAN port default IP address (default gateway) http://12.0.0.2 in the address bar.
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4. On the router Home Page, click the VoIP link on the left frame to view the ATA Configuration page. In general, configuration changes made using the web interface will be activated only upon clicking Save & Reboot button on the Save Savings / Reboot page. Note: Certain Voice Parameters do not require a Save &...
4.1 ATA Configuration Page The ATA Configuration page sets parameters for the VoIP application. The ATA Configuration page is divided into three general categories: Version Details, Line Based Config, and Non-Line Config. Version Details: This section displays the current versions of the ATAA and PTM software. Line Based Config: This section configures parameters for the selected line.
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Note: Disabling SIP Registration means incoming calls cannot be received and outgoing calls through the proxy also cannot be made; this is done to enable back-to-back direct User Agent (UA) calling. Transport Type: Select the transport to use for SIP signaling, UDP or TCP. Service Provider To Use: Select the service provider to work with the ATA for the selected line.
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Current Voice Call Status: This field displays the current registration status of t he selected line as defined in the following table: Current Voice Call Status Shown Condition NO VOICE CALL IN PROGRESS No voice calls on the selected endpoint currently VOICE CALL IN PROGRESS...
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Country Specific Ring & Tone Parameters: Click this link to display the C ountry Specific Ring & Tones configuration page to define parameters for various tones. See configuration details in Section 4.1.1. General Parameters: Click this link to display the General configuration page for non-line system-level parameters.
4.1.1 Country Specific Ring & Tones Configuration The Country Specific Ring & Tones configuration page defines parameters for the various tones (ring, dial, busy, ring back, etc.) generated by the ATA application. ATA provides default ring and tone parameters configured for operation in the USA. Flexibility is provided to change the existing ring-tone parameters, and to add new countries as well as to edit/delete existing countries.
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Ring Parameters Ring: Enter five consecutive fields separated by commas: Frequency, OnTime1, OffTime1, OnTime2, and OffTime2. Tone Parameters PSTN Dial Tone: Enter six consecutive fields separated by commas: Freq1, Amp1, Freq2, Amp2, OnTime1, OffTime1, OnTime2, and OffTime2. The PSTN is used when WAN is disconnected or not-operative.
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Unobtainable Tone: Enter six consecutive fields separated by commas: Freq1, Amp1, Freq2, Amp2, OnTime1, OffTime1, OnTime2, and OffTime2. Recall Tone: Enter five consecutive fields separated by commas: Freq1, Amp1, Freq2, Amp2, and Stutter Duration. Stutter Dial Tone: Enter five consecutive fields separated by commas: Freq1, Amp1, Freq2, Amp2, and Stutter Duration.
4.1.2 General Configuration Page The General configuration page configures system-level parameters not related to the selected line. This has five sections: SIP Device, VoIP General, BLAM Server, STUN Parameters, and Default Dial Plan Parameters. SIP Device: This section configures the following information: ••...
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VoIP General: This section configures the following information: Enable Auto Login: If enabled, the system will obtain the login information from the service provider using the MAC address. Note: When enabled and successful in obtaining the login information, no VoIP web pages will be displayed.
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• RTrace fifo size: Enter the Rtrace FIFO size in KB. A non-zero value is r equired for starting the BLAM server. The maximum value is 100. Enter 0 for stopping the BLAM server. •RDump fifo size: Enter the Rdump FIFO size in KB. A non-zero value is r equired for starting the BLAM server.
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configured for the system (see Section 4.2.4 for more details on the default value for this field). Edit the dial plan as required. When a new service provider is added, the initial dial plan string for the service provider is taken from this default dial plan string.
4.1.3 SIP Service Provider Configuration Page The SIP Service Provider configuration page sets the configuration related to the SIP service provider. Service Provider List: Enter the name of the service provider to be configured. When a service provider is selected from this drop-down list, the respective parameters are automatically displayed.
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Registration Interval (in secs): Enter the re-registration interval in seconds. The range is 0 to 2147483347 seconds. The default is 3600 seconds. Authentication Method: Select the authentication method. Only MD5 is supported. • AUTH_NONE: Disable any authentication method • AUTH_MD5: Use MD5 authentication method. Registrar Address: Enter the IP address or Domain Name of the registrar with which the ATA must register in order to receive or send calls.
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List field. This is the default selection. • ADD NEW SP: Select ADD NEW SP to add a new service provider after c licking Submit Changes according to the value that appears in the New Service Provider field. This field must not be empty.
4.1.4 User Login Account Configuration Page The User Login Account configuration page sets and configures login accounts for the service provider chosen in the index web page, i.e., for the currently selected service provider in the main web page. Service Provider Name: This field displays the service provider for which the login information is being configured.
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User ID: Enter the registration ID of the user with the registrar. Password: Enter the password used for authentication with the registrar. Auth User ID: Enter the Authorization User ID for authentication with the registrar. If not specified explicitly by the service provider, this is same as the User ID. Display Name: Enter the Display Name as it should appear on Caller ID.
4.1.5 Timer Parameters Configuration Page The Timer Parameters configuration page displays and configures timers used at the system level. This section explains the various timers available for configuration. These timers are not applicable for PSTN calls. Selected Endpoint: The endpoint for which these timer parameters are applicable. This corresponds to the current selected endpoint on the ATA Configuration webpage.
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Call Waiting Timeout (Sec): Enter the length of time in seconds the Call Waiting tone will be played when an incoming call arrives in the connected/held state. The Call Waiting tone is played at an interval of 10 sec. for USA. It is configurable using the Call Waiting tone parameters.
4.1.6 Call Feature Configuration Page The Call Feature configuration page displays and modifies call features. Enabling any of these options allows the user to apply the appropriate configuration as specified in the dial plan by using the dial-pad. Selected Endpoint: The endpoint for which these call features parameters are applicable. This corresponds to the current selected endpoint on the ATA Configuration webpage.
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No Answer. When enabled and the incoming call is not answered within a time limit, the incoming call will be forwarded to the entered Call 5.3.8. The Forwarding Number when using the dial-pad as described in Section Call Forward No Ans Timeout timer is configured as described in Section 4.1.5.
4.1.7 Address Book Configuration Page The Address Book Configuration Webpage allows for configuration of address book entries which can be used for speed dial execution of calls. The top half of the webpage displays the current address book table for the selected EndPoint.
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Number: The user phone number or name for this entry. This field is optional, if the IP Address is specified. A phone number that can be reached through the current configured proxy server can also be added as an entry in which case the IP address/Domain name and the Port number fields are not necessary.
4.1.8 Advanced Telephony Settings Page The Advanced Telephony Settings page allows configuration of the various endpoint level telephony parameters. Selected Endpoint: The endpoint for which these call advanaced telephony settings are applicable. This corresponds to the current selected endpoint on the ATA Configuration webpage.
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• Enable VAD and SUPPRESS SID: Click this option button to enable VAD w ith discontinuous transmission. When silence is detected, transmission of packets will be paused until voice is again detected. Echo Cancellation: This section configures Echo Cancellation with NLP or with CNG_NLP options.
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Debounce Times: This section configures two debounce delays. • Debounce On-Off Time (ms): Enter the delay, in ms, before informing on an o ff-hook event to ensure it is not a spike. If an on-hook event is received during this delay time, the two events will be ignored.
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Note: After clicking Submit Changes to save page settings to system RAM, you must permanently save the configuration and reinitialize the system as follows. Save Configuration: Click this link to go to the Save Settings and Reboot page. On the Save Settings and Reboot page, click Save &...
4.1.9 Advanced System Telephony Parameters Page The Advanced System Level Telephony Parameters page allows configuration of the system level telephony parameters for Caller ID and Jitter Buffer. Codec Prefereces: This section configures the frames per RTP packet for the selected codec.
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• Max accept late seq num: If the new packet sequence number is low by “Max Accept late seq num” from the reorder buffer tail sequence number, then a new sequence has started and the reorder buffer is initialized. • Initial delay (ms): Sets the initial delay for the jitter buffer. •...
4.1.10 FXO Configuration Parameters Page The FXO Configuration page allows configuration of system level FXO parameters. See Section 5.5 for more details on FXO support. If FXO support is enabled the following parameters are displayed. Attach FXO to FXS: This field determines which FXS is to be attached to the FXO line. Outgoing and incoming PSTN/FXO calls can be made and received on the attached FXS only.
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Note: PSTN Parameters are only displayed when the vophwcfg.ini is appropriately configured (see Appendix C.1). If FXO support is disabled and the hardware does not have a FXO, no parameters will be displayed. Submit Changes: Click Submit Changes to save the settings on this page to system RAM.
4.2 Building Dial Plan String 4.2.1 Syntactic Format for Dial Plan Strings The dial plan string consists of a set of dial plan rules specified according to the following syntax: DTMF digits 0–9, A–F, *, and #. Range and sub-range of digits (“[ ]”). For example, [135] specifies digits 1, 3, or 5, [5–9] specifies digits 5 to 9, and [125–8] specifies digits 1, 2, and 5 to 8.
4.2.2 Dial Plan String and Serial Codes This data defines the dial plan string as entered or configured by the user or the service provider. The user or service provider enters the dial string consisting of supported call-related dial patterns and the star service dial patterns. The call patterns can generally be entered in the order preferred by the user or service provider.
4.2.3 Framing Dial Plan Rules Dial plan interpretation and parsing is based on the incoming character and a set of expected characters as follows: 0–9, A–D, #, or * A DTMF digit is recognized as valid if it is one of the following: between 0–9 or A or B or C or D or # or *.
4.2.4 Default Dial Plan String The default dial plan string corresponds to the North American Numbering Plan (NANP). This default dial plan string is (where each rule is separated by the separator “|”):0>#t4|Nx.5t8xt2>#|011x>#x.et8xt2|1Nx.2Nx.5t8xt2>#|1:*72;>#x.etfxt2|2:*73;>#t4|3:*74; >#x.etfxt2|4:*75;>#x.etfxt2|11:*70;>#t4|12:*69;>#t4|16:*90;x>#x.dtfxt2| 18:*47;x>#[0-9*].f[0-9*].ft8 [0-9*].ft4|19:*78;x>#t4|20:#;x.3>#x.atfxt2|22:*83;x>#x.dtfxt2|23:*76;>#t4|24:*77;>#t4|25:N1 t41;># |26:*67;>#t4| [0-9*]>#[0-9*].e[0-9*].ft4 The individual default dial plan rules are summarized in Table 2-2. Table 2-2.
Caller ID Transmission [0-9*]>#[0-9*].e[0-9*].ft4 Default number call 4.2.4.20 There are three major components of any star service rule: 1. The star service rule must begin with the serial code identified in Table 2-1. This serial code identifies the feature/dial plan action to be executed. This serial code is followed by the separator “:”.
4.2.4.3 Default Dial Plan Rule 1Nx.2Nx.5tfxt2># This is the dial plan rule for dialing a long distance call. Eleven digits must be dialed starting with 1, each digit within 15 seconds of the preceding digit, optionally followed by pressing #. After the 11 digits are dialed, press # to send the digits immediately, otherwise there is a 2-second delay before the digits are sent.
4.2.4.5 Default Dial Plan Rule 1:*72;>#x.etfxt2 This is the dial plan rule for Enable Unconditional Call Forwarding star service. Digits *72 must be dialed, followed by 15 optional digits and the optional # suffix. Each digit must be dialed within 15 seconds of the preceding digit. If only the *72# digits are dialed or the IDT elapses before dialing any digits after *72, the default number configured on the Call Feature configuration page (Section 4.1.6) is sent.
4.2.4.6 Default Dial Plan Rule 3:*74;>#x.etfxt2 This is the dial plan rule for Enable Unconditional Call Forwarding on Busy star service. Digits *74 must be dialed, followed by 15 optional digits and the optional # suffix. Each digit must be dialed within 15 seconds of the preceding digit. If only the *74# digits are dialed or the IDT elapses before dialing any digits after *74, the default number configured on the Call Feature configuration page (Section 4.1.6) is sent.
4.2.4.7 Default Dial Plan Rule 4:*75;>#x.etfxt2 This is the dial plan rule for Enable Unconditional Call Forwarding on No Answer star service. Digits *75 must be dialed, followed by 15 optional digits and the optional # suffix. Each digit must be dialed within 15 seconds of the preceding digit. If only the *75# digits are dialed or the IDT elapses before dialing any digits after *75, the default number configured on the Call Feature configuration page (Section 4.1.6) is sent.
4.2.4.9 Default Dial Plan Rule 11:*70;>#t4 This is the dial plan rule for Temporary Disable Call Waiting star service. Digits *70 must be dialed, each digit within 4 seconds of the preceding digit, optionally followed by pressing #. After the*70 digits are dialed, press # to send the digits immediately, otherwise there is a 4-second delay before the digits are sent.
4.2.4.11 Default Dial Plan Rule 16:*90;x>#x.dtfxt2 This is the dial plan rule for Blind Transfer star service. Digits *90 followed by a single-digit number must be dialed, each digit within 15 seconds of the preceding digit, followed by 14 optional digits and the optional # suffix. After the *90x digits are dialed, press # to send the digits immediately, otherwise the digits will be sent 15 seconds after the last digit for digits 4-17 is dialed, or 2 seconds after digit 18 is dialed.
4.2.4.13 Default Dial Plan Rule 19:*78;x>#t4 This is the dial plan rule for speed dialing an entry in the address book. Digits *78 followed by a single digit must be dialed, each digit within 4 seconds of the preceding digit, optionally followed by pressing #. Note: The single digit is an index into the address book maintained by the system.
4.2.4.15 Default Dial Plan Rule 22:*83;>#t4 This is the dial plan rule for Disabling 3-Way Conferencing temporarily. Digits *83 must be dialed, each digit within 4 seconds of the preceding digit, optionally followed by pressing #. No number needs to be dialed. After the *83 is dialed, press # to send the rule immediately, otherwise there is a 4-second delay before the rule is executed.
4.2.4.17 Default Dial Plan Rule 24:*77;>#t4 This is the dial plan rule for Disable Call Forwarding on No Answer star service. Digits *77 must be dialed, each digit within 4 seconds of the preceding digit, optionally followed by pressing #. No number needs to be dialed. After the *77 is dialed, press # to send the rule immediately, otherwise there is a 4-second delay before the rule is executed.
4.2.4.18 Default Dial Plan Rule 25:N1t41;># This is the dial plan rule for Emergency Call. However, this rule also handles other N11 services. One digit in range 2–9 followed by the digit 1 and the digit 1 must be dialed, each digit within 4 seconds of the preceding digit.
4.2.4.19 Default Dial Plan Rule 26:*67;>#t4 This is the dial plan rule for Temporary Enable Caller ID Transmission Block. Digits *67 must be dialed, each digit within 4 seconds of the preceding digit, optionally followed by pressing #. After the*67 digits are dialed, press # to send the digits immediately, otherwise there is a 4-second delay before the digits are sent.
4.2.5 Empty Dial Plan String If a default dial plan string is not required, the Default Dial Plan String field on the General configuration page (Section 4.1.2) can be left empty, in which case the default dial pattern to accept all dialed digits will be incorporated. The default dial pattern, [0-9*]>#[0-9*].e[0-9*].ft4, is transparent to the user and will not be displayed on the General configuration page (see Section 4.1.2).
5. Using Conexant ATA 5.1 Setting up ATA for VoIP Calls This section describes the setup and configuration needed before the ATA can be used. 1. Ensure the ATA software has been flashed onto the VoIP reference board. 2. When power is applied to the board, the system initializes with the default configuration which is not initially set up to make VoIP calls.
5.3 Advanced Call Features The supplementary services described in this section, and their configuration and implementation, depend on the system of the country in which the service is activated. This section describes the following topics: • Caller ID • Call-Waiting Caller ID •...
5.3.2 Call-Waiting Caller ID The ATA plays the first Call Waiting tone, and then sends the name, telephone number, time, and date information, if these are available, before the second Call Waiting tone is played. 5.3.3 Consultation Hold This feature allows a user to put the existing call on hold and call another number. How to Use: 1.
5.3.5 Attended Transfer This feature allows a user to transfer an existing call to another telephone number after first consulting with the dialed party (transfer target) before hanging up. How to Use: 1. During an existing call, perform a hook flash to put the other party on hold and get a dial tone.
An alternate way to temporarily disable 3-way conferencing is as follows: 1. During connection to the first party, perform a hook flash to put the first party on hold and get a dial tone. 2. If the ATA is configured to use 3-way Call Conferencing (see Section 4.1.3), press *83 (optionally followed by #) on your telephone dial-pad to disable Call Conferencing for the duration of the following call.
5.3.8 Call Forwarding in USA The ATA can control Call Forwarding at the end-point level. The ATA supports three types of Call Forwarding: • Call Forwarding Unconditionally—Forwards every incoming call. • Call Forwarding On Busy—Forwards incoming calls only when the line is busy. •...
web configured number and you will hear a very brief confirm tone (default is 300 milliseconds only), followed by the normal dial tone. 5.3.8.3 Call Forwarding On No Answer How to Use: 1. Press *75 on your telephone dial-pad. 2. After you hear a very brief alert tone (default is 100 milliseconds), enter the number (optional) you want to forward the call to;...
5.3.9 Call Return in USA The ATA provides the facility to call back the last incoming call that may have been missed. This is especially useful when the phone does not have Caller ID facility or does not support Call-Waiting Caller ID. How to Use: Press *69 (optionally followed by #) on your telephone dial-pad to return the last incoming call.
Note: When the sequence of digits *47 is entered, the sequence could match the IP Dialing rule or the Default Number Call rule (Section 4.2.4.19). So, to execute the command immediately, press #. Otherwise, the action will execute after 4 seconds. 5.3.11 Speed Dialing for Address Book Entries The ATA provides the facility of speed dialing for entries configured in the address book.
5.4 PSTN Backup (Failsafe Relay Mode) In case of power failure, the WAN IP is down, or SIP registration with the SIP server is lost, the ATA will automatically switch to Failsafe Relay (Hardware Relay) mode. In this case, all calls will be directed to the PSTN CO and handled by the CO.
5.5.1 Incoming PSTN Call: There are two cases of handling a new PSTN call: CID Rx enabled: The ATA checks the FXO line state. If available, start ringing or start playing the Call Waiting tone to that Endpoint. The ATA then send the CID information to the phone if obtained.
6. Admin Privilege The links under Admin Privilege are only accessible when user is logged in as Admin. Regular user account does not have authorization to view or alter the content on the pages in the Admin Privilege section. 6.1 Miscellaneous Configuration The Miscellaneous Configuration page allows you to set miscellaneous configurations for the following: HTTP, FTP, TFTP, DMZ, Command Line Interface, DHCP, PPP, IGMP, and SNTP.
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default is Enabled. • Disable WAN side FTP access: This will disable WAN side access to the FTP server, default is Disabled. TFTP server: This field allows you to enable or disable the TFTP connection. System default is Disabled. PPP reconnect on WAN access: If enabled, the PPP session will automatically establish a connection when a packet tries to access the WAN.
6.2 System Log The System Log page shows the events triggered by the system. This page contains information that is dynamic and will refresh every 5 seconds. Clear Log: This field allows you to clear the current contents of the System Log. Save Log: This field allows you to save the current contents of the System Log by right click HERE and select “Save Target As”...
6.3 Admin Level Username / Password Configuration The Admin Password Configuration page allows you to set the password for administrator. The Admin password is same as the FTP password, so it must have at least 8-characters for the FTP to work. The Admin password can be up to 65 characters (excluding ‘&’).
6.4 Local Code Image Update The Code Image Update page allows you to upgrade the image code locally. Select Image Download to start a Code Image Update. After Image Download is selected it will take a few seconds before you can select the file to be downloaded. Browse the location of file, firmware.dlf or bootrom.dlf file, and click the Upload to start the update.
Appendix A Glossary This glossary defines acronyms and keywords used in this document. A.1 Acronyms Analog Telephony Adaptor BLAM Background Logging Application Mechanism FoIP Fax over Internet Protocol Foreign Exchange Office Foreign Exchange Station PSTN Public Switched Telephone Network Packet Telephony Module RTP Real-time Transport Protocol Session Initiation Protocol STUN Simple Transversal of UDP through NAT...
Appendix B Dial Plan for Pulver Service Provider B.1 Basic Dial Plan An example dial plan string as would be used with Pulver (a globally available VoIP service provider - http://www.fwd.pulver.com) is given below. The dial plan string represents only the basic dialing call and service rules, which can be used with the VoIP solution.
B.1.2 Explanation of the Rules Dial Plan Rule Explanation of the Rules [1-9]x.2t8>#x.6t4 Call any Pulver Number up to 9 digits, but minimum 3 digits *18x.8t8xt2># Call any Toll Free Number through Pulver Account **484x.7t4># Call any Global Village Number. This rule demands a minimum and a maximum or 7 digits following **484 1:*72;>#x.etfxt2 Enable Unconditional Call Forwarding...
B.2 Calling Other Service Provider Numbers through Pulver Pulver also provides the facility to call other service provider numbers. The following is an example list of those service providers and the corresponding recommended dial plan rules that need to be configured for each of them. Please check their website for the complete list of supported service providers: (http://www.fwd.pulver.com/content/view/full/333/ Service Provider Partner...
Appendix C Configuring *.ini Files C.1 vophwcfg.ini - Configuring PSTN Backup Support This section explains how to configure the vophwcfg.ini for enabling and disabling PSTN support for a CX82310-based system and the 1-Line reference board. C.1.1 Configuration for CX82310 Based System The relevant fields in the vophwcfg.ini are: HwPstnRelayCtrlGpio = 2 Indicates the GPIO number that controls the PSTN backup (PSTN hardware relay) (-1 if...
C.1.2 Configuration for 1-Line Reference Hardware The relevant fields in the vophwcfg.ini are: HwPstnRelayCtrlGpio = 20 Indicates the GPIO number that controls the PSTN backup (PSTN hardware relay) (-1 if hardware support is not present). HwPstnRelayPstnVal = 1 Indicates the value that should be written to HwPstnRelayCtrlGpio in order to switch to PSTN.
C.1.3 Configuration (2FXS-1FXO) Based Reference Hardware The relevant fields in the vophwcfg.ini are: HwPstnRelayCtrlGpio = 18 Indicates the GPIO number that controls the PSTN backup (PSTN hardware relay) (-1 if hardware support is not present). HwPstnRelayPstnVal = 1 Indicates the value that should be written to HwPstnRelayCtrlGpio in order to switch to PSTN.
Appendix D Ring Tone Descriptions This appendix briefly describes each ring tone supported by the Conexant ATA. PSTN Dial Tone: This dial tone indicates there is PSTN/FXO support and the ATA has not been initialized for network and SIP signaling. It is also played when the ATA has lost network connectivity.
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Unobtainable Tone: The ATA plays this tone if the called party cannot be reached by the SIP proxy server or if the called party is not registered with the SIP proxy server. Recall Tone: When the user puts the first call on hold to dial another number, this dial tone, which is like a normal dial tone with a few stutters at the beginning, is played to indicate that a call on hold exists.
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