Sipura Technology SPA2002-ER - Earthlink Truevoice Phone Adpt User Manual
  1. Manuals
  2. Brands
  3. Sipura Technology Manuals
  4. Adapter
  5. SPA2002-ER - Earthlink Truevoice Phone Adpt
  6. User manual

Sipura Technology SPA2002-ER - Earthlink Truevoice Phone Adpt User Manual

User guide
Hide thumbs
Table of Contents

Advertisement

Quick Links

Sipura Technology, Inc.
SPA User Guide
July 2004
© 2003 - 2004 Sipura Technology, Inc
Proprietary (See Copyright Notice on Page 2)
1

Advertisement

Table of Contents
loading
Need help?

Need help?

Do you have a question about the SPA2002-ER - Earthlink Truevoice Phone Adpt and is the answer not in the manual?

Questions and answers

Subscribe to Our Youtube Channel

Related Manuals for Sipura Technology SPA2002-ER - Earthlink Truevoice Phone Adpt

Summary of Contents for Sipura Technology SPA2002-ER - Earthlink Truevoice Phone Adpt

  • Page 1 Sipura Technology, Inc. SPA User Guide July 2004 © 2003 - 2004 Sipura Technology, Inc Proprietary (See Copyright Notice on Page 2)
  • Page 2 Disclaimer – Please Read: This document contains implementation examples and techniques using Sipura Technology, Inc. and, in some instances, other company’s technology and products and is a recommendation only and does not constitute any legal arrangement between Sipura Technology, Inc. and the reader, either written or implied. conclusions reached and recommendations 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...

  • Page 3: Table Of Contents

    Sipura Technology, Inc. SPA User Guide Table of Contents Product Description ........................6 1.1. SPA Hardware Overview ...................... 6 Installation Overview ........................7 Software Configuration ........................8 3.1.1.1. Firmware Upgrade ........................ 8 3.2. IVR Interface ......................... 8 3.3. Web Interface ........................11 3.3.1.
  • Page 4 4.4. Authentication Methods ...................... 44 4.5. VoIP-To-PSTN Calls (Call Type #4) ................... 46 4.5.1. One-Stage Dialing ........................46 4.5.2. Two-Stage Dialing ........................47 4.6. PSTN-To-VoIP Calls (Call Type #3) ................... 48 4.7. Terminating Gateway Calls....................50 4.8. Line 1 VoIP Outbound Call Routing (Call Type #7) ............51 4.9.
  • Page 5 5.2.10. Call Hold..........................77 5.2.11. Three-Way Calling........................77 5.2.12. Three-Way Ad-Hoc Conference Calling .................. 78 5.2.13. Call Return..........................78 5.2.14. Automatic Call Back ........................ 79 5.2.15. Call FWD – Unconditional ....................... 79 5.2.16. Call FWD – Busy ........................80 5.2.17. Call FWD - No Answer ......................

  • Page 6: Product Description

    1. Product Description This guide describes basic use of the Sipura Technology SPA phone adapter – an intelligent low- density 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 intelligent, we mean the SPA maintains the states of all the calls it terminates.

  • Page 7: Installation Overview

    The SPA has the following interfaces for networking, power and visual status indication: 1. Two (2) RJ-11 Type Analog Telephone Jack Interfaces (Figure 4, above): These interfaces accept standard RJ-11 telephone connectors. An Analog touchtone telephone or fax machine may be connected to either interface. If the service supports only one incoming line, the analog telephone or fax machine should be connected to port one (1) of the SPA.

  • Page 8: Software Configuration

    3. Software Configuration 3.1.1.1. Firmware Upgrade Firmware Upgrade via PC Utility Program: From time to time, Sipura Technology will make available a PC executable file that will facilitate the upgrade of a SPA. In order to upgrade a device via this method, the end user must have administrative permission (via password protected log-in) to perform this upgrade.
  • Page 9 Note: This translation convention only applies to the password input. For example: to input password “test#@1234” by phone keypad, you need to press the following sequence of digits: 8378001234. 2. After entering a value, press the # (pound) key to indicate end of input. To Save value, press ‘1’...
  • Page 10 Enable/Disable DHCP Enter 1 to enable Requires Password Enter 0 to disable Check IP Address None IVR will announce the current IP address of SPA. Set Static IP Address Enter IP address DHCP must be “Disabled” using numbers on otherwise you will hear, the telephone key “Invalid Option,”...

  • Page 11: Web Interface

    Check SPA’s Web Server None IVR will announce the Port port that the web server is listening on. (Default is Enable/Disable Web 7932 Enter 1 to enable Requires Password Server of SPA Enter 0 to disable 732668 Manual Reboot of Unit None After you hear “Option Successful,”...

  • Page 12: Administration Privileges

    Note: The user names for both administrator and User are fixed and cannot be changed. After making changes to SPA configuration parameters, pressing “Submit All Changes” button will apply all the changes and if necessary, automatically reboot the device. Multiple changes may be made on multiple page tabs of the web interface at the same time.

  • Page 13: Reboot Url

    http://<spa-ip-addr>/resync?[[protocol://][server-name[:port]]/profile-pathname] If no parameter follows “/resync?”, the profile rule setting in provisioning is used. See Error! Reference source not found. for detailed information about profile rule in provisioning If no protocol is specified, TFTP protocol is assumed. Note: Only TFTP is supported in the current release.

  • Page 14: Provisioning Parameters

    DNS Query Mode Do parallel or sequential DNS Query Parallel Syslog Server Specify the Syslog server name and port. This feature specifies the server for logging SPA system information and critical events. Debug Server The debug server name and port. This feature specifies the server for logging SPA debug information.

  • Page 15: Upgrade Parameters

    section Log Resync Failure provisioning discussion section GPP A thru GPP P empty GPP SA thru GPP SD empty Note: In a customized SPA, the profile rule would point to a service provider’s server. 3.4.3. Upgrade Parameters Parameter Name Description Default Upgrade Enable Upgrade Error...
  • Page 16 Remove Last Reg Remove last registration before registering a new one if value is different one. DTMF Relay MIME This is the MIME Type to be used in a SIP application/dtmf-relay Type INFO message used to signal DTMF event. Hook Flash MIME This is the MIME Type to be used in a SIP application/hook-flash Type...

  • Page 17: Dynamic Payload Types

    SIT2 RSC SIP response status code to INVITE on which to play the SIT2 Tone SIT3 RSC SIP response status code to INVITE on which to play the SIT3 Tone SIT4 RSC SIP response status code to INVITE on which to play the SIT4 Tone Try Backup RSC SIP response status code on which to retry a...

  • Page 18: Sdp Audio Codec Names

    2. The configured dynamic payloads are used for outbound calls only where the SPA presents the SDP offer. For inbound calls with a SDP offer, SPA will follow the caller’s dynamic payload type assignments 3.4.4.2. SDP Audio Codec Names Parameter Name Description Default NSE Codec Name...

  • Page 19: Line 1 And Line 2 Parameters

    header in all subsequent REGISTER requests – “Warning: 399 spa <stun type>”, where <stun type> is one of the following: "Unknown NAT Type", "STUN Server Not Reachable", "STUN Server Not Responding", "Open Internet Detected", "Symmetric Firewall Detected", "Full Cone NAT Detected", "Restricted Cone NAT Detected", "Symmetric NAT Detected";...
  • Page 20 SAS Enable Enables the FXS Line to act as a Streaming Audio Source (SAS). If enabled, the line cannot be used for making outgoing calls. Instead, it auto-answers incoming calls and streams audio RTP packets to the calling party. SAS DLG Refresh If non-zero, this is the interval at which SAS sends Intvl out session refresh (SIP re-INVITE) messages to...
  • Page 21 starting jitter buffer size value is larger for higher jitter levels. This parameter controls the rate at which to adjust the jitter buffer size to reach the minimum. If the jitter level is set to high, then the rate of buffer size decrement is slower (more conservative), else faster (more aggressive).

  • Page 22: Supplementary Services Enablement

    User ID Subscriber’s user-id. Usually a E.164 number Password Subscriber’s a/c password Auth ID Subscriber’s authentication ID Use Auth ID If set to “yes”, the pair <Auth ID> and <Password> are used for SIP authentication. Else the pair <User ID> and <Password> are used. Mini Certificate Base64 encoded of Mini-Certificate concatenated Empty...

  • Page 23: Audio Settings

    The SPA provides native support of a large set of enhanced or supplementary services. All of these services are optional. The parameters listed in the following table are used to enable or disable a specific supplementary service. A supplementary service should be disabled if a) the user has not subscribed for it, or b) the Service Provider intends to support similar service using other means than relying on the SPA.
  • Page 24 Parameter Name Description Default Preferred Codec Select a preferred codec for all calls. However, the G711u actual codec used in a call still depends on the outcome of the codec negotiation protocol.G711u, G711a, G726-16, G726-24, G726-32, G726-40, G729a, G723 Use Pref Codec Only Only use the preferred codec for all calls.

  • Page 25: Dial Plan

    resources are available per SPA. Therefore it is important 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 Description Default...

  • Page 26: Call Forward And Selective Call Forward/Blocking Settings

    User 1/2 refers to the subscriber of Line 1/2. When a call is made from Line 1/2, SPA shall use the user and line settings for that Line; there is no user login support in SPA v1.0. Per user parameter tags must be appended with [1] or [2] (corresponding to line 1 or 2) in the configuration profile.

  • Page 27: Distinctive Ring And Ring Settings

    Parameter Name Description Default CW Setting Call Waiting on/off for all calls Block CID Setting Block Caller ID on/off for all calls Block ANC Setting Block Anonymous Calls on or off DND Setting DND on or off CID Setting Caller ID Generation on or off CWCID Setting Call Waiting Caller ID Generation on or off Dist Ring Setting...

  • Page 28: Regional Parameters

    3.4.7. Regional Parameters 3.4.7.1. Call Progress Tones Parameter Name Description Default Dial Tone Played when prompting the user to enter a 350@-19,440@- phone number 19;10(*/0/1+2) Second Dial Tone An alternative to <Dial Tone> when user 420@-19,520@- tries to dial a 3-way call 19;10(*/0/1+2) Outside Dial Tone An alternative to <Dial Tone>...

  • Page 29: Ring And Cwt Cadence

    Holding Tone Indicate to the local user that the far end 600@- has placed the call on hold 16;*(.1/.1/1,.1/.1/1,. 1/9.5/1) Conference Tone Plays to all parties when a 3-way 350@- conference is in progress 16;30(.1/.1/1,.1/9.7/ Secure Call This tone is played when a call is 397@-19,507@- Indication Tone successfully switched to secure mode.

  • Page 30: Control Timer Values (Sec)

    Ring6 Name Name in an INVITE’s Alert-Info Header to pick Bellcore-r6 distinctive ring/CWT 6 for the inbound call Ring7 Name Name in an INVITE’s Alert-Info Header to pick Bellcore-r7 distinctive ring/CWT 7 for the inbound call Ring8 Name Name in an INVITE’s Alert-Info Header to pick Bellcore-r8 distinctive ring/CWT 8 for the inbound call Ring Waveform...

  • Page 31: Vertical Service Code Assignment

    voltage is removed after the caller hangs up. disable After that tip-to-ring voltage is restored and dial tone will apply if the attached equipment is still off hook. CPC is disabled if this value is set to Range= 0 to 1.000 (s) Resolution = 0.001 (s) Notes: 1.
  • Page 32 CW_Per_Call_Act_Code Enable Call Waiting for the next call CW_Per_Call_Deact_Code Disable Call Waiting for the next call Block_CID_Act_Code Block CID on all outbound calls Block_CID_Deact_Code Unblock CID on all outbound calls Block_CID_Per_Call_Act_Code Block CID on the next outbound call Blcok_CID_Per_Call_Deact_Code Unblock CID on the next inbound call Block_ANC_Act_Code Block all anonymous calls Block_ANC_Deact_Code...
  • Page 33 processed by the SPA. You can empty the corresponding *code that you do not want to SPA to process. Feature Dial Services Codes One or more *code can be configured into this parameter, such as *72, or *72|*74|*67|*82, etc. Max total length is 79 chars.

  • Page 34: Outbound Call Codec Selection Codes

    1. These codes automatically appended to the dial-plan. So no need to include them in dial-plan (although no harm to do so either). 3.4.7.5. Outbound Call Codec Selection Codes: Parameter Name Description Default Prefer G711u Code Dialing code will make this codec the preferred *017110 codec for the associated call.
  • Page 35 GMT+02:00, …, GMT+13:00 FXS Port Impedance Electrical impedance of the FXS port. FXS Port Input Gain Input Gain in dB. Valid values are 6.0 to – infinity. Up to 3 decimal places FXS Port Output Gain Similar to <FXS Port Input Gain> but apply to the output signal DTMF Playback Level Local DTMF playback level in dBm (up to 1...

  • Page 36: Call Statistics Reporting

    • Off Hook Caller ID – This is used to delivery caller-id on incoming calls when the attached phone is off hook. See figure below (f). This can be call waiting 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).
  • Page 37 Hook State State of the hook switch: On or Off Registration State Registration state of the line: Not Registered, Registered or Failed Last Registration At Local time of the last successful registration Next Registration In Number of seconds before the next registration renewal Message Waiting Indicate whether new voice mails available: Yes or No Call Back Active...

  • Page 38: Spa-3000 Configuration

    4. SPA-3000 Configuration 4.1. Overview The SPA-3000 has 1 FXS and 1 FXO port. Each port is a RJ11 connector – the FXS Port is labeled “PHONE” and the FXO Port “LINE.” A standard analog telephone can be connected to the FXS/PHONE port to provide VoIP services just as with the SPA-1000 and SPA-2000.

  • Page 39: Spa-3000 Voice Configuration Organization

    would be highly undesirable especially if the call is an emergency). The SPA-3000 uses the following algorithm to determine when to close the relay: Off-Hook FXO port Close PHONE Port Relay If (Loop Current is 0) { Done (since PSTN line is not connected or inactive) Else { If (PHONE Port is on-hook) { Done (since it is not using the PSTN line)

  • Page 40: Fxs Interface

    4. Regional – Call progress tones and cadences, ring cadences, * codes, international control for the FXS Interface. The tone and cadence parameters also apply to the VoIP2 and FXO interfaces 5. Line 1 – Audio, NAT, SIP, Network, Gateway, Supplementary Services, Polarity parameters for VoIP1 and FXS 6.

  • Page 41: Fxo Interface

    The FXS interface is the same as that in the SPA-2000. Please refer to the SPA-2000 section for more details on configuration of this interface. 4.2.2. FXO Interface This is the interface to the FXO/LINE port. It controls the exchange of signals between the SPA and the PSTN line attached to the FXO/LINE port.

  • Page 42: Voip Interfaces

    Limit the use of the gateway by PSTN Caller-ID number Selectively forward PSTN callers to different VoIP destinations Notes: The <Caller-ID Method> parameter under [Regional] tab only controls the Caller-ID signal format sent by the SPA to the Caller-ID device attached to the FXS/PHONE port; it does not apply to the caller-id signal format sent to the SPA by the PSTN switch via the FXO/LINE port.

  • Page 43: Determining The Availability Of The Pstn Line

    VoIP1-to-VoIP2 and VoIP2-to-VoIP1 are internal SIP calls. Signaling and media packets are sent to the loop back address 127.0.0.1. Hence call types #6 and #7 are possible even if the Ethernet port is not connected The following sections describe each type of calls in more details. 4.2.5.

  • Page 44: Authentication Methods

    4.4. Authentication Methods VoIP Callers can be authenticated by one of the following methods by setting the <VoIP Caller Auth Method> parameter: 1. No Authentication: All callers will be accepted for service. The dial plan to be used for all VoIP callers for this case is the one selected in <VoIP Caller Default DP>.
  • Page 45 set to “none” (and so the <VoIP Caller Default DP> applies and 1-stage dialing is possible in this case). Notes: One-stage dialing is possible only if <VoIP Caller Auth Method> is “none” or “HTTP Digest”. Unless the caller number is in the <VoIP Access List >, the “PIN” method requires 2-stage dialing: Caller will need to dial the target PSTN number after entering a valid PIN.

  • Page 46: Voip-To-Pstn Calls (Call Type #4)

    wildcard characters ‘?’ and ‘*’ (same syntax as <PSTN Caller ID Pattern>). If the list is not blank and the Caller-ID number matches any of the patterns in the list, that PSTN caller will be granted access to the VoIP gateway as if the Authentication Method is set to “none” (and so the <PSTN Caller Default DP>...

  • Page 47: Two-Stage Dialing

    dialed-number is not specified or is the same as <User ID>, the SPA interprets this as a request for 2- stage dialing (see next section). Otherwise, the SPA processes the dialed-number by a corresponding dial plan. If the dial plan processing fails, the SPA replies with a 403 response. Otherwise, it replies with a 200 response and at the same time takes the FXO port off hook and dials the final number returned from the dial plan to the PSTN switch.

  • Page 48: Pstn-To-Voip Calls (Call Type #3)

    If (Authentication Method is HTTP) { If (Authorization Invalid) { Reply 401 Else { If (Target Number Valid) { Reply 200 Start PSTN Gateway Service Else { Reply 403 Else { Reply 200 Get VoIP PIN from Caller If (Valid PIN) { Start PSTN Gateway Service Else { Send BYE...
  • Page 49 switch (for US telcos, this value should be at least 3-4s). If <PSTN Caller Auth Method> is “PIN”, the SPA then prompts the caller to enter the PIN number. If the given PIN matches one of <PSTN Caller n PIN> values, the SPA plays the Dial Tone to the FXO port and is ready to accept digits of the VoIP target number from the PSTN caller.

  • Page 50: Terminating Gateway Calls

    The configuration parameters mentioned in this section are: [PSTN Line]<PSTN-To-VoIP Gateway Enable> [PSTN Line]<Line Enable> [PSTN Line]<Make Call Without Reg> [PSTN Line]<PSTN Caller ID Pattern> [PSTN Line]<PSTN Access List> [PSTN Line]<PSTN Answer Delay> [PSTN Line]<PSTN CID For VoIP CID> [PSTN Line]<PSTN CID Number Prefix> [PSTN Line]<PSTN CID Name Prefix>...

  • Page 51: Line 1 Voip Outbound Call Routing (Call Type #7)

    is a SIP Re-INVITE request which the VoIP peer must reply with a 200 class response. The sending of dialog refresh messages can be disabled by setting <VoIP DLG Refresh Intvl> to 0. When any of the above occurs, the SPA takes the FXO port on hook and sends the proper SIP signaling messages to end the VoIP call leg.

  • Page 52: Line 1 Voip Fallback To Pstn

    <8,:1408>xxxxxxx<:@pstn.sipura.com:5 User dials 8 to start Outside Dial Tone. When user dials 061;usr=joe;pwd=joe_pwd;nat> a 7-digit 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 <8,:1408>xxxxxxx<:@gw2:5061;usr=”Al User dials 8 to start Outside Dial Tone.

  • Page 53: Pstn Call Ring Thru Line 1 (Call Type #6)

    destination. This is Call Type #5 and the voice path is (3) (8) (7). Similarly, Line 1 can apply Call- Forward-All, Call-Forward-On-Busy, and Call-Forward-Selective features and send the caller to use the PSTN gateway. Only the PIN Authentication method is allowed when a VoIP caller is forwarded to access the PSTN gateway from Line 1.

  • Page 54: Symmetric Rtp

    affects both ringing signal and call-waiting tone. Unlike the <Default Ring> setting under the [User 1] tab, the <Default Ring> parameter under the [PSTN User] tab has the extra choice of “Follow Line 1”, which means to follow the Line 1 ringer settings (including distinctive ringing rules) instead of forcing it to use a particular ring cadence.

  • Page 55: Setup Voip1 And Voip2 With Same Voip Account

    [Line 1]<Proxy> = fwd.pulver.com [Line 1]<Outbound Proxy> = fwdnat.pulver.com:5082 [Line 1]<Use Outbound Proxy> = yes [Line 1]<Use OB Proxy in Dlg> = yes [Line 1]<User ID> = userid_A [Line 1]<Password> = password_A [Line 1]<Register> = yes [Line 1]<Register Expires> = 3600 [PSTN Line]<Line Enable>...

  • Page 56: Pstn Call Answered By Line 1

    [PSTN Line]<PSTN Ring Thru Line> = no [PSTN Line]<PSTN Answer Delay> = 14 [PSTN Line]<PSTN Caller 1 PIN> = 1234 [PSTN Line]<PSTN Caller n PIN> = (blank), n = 2 to 8 [PSTN Line]<PSTN Caller 1 DP> = 1 [PSTN Line]<Dial Plan 1> = (1408xxxxxxx|408xxxxxxx|xxxxxxx|1800xxxxxxx|800xxxxxxx) [PSTN Line]<PSTN Caller Auth Method>...

  • Page 57: Voip-To-Pstn Call Via Voip2 Interface With Pin Authentication

    If Line 1 is busy when the PSTN line rings, the SPA will not attempt to ring through, even if Line 1 later becomes idle while the PSTN is still ringing. c) Line 1 connected on a VoIP call (via the VoIP1 interface) at the time the PSTN rings When the PSTN line rings, the SPA plays Call Waiting Tone to alert the Line 1 user and shows call- waiting Caller-ID to the Line 1 phone.

  • Page 58: Line 1 Forward-On-No-Answer To Pstn Gateway

    [PSTN Line]<VoIP Caller Auth Method> = HTTP Digest [PSTN Line]<VoIP User 1 Auth ID> = jdole [PSTN Line]<VoIP User n Auth ID> = (blank), n = 2 to 8 [PSTN Line]<VoIP User 1 Password> = silicon-valley [PSTN Line]<VoIP User 1 DP> = 3 [PSTN Line]<Dial Plan 3>...

  • Page 59: Line 1 Forward-All To Pstn Gateway

    The <VoIP Caller ID Pattern> and <VoIP Access List> parameters still apply to the VoIP callers when they access the PSTN gateway via the VoIP1 interface. If the caller is not allowed by the <VoIP Caller ID Pattern> the SPA will not answer the call. If the caller belongs to the <VoIP Access List>, authentication is not required.

  • Page 60: From Line 1 Route 311 And 911 Calls To Pstn-Gateway

    4.13.12. From Line 1 Route 311 and 911 Calls to PSTN-Gateway Insert the rule “[39]11<:@gw0>” to [Line 1]<Dial Plan>, and set [PSTN Line]<Line 1 VoIP Caller DP> = none When the user picks up the Line 1 phone and dials 311 or 911, the call is routed to the PSTN gateway.

  • Page 61: Pstn Line - Voip Users And Passwords (Http Authentication)

    link. Choose from {none, 1, 2, 3, 4, 5, 6, 7, 8}. Note: Authentication is skipped for Line 1 VoIP caller VoIP Caller ID A comma separated list of caller number templates [blank] Pattern such that callers with numbers not matching any of these templates will be rejected for PSTN gateway service regardless of the setting of the authentication method.

  • Page 62: Pstn Line - Pstn-To-Voip Gateway Setup

    compute the credentials using this password, or the INVITE will be challenged with a 401 response VoIP User 2–8 Same as above with ‘1’ replaced by ‘2’ – ‘8’ [blank] Password 4.14.4. PSTN Line – PSTN-To-VoIP Gateway Setup Parameter Description Default PSTN-To-VoIP Enable or disable PSTN-To-VoIP Gateway functionality.

  • Page 63: Pstn Line - Fxo Timer Values - In Seconds

    “Anonymous”. For example: 1408*, 1512???1234, Anonymous PSTN Access List A comma separated list of caller number templates [blank] such that PSTN callers with numbers matching any of these templates will be accepted for VoIP gateway service without authentication. PSTN Caller 1 PIN The first of 8 PIN numbers for authenticating PSTN [blank] callers to obtain VoIP gateway services.

  • Page 64: Pstn Line - Pstn Disconnect Detection

    PSTN Ring Delay after a ring burst before the SPA decides that Timeout PSTN ring has ceased 4.14.6. PSTN Line – PSTN Disconnect Detection Description Default Parameter Detect CPC CPC is a brief removal of Tip-and-ring voltage. If enabled, the SPA will disconnect both call legs when this the signal is detected during a gateway call Detect Polarity If enabled, SPA will disconnect both call legs when this...

  • Page 65: Pstn Line - International Control

    detection 5. 6 segments of on/off time (seconds) can be specified. A 10% margin is used to validated cadence characteristics of the tone Min CPC Duration Minimum duration (in seconds) of a low tip-and-ring voltage (below 1V) for the SPA to recognize as a CPC signal or PSTN line removal.

  • Page 66: Line 1 And Pstn Line - Audio Configuration

    Delay on the PSTN line. Choose from {0, 512, 768, 1024, 1280, 1536, 1792} (ms) Ring Timeout Delay in de-asserting a ringing signal after detecting 640ms that it has been removed from the PSTN line. Choose from {0, 128, 256, 384, 512, 640, 768, 896, 1024, 1152, 1280, 1408, 1536, 1664, 1792, 1920} (ms) Ring Threshold Minimum Vrms threshold to detect ringing.

  • Page 67: Line 1 - Voip Fallback To Pstn

    4.14.10. Line 1 – VoIP Fallback To PSTN Parameter Description Default Auto PSTN Fallbak If enabled, the SPA will automatically route all calls to the PSTN gateway when the Line 1 proxy is down (registration failure or network link down). 4.14.11.

  • Page 68: User1 - Selective Call Forward Settings

    4.14.13. User1 – Selective Call Forward Settings Parameter Description Default Cfwd Sel1 Dest Same as Cfwd All Dest [blank] Cfwd Sel2–8 Dest Same as Cfwd All Dest [blank] Cfwd Last Dest Same as Cfwd All Dest [blank] 4.14.14. Regional – Call Progress Tones Parameter Description Default...

  • Page 69: Pstn User - Pstn Ring Thru Line 1 Ring Settings

    4.14.18. PSTN User – PSTN Ring Thru Line 1 Ring Settings Parameter Description Default Default Ring The default ring to be used to ring through Line 1. Choose from {1,2,3,4,5,6,7,8,Follow Line 1}. If “Follow Line 1” is selected, the ring to be used is determined by Line 1’s distinctive ring settings.

  • Page 70: Pstn/Voip Caller Commands Via Dtmf

    PSTN Caller Accepted Connected to PSTN PSTN Offhook/VoIP Ended PSTN Tone Indicate what tone is being played to the PSTN call leg PSTN Peer Name Name of the party at the PSTN call leg PSTN Peer Number Phone number of the party at the PSTN call leg VoIP State Same as Line 1 Call 1 Mapped SIP Port...

  • Page 71: Basic Services

    This section of the Administration Guide, describes how some of the supported basic and enhanced, or supplementary services could be implemented. The implementations described below by no means are the only way to achieve the desired service behavior. To understand the specific implementation options of the below features, including parameters, requirements and contingencies please refer the section Configuration Parameters, section 0.

  • Page 72: Caller Id

    5.2.1. Caller ID Service Description If available, the SPA supports the generation and pass through of Caller ID information. User Action Required to Activate or Use No user action required. The user’s telephone equipment must support Caller ID to display the caller’s name and/or number.

  • Page 73: Call Waiting

    User Action Required to Activate or Use Lift the receiver Listen for dial tone Press *__ Listen for dial tone Dial the telephone number you are calling You must repeat this process at the start of each call Expected Call and Network Behavior The user activates this service to hide his Caller ID when making an outgoing call.
  • Page 74 User Action Required to Activate or Use To temporarily disable Call Waiting (for the length of one call): Before placing a call: Lift Receiver Press *__ Listen for dial tone then dial the number you want to call. Call Waiting is now disabled for the duration of this call only.

  • Page 75: Call-Waiting With Caller Id

    5.2.6. Call-Waiting with Caller ID Service Description When the user is on the phone and has Call Waiting active, the new caller’s Caller ID information will be displayed on the users phone display screen at the same time the user is hearing the Call Waiting beeps / tones.

  • Page 76: Attendant Call Transfer

    5.2.8. Attendant Call Transfer Service Description Attendant Call Transfer lets a customer use their Touchtone phone to send a call to any other phone, inside or outside their business, including a wireless phones. User Action Required to Activate or Use While in a call with the party to be transferred: Press the switch hook or flash button on the phone to place the party on hold...

  • Page 77: Call Hold

    phone to place the party on hold Enter *__ Dial the number to which you will transfer the caller The call is transferred when a complete number is entered. You will hear a short confirmation tone, followed by regular dial tone Expected Call and Network Behavior When the user presses the switch hook or flash button, the transferee is placed on hold.

  • Page 78: Three-Way Ad-Hoc Conference Calling

    Dial the number of the 3 party. When the 3 party answers you may have a conversation with them while the other party is on hold. To hold a conference with the party on hold and the 3 party, simply press the switch hook or flash button Expected Call and Network Behavior The SPA supports up to two calls per line.

  • Page 79: Automatic Call Back

    Service Description The SPA supports a service that allows the SPA to automatically dial the last caller’s number. User Action Required to Activate or Use Pick up the receiver Listen for dial tone Press *__ to dial back the last caller that tried to reach you.

  • Page 80: Call Fwd - Busy

    not ring or provide call waiting when Call FWD – Unconditional is activated. User Action Required to Activate or Use Lift the receiver Listen for dial tone Press *__ Listen for dial tone and enter the telephone number you are forwarding your call to. Activation will be confirmed with three short bursts of tone and your forwarding will be activated.

  • Page 81: Call Fwd - No Answer

    activated. Alternatively, the user can activate this feature from a web browser interface. Expected Call and Network Behavior This feature allows a user the option to divert (forward) calls to their telephone number to any number when their phone is busy or in conference by using the touchtone keypad of their telephone or web browser interface.

  • Page 82: Anonymous Call Blocking

    service is activated or deactivated from the phone being forwarded or the web browser interface. User Action Required to Deactivate or End Lift the receiver Listen for dial tone Press *__ You will hear a confirmation tone signaling your change has been accepted. Alternatively, the user can deactivate this feature from a web browser interface.

  • Page 83: Speed Calling - Up To Eight (8) Numbers Or Ip Addresses

    accompanies incoming calls from designated telephone numbers. If the user is engaged in conversation and a call from one of the designated numbers arrives, a distinctive call waiting tone (i.e. short- long-short) accompanies the incoming call. Calls from other telephone numbers ring normally.
  • Page 84 The SPA syntax for the dial plan closely resembles the corresponding syntax specified by MGCP and MEGACO. Some extensions are added that are useful in an end-point. The dial plan functionality is regulated by the following configurable parameters: • Interdigit_Long_Timer •...
  • Page 85 Any one of a set of terminating events triggers the SPA to either accept the user-dialed sequence, and transmit it to initiate a call, or else reject it as invalid. The terminating events are: • No candidate sequences remain: the number is rejected. •...
  • Page 86 Interdigit Timer Master Override: The long and short interdigit timers can be changed in the dial plan (affecting a specific line) by preceding the entire plan with the following syntax: • Long interdigit timer: ‘L’ ‘:’ delay-value ‘,’ • Short interdigit timer: ‘S’ ‘:’ delay-value ‘,’ Thus, “L=8,( .
  • Page 87 ( <9,:> 1 xxx xxxxxxx | <8,:1212> xxxxxxx ) The following allows only placing international calls (011 call), with an arbitrary number of digits past a required 5 digit minimum, and also allows calling an international call operator (00). In addition, it lengthens the default short interdigit timeout to 4 seconds.

This manual is also suitable for:

Spa-2000Spa-1000Spa-3000

Table of Contents