Linksys SPA2002-ER User Guide

Linksys SPA2002-ER - Earthlink Truevoice Phone Adpt Manual

Linksys SPA2002-ER manual content summary:

  • Linksys SPA2002-ER | User Guide - Page 1
    Sipura Technology, Inc. SPA User Guide July 2004 © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 1
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    and statements made are based on generic network, service and application requirements and should be regarded as a guide to assist you in forming your own opinions and decision regarding your particular situation. As well, Sipura Technology reserves the right to change the features and
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    Sipura Technology, Inc. SPA User Guide Table of Contents 1. Product Description ...6 1.1. SPA Hardware Overview ...6 2. Installation Overview ...7 3. Software Configuration ...8 3.1.1.1. Firmware Upgrade ...8 3.2. IVR Interface ...8 3.3. Web Interface ...11 3.3.1. Web Interface Conventions ...11
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    - PSTN Line Status ...69 4.14.20. PSTN/VoIP Caller Commands via DTMF 70 5. User Guidelines ...70 5.1. Basic Services ...71 5.1.1. Originating a Phone Call ...71 5.1.2. Receiving a Phone Call ...71 5.2. Enhanced Services ...71 5.2.1. Caller ID ...72 5.2.2. Calling Line Identification Presentation
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    Call FWD - No Answer ...81 5.2.18. Anonymous Call Blocking...82 5.2.19. Distinctive / Priority Ringing and Call Waiting Tone 82 5.2.20. Speed Calling - Up to Eight (8) Numbers or IP Addresses 83 6. Appendix I: Dial Plan ...83 © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice
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    1. Product Description This guide describes basic use of the Sipura Technology SPA phone adapter - an intelligent lowdensity Voice over IP (VoIP) gateway. The SPA enables carrier class residential and business IP Telephony services delivered over broadband or high-speed Internet connections. By
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    service supports only one incoming line, the analog telephone or fax machine should be connected to port one (1) of the SPA. Port one (1) is the outermost telephone port on the SPA and is labeled "Phone 1." The SPA-3000 has an RJ-11 interface labeled "Line" which can be used to connect the adapter
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    attain basic IP network connectivity and the more advanced browser-based configuration menu may be accessed. 1. The SPA IVR uses the following conventions: By factory default there is no password and no password authentication is prompted for all the IVR settings. If administrator password is set
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    reboot. 3. After one minute of inactivity, the unit times out. The user will need to re-enter the configuration menu from the beginning by pressing * * * *. 4. If, while entering a value (like an IP address) and you decide to exit without entering any changes, you may do so by pressing the * (star
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    IP 131 Address Check MAC Address 140 Check Firmware Version 150 Check Primary DNS 160 Server Setting Set Primary DNS Server 161 © 2003 - 2004 Sipura Technology, Inc Enter 1 to enable Enter 0 to disable Requires Password None IVR will announce the current IP address of SPA. Enter IP
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    at the IP address of the SPA; to use the administrator privilege, use URL http://IP_Address_Of_SPA/admin/. See 3.3.2 for more information about administration privileges. o Version 1.0 of the SPA supports Internet Explorer 5.5 and above and Netscape 7.0 and above. o The web configuration pages can
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    changes to SPA configuration parameters, pressing reset the values on the web page. 3.3.2. Administration Privileges The SPA supports two levels of administration privileges: Administrator and User, both privileges can be password protected. Important note: by factory default, there are no passwords
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    Only TFTP is supported in the current firmware version 1.0.9 or later. Enable/disable Admin pages of web server of SPA The password for administrator The password for User Default Yes Yes Parameter Name DHCP Host Name Domain Static IP NetMask Gateway Primary DNS Secondary DNS Network Configuration
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    IP address or name of primary NTP server. Parallel 0 IP address Reset Resync Random Delay Resync Periodic Resync Error Retry Delay Resync From SIP Resync After Upgrade Default yes yes 2 3600 3600 Yes Yes /spa.cfg See provisioning discussion section See provisioning discussion © 2003 - 2004 Sipura
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    , the profile rule would point to a service provider's server. 3.4.3. Upgrade Parameters Parameter Name Upgrade Enable Upgrade Error Retry Delay Upgrade Rule Log Upgrade Request Msg Description Log Upgrade Success Msg Log Upgrade Failure Msg Default Yes 3600 empty See provisioning discussion
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    to play the SIT1 Tone no application/dtmf-relay application/hook-flash no .5 4 5 32 32 32 32 32 180 30 1 7200 30 1200 © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 16
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    0.02 0 Notes: 1. Reorder or Busy Tone will be played by default for all unsuccessful response status code 2. and @, NAME is set to (or "Anonymous" if user blocks caller ID), and TOOL is set to the Verdor/Hardware-platform-software-version (such as Sipura
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    configured codec names. Comparison is case-insensitive. 3.4.4.3. NAT Support SIP responses if received from IP and VIA sent-by IP differ Insert rport parameter in configured STUN server. The result of the discovery will be reported in a Warning Default No No No No No No No No © 2003 - 2004 Sipura
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    in the configuration profile. It is omitted below for readability. 3.4.5.1. User Account Information Parameter Name Description Line Enable MOH Server2 Enable this line for service The User ID or URL of the auto-answering SAS to contact for MOH services. Examples: 5000, [email protected]
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    will stream RTP packets to the SAS line. The default value is [empty]. Enable the use of externally mapped of IP address and SIP/RTP ports in SIP messages. The mapping may be discovered by any of the supported methods. If set to "yes", the configured is sent periodically every
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    is slower (more conservative), else faster (more aggressive). Enable the support or the 100rel SIP extension for reliable transmission of provisional responses (18x name to appear in caller-id No No No No Yes Yes No No 3600 No No 3600 © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright
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    attribute and the remote destination address and port must not be 0 3. SAS Notes: • Either or both of lines 1 and 2 can be configured as an SAS server. Blocking, DND, and Caller-ID Delivery features are not available on an SAS line. 3.4.5.2. Supplementary Services Enablement © 2003 - 2004 Sipura
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    Cfwd All Serv Cfwd Busy Serv Description Enable Call Waiting Service Enable Block Caller ID Service Enable Block Anonymous Calls Service Enable Distinctive Ringing Service Enable Call Forward All Service Enable Call Forward Busy Service Default Yes Yes Yes Yes Yes Yes Cfwd No Ans Serv Enable
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    Adapt Enable Echo Supp Enable G729a Enable1 G723 Enable1 G726-16 Enable1 G726-24 Enable1 G726-32 Enable1 G726-40 Enable1 FAX Passthru Enable FAX CED Detect Enable FAX CNG Detect Enable FAX Passthru Codec FAX Codec Symmetric FAX Passthru Method FAX not support this codec > paramter Default G711u No
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    to disable the use of G.729a in order to guarantee the support of 2 simultaneous G.723/G.726 codec. 3.4.5.4. Dial Plan See section 6 for additional information regarding the configuration of the SPA dial plan. Parameter Name Dial Plan Enable IP Dialing Description Per-line dial plan script Enable
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    8 ID of caller blocked via the "Block Last Caller" service ID of caller accepted via the "Accept Last Caller" service The Caller number that is actively forwarded to by using the Call Forward Last activation code Forward number for the Default 20 3.4.6.2. Speed
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    Waiting on/off for all calls Block Caller ID on/off for all calls Block Anonymous Calls on or off DND on or off Caller ID Generation on or off Call Waiting Caller ID Generation on or off Distinctive Ring on or off If yes, all outbound calls are secure calls by default Default Yes No No No Yes Yes
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    to trigger this tone is configurable (see Section ???) See all is activated Default 350@-19,440 20(.380/0/1,.380 /0/2,.380/0/3,0/4/0) 985@-16,1371@16,1777@16;20(.380/0/1,.274 /0/2,.380/0/3,0/4/0) 350@-19,440@19;2(.1/.1/1+2);10(* /0/1+2) 350@-19,440@19;2(.2/.2/1+2);10(* /0/1+2) © 2003 - 2004 Sipura
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    pick distinctive ring/CWT 5 for the inbound call Default 60(2/4)" 60(.3/.2, 1/.2,.3/4)" 60(.8/.4,.8/4) 60(.4/.2,.3/.2,.8/4) 60(.4/.2,.3/.2,.8/4) 60 Bellcore-r1 Bellcore-r2 Bellcore-r3 Bellcore-r4 Bellcore-r5 © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 29
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    entering digits when dialing. Range: 0 - 64 sec Delay in seconds after caller hangs up when the SPA will start removing the tip-and-ring voltage to for which the tip-to-ring Default 0.1 0.9 0 5 1800 30 0.5 0.5 10 3 2 0 (CPC © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice
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    should be used instead. 4. Without CPC enabled, reorder tone will is played after a configurable delay. If CPC is enabled, dial tone will be played when tip-to-ring voltage is restored. 3.4.7.4. Vertical Service Code Assignment Parameter Name Call Return Code Blind Transfer Code Cfwd All Act Code
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    Unblock CID on the next inbound call *82 Block_ANC_Act_Code Block all anonymous calls *77 Block_ANC_Deact_Code Unblock all anonymous calls *87 DND_Act_Code Enable Do Not Disturb *78 DND_Deact_Code Disable Do Not Disturb *79 CID_Act_Code Enable Caller-ID Generation
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    Services Codes Notes: processed by the SPA. You can empty the corresponding *code that you do not want to SPA to process. One or more *code can be configured BLock Caller ID (*67). Notes: - The *codes should not conflict with any of the other vertical service plays Prompt tone by default. - If the *
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    be 2-digit or 4-digit Setting the local time; second is optional. Number of hours to add to GMT to form local time for caller-id generation. Choices: GMT12:00, GMT-11:00,..., GMT, GMT+01:00, Default GMT-07:00 © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 34
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    Detect ABCD Playback ABCD Caller ID Method FXS Port Power Limit Protect IVR Factory Reset GMT+02:00, - (c). All CID methods can be applied for this type of caller-id • On Hook Caller ID Not Associated with caller-id methods: "Bellcore", "ETSI FSK", and "ETSI FSK With PR" © 2003 - 2004 Sipura Technology,
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    caller ID (CIDCW) or to notify the user that the far end party identity has changed or updated (such as due to a call transfer). This is only available if the caller-id last reboot. E.g., 25 days and 18:12:36 Total number of broadcast packets IP address used for NAT mapping © 2003 - 2004 Sipura
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    Caller Number Mapped SIP Port Call 1/2 Status State Tone Encoder Decoder FAX last number called The number of the last caller NAT Mapped SIP Port State of the call /32/40, G729a, or G729ab Indicate whether FAX pass-through mode has been initiated: Yes or if the peer supports RTCP Total number of packets
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    VoIP to obtain PSTN service. - VoIP User - a VoIP Caller which has a user account (user-id and password) on the SPA-3000 - PSTN Caller - one who calls the SPA-3000 from the PSTN to obtain VoIP service. Two VoIP services can be configured control of SPA firmware). Before power after 10-20s to indicate
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    invoke the IVR (voice configuration menu), the PHONE Port Relay must be closed. 4.2. SPA-3000 Voice Configuration Organization block diagram of the voice interfaces and the group of configuration parameters for both VoIP interfaces © 2003 - 2004 Sipura Technology, Inc Proprietary (See
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    and receive gains • DTMF playback level and timing • Caller-ID delivery signal format • Voice-Mail messages waiting indication • Impedance • On/off hook and hook flash detection timing • FAX tones detection © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 40
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    port on-hook To get the most out of your SPA-3000, it is highly recommended to connect a PSTN service to the LINE port with Type I Caller-ID subscription. This will allow the following additional functionalities: © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 41
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    /LINE port. At present the SPA can only decode Bellcore FSK style Caller-ID signal sent by the PSTN switch. 4.2.3. VoIP Interfaces There are 2 VoIP interfaces in the SPA-3000. Each VoIP interface can be configured to register with a VoIP Service Provider (VSP), and to receive and make calls over the
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    the loop back address 127.0.0.1. Hence VoIP caller requesting PSTN gateway functions will be turned down with a "Service Not Available" response. 4.3. Gateway Call Restriction by Dial Plan Gateway calls can be restricted on a per caller basis using dial plans. Up to 8 dial plans can be configured
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    ?.4. If the list is not blank and the source IP address of the inbound INVITE matches any of the patterns in the list, the VoIP caller will be granted access to the PSTN gateway as if the Authentication Method is © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 44
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    that the PSTN line connected to the SPA comes with the caller-id delivery service. If Caller-ID signal is not present or the Caller-ID number is blocked, the SPA treats the PSTN Caller-ID as "Anonymous". Each Caller-ID number is a case-insensitive alphanumeric string with special wildcard characters
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    #4) In order to obtain PSTN services via VoIP, the VoIP caller must establish connection with the SPA3000 by way of a standard SIP INVITE request addressed to the SIP account configured under the [PSTN Line] tab. The PSTN gateway can be configured to support 1-stage and 2-stage dialing as described
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    code for accepting a VoIP caller for PSTN gateway service. If (VoIP Caller-ID Pattern Blank or VoIP Caller-ID Matches a VoIP Caller-ID Pattern) { If (VoIP Caller in VoIP Access-List or Authentication Disabled) { Reply 200 Start PSTN Gateway Service } Else { © 2003 - 2004 Sipura Technology, Inc
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    PSTN Gateway Service } Else { Send BYE } } } } Else { Reply 403 } The configuration parameters mentioned in this section are: - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line]
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    and blocking functions are configured under [PSTN User]. SPA does not authenticate the PSTN caller if he is configured to be forwarded. Below is the pseudo code for accepting a PSTN caller for VoIP gateway service. If (PSTN Caller ID Pattern Blank or PSTN Caller-ID Matches a PSTN Caller-ID Pattern
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    configuration parameters mentioned in this section are: - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line] - [PSTN Line]
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    1 VoIP Outbound Call Routing (Call Type #7) The voice path for this call type is (1)Æ(2)Æ(4)Æ(6)Æ(7). Calls made from Line 1 are routed through the configured Line 1 service provider by default. This behavior can be overridden by IP dialing where the calls can be routed to any IP address entered
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    number, SPA prepends it with 1408 and routes the call to pstn.sipura.com:5061, with user-id = joe, and pwd = bell_pwd, and NAT mapping enabled User line. When Line 1 VoIP service is down (due to registration failure or loss of Ethernet link), SPA can be configured to automatically route all outbound
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    will be encoded with the VoIP account information configured for the [PSTN Line] instead of the PSTN Caller-ID name and number. Note that does not apply when ringing thru Line 1. When the PSTN call rings thru Line 1, you can assign a different under the
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    decoded PSTN caller-id name and number into the SIP INVITE message sent to the VoIP1 interface, it also prepends the name and number by the prefixes configured in and respectively. If the PSTN caller hangs up before Line 1 or the VoIP gateway answers
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    ] = yes [PSTN Line] = userid_B [PSTN Line] = password_B [PSTN Line] = yes [PSTN Line]
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    Ringn Cadence> = (default), n = 1 to 8 Assume the PSTN line has Type I Caller-ID service and the caller's name is "Joe Smith" and the caller's number is 14089991234. a) Line 1 idle at the time the PSTN rings When the PSTN line rings, Line 1 rings also. If Line 1 is picked up before the VoIP gateway
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    service, the caller will get dead air after the SPA takes the FXO port off-hook. 4.13.6. VoIP-to-PSTN Call via VoIP2 Interface With HTTP Digest Authentication: Assume the PSTN line is available and in service and the VoIP2 interfaces are properly configured. Setup: © 2003 - 2004 Sipura Technology
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    - [PSTN Line] = HTTP Digest - [PSTN Line] = jdole - [PSTN Line] = (blank), n = 2 to 8 - [PSTN Line] = silicon-valley - [PSTN Line] = 3 - [PSTN Line] = (1408xxxxxxx|1510xxxxxxx|
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    Caller-ID pattern. For example: - [User 1] = gw0 - [User 1] = 1408* - [User 1] = 14154455666@gw0 - [User 1] = 1510* With this setup any VoIP caller in the 408 area code will be connected to the PSTN gateway for service, while
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    are not supported here) Same as above with '1' replaced by '2' - '8' Default (xx.) (xx.) 4.14.2. PSTN Line - VoIP-To-PSTN Gateway Setup Parameter VoIP-To-PSTN Gateway Enable VoIP Caller Authentication Method VoIP PIN Max Retry One Stage Dialing Line 1 VoIP Caller DP VoIP Caller Default DP Line
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    callers with source IP address matching any of the templates will be accepted for PSTN gateway service without further authentication. For example: 192.168.*.*, 66.43.12.1?? The first of 8 PIN numbers that can be specified to control access to the PSTN gateway by a VoIP Caller, when the
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    considered for VoIP gateway service. The PSTN service must include Type I Caller-ID Delivery Service for this feature to work properly. If caller-id is blocked or not available, the caller-id is assumed to be Default yes yes [blank] [blank] none 3 yes 1 [blank] © 2003 - 2004 Sipura Technology, Inc
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    response is not greater than 2xx, the SPA will disconnect both PSTN and VoIP call legs automatically. Delay in seconds before starting to ring thru Line 1 after the PSTN starts ringing. In order for Line 1 to have the caller-id information, the delay should be set to larger than the delay required
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    service. This is minimum length of PSTN silence (or inactivity) in seconds to trigger a gateway call disconnection if is "yes" This is minimum length of VoIP with some restrictions. Default value is standard US reorder (fast busy) tone, for 4 seconds. Default yes yes yes yes
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    as a ringing signal by the SPA. Choose from {100, 150, 200, 256, 384, 512, 640, 1024} (ms) Delay in asserting a ringing signal after detecting it Default 600 0 0 3.5 10 Less than 0.5ms no 10 100 256ms 512ms © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 65
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    another extension and will therefore not permit PSTN gateway service. 640ms 13.5-16.5 Vrms High 30 4.14.8. Line 1 and PSTN Line - Audio Configuration Parameter Symmetric RTP Description If enabled, the SPA will stream RTP packets to the source IP address and port of the last received RTP packet
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    1. uid - the authentication user-id, 2. pwd - the authentication password, and 3. nat - if this parameter is present, use NAT mapping Each parameter is separated by a semi-colon (;). Default [blank] Furthermore, it recognizes "gw0", "gw1", ..., "gw4" as the locally configured gateways, where "gw0
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    SPA will skip authentication Same as above with 1 replaced with '2' - '8' VoIP destination to forward a PSTN caller matching . If this entry is [blank], the PSTN caller is blocked for VoIP service. Same as above with 1 replaced with '2' - '8' Default 4.14.16. PSTN User - PSTN-To
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    selected, the ring to be used is determined by Line 1's distinctive ring settings. Default 1 4.14.19. Info - PSTN Line Status State Parameters Description Hook State Line Voltage Loop Current Last Called VoIP Number Last VoIP Caller Last PSTN Disconnect Reason Last Called PSTN Number Last PSTN
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    Line 1 Call 1 4.14.20. PSTN/VoIP Caller Commands via DTMF Command **# **1 Description Disconnect the PSTN line (SPA will take the FXO port on-hook) End the current call and restarts dial tone 5. User Guidelines The SPA can be configured to the custom requirements of the service provider, so that
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    an E.164 ID (phone number) so that they may be reached from wired or wireless callers on the PSTN or IP network. The SPA supplies ring voltage to the attached telephone set to alert the user of incoming calls. Hang-up the telephone. 5.2. Enhanced Services © 2003 - 2004 Sipura Technology, Inc
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    INVITE message, the SPA sends the caller's name and number as it is configured in the profile. No user action required. See CLIP and CLIR. 5.2.2. Calling Line Identification Presentation (CLIP) Service Description Some users will elect to block their Caller ID information for all outgoing calls
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    each call The user activates this service to hide his Caller ID when making an outgoing call. No action required. This service is only in effect for the Call Waiting Service Description The SPA supports disabling of call waiting permanently or on a per call basis. © 2003 - 2004 Sipura Technology,
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    accepted. Callers who dial your number will receive a busy signal or, if available, the caller will be service, do the following: Lift the receiver Listen for dial tone Press *__ You will hear a confirmation tone signaling your request to cancel Call Waiting has been accepted. © 2003 - 2004 Sipura
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    support Call-Waiting with Caller ID. In between call waiting tone bursts, the SPA can generate a Caller-ID signal to the attached phone when it is off hook. Not applicable. 5.2.7. Voice Mail Service Follow instructions of the voice mail system or simply hang-up the telephone. © 2003 - 2004 Sipura
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    Dial the number to which you will transfer the caller Stay on the line until the called number Not applicable. 5.2.9. Unattended or "Blind" Call Transfer Service Description Unattended or "Blind" Call Transfer lets a customer Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 76
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    Dial the number to which you will transfer the caller The call is transferred when a complete number is ) Hang-up the telephone. 5.2.11. Three-Way Calling Service Description User Action Required to Activate or Use The user Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 77
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    flash button The SPA supports up to two calls Ad-Hoc Conference Calling Service Description This feature allows have the original caller and the party (see instructions above) Expected Call service). If you also have Call Transfer you can also hang up at any time to transfer the original caller
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    to Deactivate or End The SPA supports a service that allows the SPA to automatically dial the last caller's number. Pick up the receiver Listen Unconditional Service Description All calls are immediately forwarded to the designated forwarding number. The SPA will © 2003 - 2004 Sipura Technology,
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    to any number using the touchtone keypad of their telephone or web browser interface. This service is activated or deactivated from the phone being forwarded or the web browser interface. Lift forwarding will be © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 80
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    Service Description User Action Required to Activate or Use Expected Call and Network Behavior Calls are forwarded to the designated forwarding number after a configurable : The forward delay is entered from the web interface. Default is 20s This feature allows a user the option to divert (forward
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    Service Description User Action Required to Activate or Use Expected Call and Network Behavior User Action Required to Deactivate or End By setting the corresponding configuration parameter on the SPA, the subscriber has the option to block incoming calls that do not reveal the caller's Caller ID
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    20. Speed Calling - Up to Eight (8) Numbers or IP Addresses Service Description The SPA supports user programming of up to 8 long distance, local, international or emergency numbers and/or IP addresses To enter IP addresses, a be configured with
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    and [2]) Other timers are configurable via parameters, but do not directly pertain to the dial plan itself. They are discussed elsewhere in this document. Interdigit Long Timer: ParName: Default: Interdigit_Long_Timer 10 The Interdigit_Long_Timer specifies the default maximum time (in seconds
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    barred (rejected) by placing a '!' character at the end of the sequence. Thus, "1900xxxxxxx!" automatically rejects all 900 area code numbers from being dialed. © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 85
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    syntax allows for the implementation of Hot-Line and Warm-Line services. To achieve this, one sequence in the plan must start 2. Likewise, if Enable_IP_Dialing is enabled, then ip dialing is also accepted on the associated line. Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2) 86
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    operator (00). In addition, it lengthens the default short interdigit timeout to 4 seconds. S:4, ( blocks 9xx area codes. ( 1 [2-8]xx [2-9]xxxxxx ) The following allows arbitrary long distance dialing, but explicitly blocks - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2)
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© 2003 - 2004 Sipura Technology, Inc
Proprietary (See Copyright Notice on Page 2)
1
Sipura Technology, Inc.
SPA User Guide
July 2004