Cisco Linksys RT31P2 Administration Manual
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Cisco Linksys RT31P2 Administration Manual

Administration guide
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Linksys PAP2 and RT31P2
PHONE ADAPTER Administration Guide
August 2004
© 2004 Linksys Proprietary (See Copyright Notice on Page 2)
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Summary of Contents for Cisco Linksys RT31P2

  • Page 1 Linksys PAP2 and RT31P2 PHONE ADAPTER Administration Guide August 2004 © 2004 Linksys Proprietary (See Copyright Notice on Page 2)
  • Page 2 Disclaimer – Please Read: This document contains implementation examples and techniques using Linksys and, in some instances, other company’s technology and products and is a recommendation only and does not constitute any legal arrangement between Linksys and the reader, either written or implied. The 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...

  • Page 3: Table Of Contents

    VoIP-NAT Interworking......................... 16 1.3. Voice Quality Overview ....................16 Hardware Overview .........................17 2.1. Phone Adapter LED Status .....................19 2.2. Broadband Router (RT31P2) LED Status ................19 Software Configuration Mechanisms..................20 3.1. Configuration Profile Formats ..................21 3.1.1. Using the Supplemental Profile Compiler ..................23 3.1.2.
  • Page 4 4.9. Progress Tone and Ring Configuration................67 4.9.1. Distinctive Ring and Other Ring Settings..................67 4.9.2. Progress Tones ..........................69 4.10. Less Frequently Used Paramters ..................70 4.10.1. Advanced Protocol Parameters ....................70 4.10.2. Additional User Account Information..................73 4.10.3. Per-Line Polarity Settings ......................75 4.10.4.
  • Page 5 7.2.1.3. SIP Support in Network Address Translation Networks – NAT ........... 96 7.2.2. Codec Name Assignment......................96 7.2.3. Secure Calls ..........................97 7.2.4. Voice Algorithms: ......................... 97 7.2.4.1. G.711 (A-law and mµ-law) ....................97 7.2.4.2. G.726 ..........................97 7.2.4.3. G.729A..........................97 7.2.4.4.

  • Page 6: Introduction

    1. Introduction This guide describes basic administration and use of the Linksys Technology PHONE ADAPTER phone adapter – an intelligent low-density Voice over IP (VoIP) gateway. The PHONE ADAPTER enables carrier class residential and business IP Telephony services delivered over broadband or high-speed Internet connections.
  • Page 7 SIP enables the implementation of intelligent endpoints to support scalable advanced services. In a nutshell, SIP is a distributed signaling protocol (as opposed to a centralized protocol such as SS7, MGCP or MEGACO/H.248). With a distributive protocol, the intelligence does not necessarily reside on a central server, but can be built into the individual endpoints.

  • Page 8: Components Of A Sip Network

    1.1.1. Components of a SIP Network Subscriber Service Database Provider Domain Provisioning Billing Application Application Server Proxy Server Server Server Server Phone Adapter PSTN Gateway PSTN Network Broadband Gateway PSTN Modem Private IP (Internet) Network PSTN Gateway Router Subscriber Domain Figure 1 -- Components of a SIP IP Telephony Network IP Telephony Gateway (PHONE ADAPTER): The PHONE ADAPTER is a small device that sits at the subscriber’s premises.

  • Page 9: Provisioning Overview

    gateways, etc. The default router in this case is the IP address of the LAN interface of the router itself. Performs NAT on packets sent from the private network to the public network. This is an important feature such that recipients of the private packets will perceive them as originated from a public IP address (the router’s WAN interface) and will therefore return messages to the proper public IP address and port.

  • Page 10: Security Overview

    ADAPTER unit is shipped from the factory, it contains a default common Profile and is considered Unprovisioned. To save costs and expedite delivery, however, it is very desirable that an Unprovisioned unit can be shipped directly from the factory to the subscriber’s location without any preprocessing by the Service Provider.

  • Page 11: Proxy Servers

    The SIP signaling messages – The SIP messages exchanged between the SIP proxy server and the PHONE ADAPTER should be encrypted with a secret key. This can be achieved, for instance, by transporting SIP over TLS. RTP packets – The RTP payload exchanged between SIP user agents can be encrypted with a secret key to protect against eavesdropper.

  • Page 12: Basic Services

    1.1.4.1. Basic Services 1.1.4.1.1. Making Calls to PSTN and IP Endpoints This is the most basic service. When the user picks up the handset, the PHONE ADAPTER provides dial tone and is ready to collect dialing information via DTMF digits from a touch tone telephone. While it is possible to support overlapped dialing within the context of SIP, the PHONE ADAPTER collects a complete phone number and sends the full number in a SIP INVITE message to the proxy server for further call processing.
  • Page 13 1.1.4.2.3. Voice Mail Message Waiting Indication Service Providers may provide voice mail service to their subscribers. When voice mail is available for a subscriber, a notification message will be sent from the Voice Mail server to the PHONE ADAPTER. The PHONE ADAPTER indicates that a message is waiting by, playing stuttered dial tone (or other configurable tone) when the user picks up the handset.

  • Page 14: Pstn Interworking

    telephone number to forward calls to. The PHONE ADAPTER provides audio instructions to prompt the user for a forwarding number and confirms that the requested service has been activated. Call FWD – Unconditional All calls are immediately forwarded to the designated forwarding number. The PHONE ADAPTER will not ring or provide call waiting when Call FWD –...

  • Page 15: Network Address Translation (Nat) Traversal

    If the service provider is offering origination and/or termination on endpoint equipment then it is very likely that the softswitch chosen for network operations will support multiple PSTN and VoIP signaling protocols. The table below lists the most commonly accepted, de-facto standards used when implementing a VoIP signaling scheme based on the type of gateway or endpoint equipment being deployed: VoIP Equipment Type Typical Port Density...

  • Page 16: Voip-Nat Interworking

    address/port to the corresponding private source address/port. These characteristics of a NAT can be exploited by an PHONE ADAPTER to let external entities send SIP messages and RTP packets to it when it is installed on a private network. 1.2.2. VoIP-NAT Interworking In the case of SIP, the addresses where messages/data should be sent to an PHONE ADAPTER are embedded in the SIP messages sent by the device.

  • Page 17: Hardware Overview

    CPE device. Figures Figure 2 and Figure 3 show the front and rear, of the PHONE ADAPTER, respectively. Figures 4 and 5 show the front and rear, of the RT31P2 Broadband Router, respectively.
  • Page 18 If the service supports only one incoming line, the analog telephone or fax machine should be connected to port one (1) of the RT31P2. Port one (1) is the outermost telephone port on the RT31P2 and is labeled “Phone 1.”...

  • Page 19: Phone Adapter Led Status

    This interface accepts the PHONE ADAPTER power adapter that came with the unit. Linksys does not support the use of any other power adapters other then the power adapter that was shipped with the PHONE ADAPTER unit or the Broadband Router (RT31P2) © 2004 Linksys Proprietary (See Copyright Notice on Page 2)

  • Page 20: Software Configuration Mechanisms

    Please check to make sure that you have the following package contents: 1. Linksys Phone Adapter Unit or Linksys Broadband Router (RT31P2) 2. Ethernet Cable 3. 5 Volt (PAP2) or 12 Volt (RT31P2) Power Adapter 4. CD with User Guide You will also need: 1.

  • Page 21: Configuration Profile Formats

    The PHONE ADAPTER also provides a Web Interface with two-level access (user-level and admin- level) to configuration parameters. For standalone PHONE ADAPTERS (which contain no router or NAT functionality), an IVR (Interactive Voice Response) interface is also available for configuring basic networking.
  • Page 22 respectively. If neither mark is present, the parameter is made inaccessible to the user from the web interface. Note that this syntax has no effect on the admin-level access to the parameters. When using the SPC, a service provider is given full control over which parameters become inaccessible, read-only, or read-write following provisioning of the PHONE ADAPTER.

  • Page 23: Using The Supplemental Profile Compiler

    # The Phone Adapter1234.txt file above is equivalent to . . . Param1 “base value 1” ; Param2 “base value 2” ; . . . Param1 “new value overrides base” ; Param7 “particular value 7” ; . . . A sample plain-text file, containing default parameter-value and access settings for the PHONE ADAPTER can be obtained from the profile compiler tool, using the following command-line arguments.

  • Page 24: Encrypting And Compressing Xml Configuration Files

    An encrypted CFG file requires either a password (or quoted pass-phrase) or a hex-string. The following lines illustrate command-line invocations for various combinations of keys and algorithms. spc –-rc4 –-ascii-key apple4sale pap2.txt pap2.cfg spc –-aes –-ascii-key lucky777 pap2.txt pap2.cfg spc –-aes –-ascii-key “my secret phrase” pap2.txt pap2.cfg spc –-aes –-hex-key 8d23fe7...a5c29 pap2.txt pap2.cfg A CFG file can be both targeted and key encrypted, as suggested by the following example: spc –-target 000e08aaa010 –-aes –-hex-key 9a20...eb47 a.txt a.cfg...

  • Page 25: Secure Initial Configuration

    This utility generates 8-bytes of salt (which is prepended to the encrypted configuration file), and then calculates an Initialization Vector (IV) and an 256-bit encryption key using the key phrase provided on the command line. The TA recognizes the leading characters "Salted__" as a hint to find the salt and decrypt the configuration file.

  • Page 26: Web Interface

    RT31P2: http://IP_Address_Of_PHONE ADAPTER/Voice_adminPage.htm . The default IP address for the LAN interface of the RT31P2 is 192.168.15.1. See the next section for more information about administration privileges. The PHONE ADAPTER supports Internet Explorer 5.5 and above and Netscape 7.0 and above.

  • Page 27: Administration Privileges

    Upgrade, Reboot, Profile Resync, and Factory Reset. Administrator privilege is needed for these functions. Note that on the RT31P2, these URLs are only accessible from the LAN interface, unless the Admin_Passwd has been set and the Enable_Web_Admin_Access parameter is set.

  • Page 28: Resync Url

    http://<PAP2-ip-addr>/admin/upgrade?[protocol://][server-name[:port]][/firmware-pathname] If no protocol is specified, TFTP is assumed. Note: Only TFTP is supported in the current release. If no server-name is specified, the host that requests the URL is used as server-name. If no port specified, default port of the protocol is used. (69 for TFTP, 80 for http, 443 for HTTPS) The “firmware-pathname”...

  • Page 29: Configuration Via The Ivr (Pap2 Only)

    3.5. Configuration via the IVR (PAP2 only) Administrators and/or users can check (read) and set (write) basic network configuration settings via a touchtone telephone connected to one of the RJ-11 phone ports of the PAP2 model PHONE ADAPTER. Please Note: Service Providers offering service using the PHONE ADAPTER may restrict, protect or turn off certain aspects of the unit’s IVR and web configuration capabilities.
  • Page 30 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) key twice within a half second window of time.
  • Page 31 PHONE ADAPTER. Set Network Mask Enter value using DHCP must be “Disabled” numbers on the otherwise you will hear, telephone key pad. “Invalid Option,” if you try Use the * (star) key to set this value. when entering a Requires Password decimal point.

  • Page 32: Configuration Parameters

    877778 User Factory Reset of Unit Enter 1 to confirm PHONE ADAPTER will Enter *(star) to prompt for confirmation. WARNING: cancel operation After confirming, you will ALL “User-Changeable” NON- hear “Option Successful.” DEFAULT SETTINGS WILL BE Hang-up. Unit will reboot LOST! and all “User Changeable”...
  • Page 33 wildcard characters. It can contain up to 39 characters. Examples: “1408*, 1510*”, “1408123????, 555?1”. RscTmplt – A template of SIP Response Status Code, such as “404, 5*”, “61?”, “407, 408, 487, 481”. It can contain up to 39 characters. CadScript – A mini-script that specifies the cadence parameters of a signal. Up to 127 characters.
  • Page 34 Number of Frequencies = 2 Frequency 1 = 350 Hz at –19 dBm Frequency 2 = 440 Hz at –19 dBm ToneScript – A mini-script that specifies the frequency, level and cadence parameters of a call progress tone. May contain up to 127 characters. Syntax: FreqScript;Z ].

  • Page 35: Conventions

    Segment 2: On=0.38s, Off=0s, with Frequency 2 Segment 3: On=0.38s, Off=0s, with Frequency 3 Segment 4: On=0s, Off=4s, with no frequency components Total Tone Length = 20s ProvisioningRuleSyntax – Scripting syntax used to define configuration resync and firmware upgrade rules. Refer to the provisioning discussion in the next section for a detailed explanation of the syntax.
  • Page 36 GPP_A through GPP_P GPP_SA through GPP_SD Provision Enable: ParName: Provision_Enable Default: Enable The CFG profile must be requested by the PHONE ADAPTER, and cannot be pushed from a provisioning server (although a service provider can effectively push a profile by triggering the request operation remotely via a SIP NOTIFY).
  • Page 37 active call is in progress. The PHONE ADAPTER will wait up to Forced_Update_Delay seconds for both lines to become idle. If the adapter still is not idle, the adapter will perform a resync anyway. Resync Error Retry Delay: ParName: Resync_Error_Retry_Delay Default: 3600 If a resync attempt fails, the PHONE ADAPTER will retry with a delay indicated by the...
  • Page 38 These strings each supports one level of macro expansion, using a small set of variables. Following macro substitution, the rule is evaluated to obtain the URL of the CFG file to be requested from the provisioning server. The URL can be partially specified, in which case default values are assumed for the unspecified terms.
  • Page 39 expr = DQUOT val DQUOT options = "[" *option "]" option = key-opt / alias-opt / post-opt key-opt = "--key" key-string key-string = password / quoted-pass-phrase / hex-string alias-opt = "--alias" val post-opt = "--post" val url = [ method "://" [ server [":"...
  • Page 40 In addition, the contents of the general purpose parameters, GPP_A, through GPP_P, are available as macro variables A through P, respectively. A secondary set of general purpose parameters is also available for macro substitution, GPP_SA, GPP_SB, GPP_SC, GPP_SD, using the respective expressions SA, SB, SC, and SD. These parameters are not accessible through the web interface, and can only be set via a configuration profile.
  • Page 41 Profile_Rule “[--key $B] ps.global.com/profiles/active/$A/pap2.cfg”; GPP_A “Dz3P2q9sVgx7LmWbvu”; GPP_B “83c1e792bc6a824c0d18f429bea52d8483f2a24b32d75bc965d05e38c163d5ef”; In practice, the first provisioning stage (which individualizes each PHONE ADAPTER into fetching a unique CFG file) could be preconfigured during manufacturing. For added security, the second stage, which introduces strong encryption, may be performed in- house, prior to shipping an PHONE ADAPTER to each end-user.
  • Page 42 string supports one level of macro substitution, with the same variables as for the Profile_Rule above. An empty string does not generate a syslog message. General Purpose Parameters: ParName: GPP_A through GPP_P Default: empty GPP_A through GPP_P are the 16 General Purpose Parameters, usable by both the provisioning and the upgrade logic.

  • Page 43: Firmware Upgrade

    Log Resync Request Syslog message generated when attempting ProfileMsg a resync provisioning discussion section Log Resync Success Syslog message generated after a ProfileMsg successful resync provisioning discussion section Log Resync Failure Syslog message generated after a failed ProfileMsg resync provisioning discussion section GPP A thru GPP P...
  • Page 44 The firmware file must be requested by the PHONE ADAPTER and cannot be pushed from an upgrade server (although a service provider can effectively push a new firmware load by triggering the request operation remotely via the CFG file). The functionality is controlled by the Upgrade_Enable parameter.
  • Page 45 The upgrade will fail if the new firmware load does not satisfy the upgrade rule condition that suggested the URL. This alleviates the possibility of infinite upgrade loops, in case the device has been misconfigured. The rule syntax is the same as for the Profile_Rule described in a previous section, except that there are no supported optional qualifiers for upgrades at this time.

  • Page 46: Provisioning Server Redundancy

    string supports one level of macro substitution, with the same variables as for the Upgrade_Rule above. An empty string does not generate a syslog message. Parameter Name Description Type Default Upgrade Enable Master enable for firmware upgrade Bool operations Upgrade Error Retry interval following upgrade failure Time0 3600...

  • Page 47: Basic Networking Configuration

    (for example, using DHCP) or are configured by the end user of the device. Note that the RT31P2 ignores the following parameters: DHCP, Static_IP, NetMask, and Gateway. Other than the DNS_Server_Order and DNS_Query_Mode, the rest these parameters also can be configured from the RT31P2 User GUI.

  • Page 48: Basic Account Configuration

    Parallel DNS query mode: PHONE ADAPTER will send the same request to all the DNS servers at the same time when doing a DNS lookup, the first incoming reply will be accepted by PHONE ADAPTER. To log SIP messages, Debug Level must be set to at least 2. If both Debug Server and Syslog Server are specified, _Syslog messages are also logged to the Debug Server.

  • Page 49: Configuration For Nat Traversal

    Note: The Linksys model RT31P2 includes NAT (Network Address Translator) functionality. As long as the IP address of the "WAN Port" is a public IP address, the RT31P2 can be configured with all NAT Traversal features (NAT Traversal off), since the PHONE ADAPTER portion shares the same IP address as the WAN Port.
  • Page 50 ADAPTER. Also, set the Substitute_VIA_Addr and NAT_Mapping_Enable parameters. Follow the instructions of the NAT software to configure static NAT mappings between the external address and ports (EXT_SIP_Port, EXT_RTP_Port_Min) and the internal address and ports (SIP_Port, RTP_Port_Min). Set the RTP_Port_Max parameter to a smaller number (for example, RTP_Port_Min plus 8).

  • Page 51: Media And Sdp (Session Description Protocol) Configuration

    Ext SIP Port External port to substitute for the actual SIP port of Port the unit in all outgoing SIP messages. If “0” is specified, no SIP port substitution is performed. Ext RTP Port Min External port mapping of <RTP Port Min>. If this Port value is non-zero, the RTP port number in all outgoing SIP messages is substituted by the...

  • Page 52: Codec And Audio Settings

    The administrator can select a method for conveying DTMF and hookflash on a per-line basis. In addition, the administrator can also configure the MIME type (Content-Type header) used when conveying DTMF or hookflash in SIP INFO messages. The MIME type is set once for both lines. DTMF Tx Method Method to transmit DTMF signals to the far end: Choice:...

  • Page 53: Dynamic Payload Types And Sdp Codec Names

    FAX CED Detect Enable Enable detection of FAX tone. Bool FAX CNG Detect Bool Enable FAX Passthru Codec Codec to use for fax passthru {G711u, G711u G711a} FAX Codec Symmetric Force unit to use symmetric codec during FAX Bool passthru FAX Passthru Method Choices: None / NSE / ReINVITE Choice...

  • Page 54: Secure Media Implementation

    G726r40 Codec Name G726-40 Codec name used in SDP Str31 G726-40 G729a Codec Name G729a Codec name used in SDP Str31 G729a G729b Codec Name G729b Codec name used in SDP Str31 G729ab G723 Codec Name G723 Codec name used in SDP Str31 G723 Notes:...
  • Page 55 Expiration Date (12B) Public Key (512b or 64B) Signature (1024b or 512B) The signing agent is implicit and must be the same for all PHONE ADAPTER’s that intended to communicate securely with each other. The public key of the signing agent is pre-configured into the PHONE ADAPTER’s by the administrator and will be used by the PHONE ADAPTER to verify the Mini-Certificate of its peer.

  • Page 56: Outbound Call Codec Selection Codes

    e3VgYxWCQNa335YCnDsenASeBxuMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZ YTccnZ75TuTjj13qvYs= 5nEtOrkCa84/mEwl3D9tSvVLyliwQ+u/Hd+C8u5SNk7hsAUZaA9TqH8Iw0J/IqSrsf6scsmundY5j7Z5m K5J9uBxSB8t8vamFGD0pF4zhNtbrVvIXKI9kmp4vph1C5jzO9gDfs3MF+zjyYrVUFdM+pXtDBxmM+f GUfrpAuXb7/k= - user-name is the name of the subscriber, such as “Joe Smith”. Maximum length is 32 characters - user-id is the user-id of the subscriber and must be exactly the same as the user-id used in the INVITE when making the call, such as “14083331234”.

  • Page 57: Supplementary Services

    Prefer G723 Code Dialing code will make this codec the preferred ActCode *01723 codec for the associated call. Force G723 Code Dialing code will make this codec the only ActCode *02723 codec that can be used for the associated call. Prefer G726r16 Code Dialing code will make this codec the preferred ActCode...

  • Page 58: Supplementary Services Activated Internally

    CWCID Serv Enable Call Waiting Caller ID Service Bool Call Return Serv Enable Call Return Service Bool Call Back Serv Enable Call Back Service Bool Three Way Call Serv Enable Three Way Calling Service Bool Three Way Conf Enable Three Way Conference Service Bool Serv Attn Transfer Serv...
  • Page 59 Cfwd Last Deact Code Cancel call forward last ActCode Block Last Act Code Block the last inbound call ActCode Block Last Deact Code Cancel blocking of the last inbound call ActCode Accept Last Act Code Accept the last outbound call. Let it ring ActCode through when DND or Call Forward All is in effect...

  • Page 60: Call Forwarding Implemented Internally

    Secure Call Setting If yes, all outbound calls are secure calls by default Bool 4.7.2. Call Forwarding Implemented internally The PHONE ADAPTER supports local call forwarding services (Call Forward All, Call Forward Busy, Call Forward No Answer, and Selective Call Forwarding for up to 8 numbers). Parameter Name Description Type...

  • Page 61: Dial Plan Configuration

    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. This parameter applies when the user has a dial tone (1st or 2nd dial tone). Enter *code (and the following target number according to current dial plan) entered at the dial tone triggers the PHONE ADAPTER to call the target number prepended by the *code.
  • Page 62 The dial plan functionality is regulated by the following configurable parameters: Interdigit_Long_Timer Interdigit_Short_Timer Dial_Plan ([1] and [2]) Enable_IP_Dialing 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: Interdigit_Long_Timer...
  • Page 63 Any one of a set of terminating events triggers the PHONE ADAPTER 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. Only one candidate sequence remains, and it has been matched completely: the number is accepted and transmitted after any transformations indicated by the dial plan, unless the sequence is barred by the dial plan (barring is discussed later), in which case the number is...
  • Page 64 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 65 ( 1 xxx xxxxxxx ) The following also allows 7-digit US-style dialing, and automatically inserts a 1 + 212 (local area code) in the transmitted number. ( 1 xxx xxxxxxx | <:1212> xxxxxxx ) For an office environment, the following plan requires a user to dial 8 as a prefix for local calls and 9 as a prefix for long distance.

  • Page 66: Speed Dialing Settings

    ( P5 <:1000> | xxxx ) Explanation of Default Dial Plan The Default Dial Plan script for each line is: “(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxx|xxxxxxxxxxxx.)” Dial Plan Entry Functionality Allow arbitrary 2 digit star code [3469]11 Allow x11 sequences Operator Int’l Operator [2-9]xxxxxx US "local" number 1xxx[2-9]xxxxxx US 1 + 10-digit long distance number xxxxxxxxxxxx.

  • Page 67: Progress Tone And Ring Configuration

    4.9. Progress Tone and Ring Configuration The progress tones and ring tones on the PHONE ADAPTER are extremely configurable. There are 18 configurable call progress tones, 8 configurable ringing cadences, and 8 configurable call waiting cadences. Progress tones and Ring cadences are configured using FreqScipts and CadScripts respectively (described in Section 4.1).
  • Page 68 Ring8 Name Name in an INVITE’s Alert-Info Header to pick Str31 Bellcore-r8 distinctive ring/CWT 8 for the inbound call Cfwd Ring Splash Duration of ring splash when a call is forwarded Time3 (0 – 10.0s) Cblk Ring Splash Duration of ring splash when a call is blocked (0 – Time3 10.0s) VMWI Ring Splash...

  • Page 69: Progress Tones

    CWT8 Cadence Cadence script for distinctive CWT 8 CadScript 2.3(..3/2) Ring Waveform Waveform for the ringing signal {Sinusoid, Sinusoid Trapezoid} Ring Frequency Frequency of the ringing signal. Valid values Uns8 are 10 – 100 (Hz) Ring Voltage Ringing voltage. 60-90 (V) Uns8 CWT Frequency Frequency script of the call waiting tone.

  • Page 70: Less Frequently Used Paramters

    Confirm Tone This should be a brief tone to notify the ToneScript 600@- user that the last input value has been 16;1(.25/.25/1)" accepted. SIT1 Tone An alternative to <Reorder Tone> played ToneScript 985@-16,1428@- when an error occurs while making an 16,1777@- outbound call.
  • Page 71 Max Redirection Number of times to allow an INVITE to be Uns8 redirected by a 3xx response to avoid an infinite loop. Note: This parameter currently has no effect: there is no limit on number of redirection. Max Auth Maximum number of times a request may be Uns8 challenged (0-255) SIP User Agent...
  • Page 72 value is larger than this, then the maximum value is used Reg Retry Intvl Interval to wait before the PHONE ADAPTER Time0 retries registration again after encountering a failure condition during last registration Reg Retry Long When Registration fails with a SIP response Time0 1200 Interval...

  • Page 73: Additional User Account Information

    carries no RR. The SDES contains CNAME, NAME, and TOOL identifiers. The CNAME is set to <User ID>@<Proxy>, NAME is set to <Display Name> (or “Anonymous” if user blocks caller ID), and TOOL is set to the Verdor/Hardware-platform-software-version (such as Linksys/PHONE ADAPTER2000-1.0.31(b)).
  • Page 74 SAS line’s own IP address is used in the c = line and a=sendrecv. In that case the SAS client will stream RTP packets to the SAS line. The default value is [empty]. SIP Debug Option None, 1-line, full, exclude OPTIONS, exclude Choice none REGISTER, exclude NOTIFY, …...

  • Page 75: Per-Line Polarity Settings

    • IVR can still be used on an SAS line, but the user needs to follow some simple steps: a) Connect a phone to the port and make sure the phone is on-hook, b) power on the PHONE ADAPTER and c) pick up handset and press * * * * to invoke IVR in the usual way.

  • Page 76: Miscellaneous Parameters

    CPC Delay Delay in seconds after caller hangs up when the PHONE ADAPTER will start removing the tip-and-ring voltage to the attached equipment of the called party. Range= 0 to 255(s) Resolution = 1 (s) CPC Duration Duration in seconds for which the tip-to-ring 0 (CPC voltage is removed after the caller hangs up.
  • Page 77 DTMF Playback Level Local DTMF playback level in dBm (up to 1 PwrLevel -10.0 decimal place) DTMF Playback Length Local DTMF playback duration in ms Time3 Detect ABCD Enable local detection of DTMF ABCD Bool Playback ABCD Enable local playback of OOB DTMF ABCD Bool Caller ID Method The following choices are available:...
  • Page 78 a) Bellcore/ETSI Onhook Post-Ring FSK First Ring b) ETSI Onhook Post-Ring DTMF First DTMF Ring c) ETSI Onhook Pre-Ring FSK/DTMF Polarity DTMF/ First Reversal (DTAS) Ring d) Bellcore Onhook FSK w/o Ring e) ETSI Onhook FSK w/o Ring Polarity Reversal (DTAS) f) Bellcore/ETSI Offhook FSK Wait For...

  • Page 79: Expected Feature Behavior

    5. Expected Feature Behavior The PHONE ADAPTER can be configured to the custom requirements of the service provider, so that from the subscriber’s point of view, the service behaves exactly as the service provider wishes – with varying degrees of control left with the end user. This means that a service provider can leverage the programmability of the PHONE ADAPTER to offer sometimes subtle yet continually valuable and differentiated services optimized for the network environment or target market(s).

  • Page 80: Caller Id

    from wired or wireless callers on the PSTN or IP network. The PHONE ADAPTER supplies ring voltage to the attached telephone set to alert the user of incoming calls. User Action Required to Deactivate or End Hang-up the telephone. 5.3. Caller ID Service Description If available, the PHONE ADAPTER supports...

  • Page 81: Calling Line Identification Restriction (Clir) - Caller Id Blocking

    effect for the duration of the current call. 5.5. Calling Line Identification Restriction (CLIR) – Caller ID Blocking Service Description This feature allows the user to block the delivery of their Caller ID to the number they are calling. This feature must be activated prior to dialing each call and is only in effect for the duration of each call.

  • Page 82: Disable Or Cancel Call Waiting

    alerting them to the second call. The person calling will hear normal ringing. User Action Required to Deactivate or End See Cancel Call Waiting. 5.7. Disable or Cancel Call Waiting Service Description The PHONE ADAPTER supports disabling of call waiting permanently or on a per call basis. User Action Required to Activate or Use To temporarily disable Call Waiting (for the length of one call):...

  • Page 83: Call-Waiting With Caller Id

    no user action is required. If you deactivated call waiting and wish to reinstate the 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.

  • Page 84: Attendant Call Transfer

    instructions to listen to your messages. Expected Call and Network Behavior When voice mail is available for a subscriber, a notification message will be sent from the Voice Mail server to the PHONE ADAPTER. When the user dials their own phone number, PHONE ADAPTER connects...

  • Page 85: Unattended Or "Blind" Call Transfer

    remain in a call. User Action Required to Deactivate or End Not applicable. 5.11. Unattended or “Blind” Call Transfer Service Description Unattended or “Blind” 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.

  • Page 86: Three-Way Calling

    Expected Call and Network Behavior User Action Required to Deactivate or End Hang-up the telephone. 5.13. Three-Way Calling Service Description The user can originate a call to a 3rd party while engaging in an active call. User Action Required to Activate or Use Press the switch hook or flash button on the phone to place the first party on hold Listen for three short tones followed by dial...

  • Page 87: Call Return

    Call the first party in the normal manner Follow the directions for adding a third party (see instructions above) Expected Call and Network Behavior The PHONE ADAPTER can host a 3-way conference and perform 3-way audio mixing (without the need of an external conference bridge device or service).

  • Page 88: Call Fwd - Unconditional

    lines are idle, the user hears a special ring. During the monitoring process the user can continue to originate and receive calls without affecting the Call Return on Busy request. Call Return on Busy requests can be canceled by dialing the deactivation code. User Action Required to Deactivate or End Lift the receiver Listen for dial tone...

  • Page 89: Call Fwd - Busy

    5.18. Call FWD – Busy Service Description Calls forwarded designated forwarding number if the subscriber’s line is busy because of the following; Primary line already in a call, primary and secondary line in a call or conference. User Action Required to Activate or Use Lift the receiver Listen for dial tone Press *__...

  • Page 90: Anonymous Call Blocking

    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. Alternatively, the user can activate this feature from a web browser interface. Note: The forward delay is entered from the web interface.

  • Page 91: Speed Calling - Up To Eight (8) Numbers Or Ip Addresses

    ringing and call waiting tone patterns to be played when incoming calls arrive. The choice of alerting pattern to use is carried in the incoming SIP INVITE message inserted by the SIP Proxy Server (or other intermediate application server in the Service Provider’s domain).

  • Page 92: Troubleshooting

    Expected Call and Network Behavior Pick up the receiver Listen for dial tone Press single digit code assigned to the stored number (2-9) Press # to signal dialing complete The number is automatically dialed normally. User Action Required to Deactivate or End None 6.

  • Page 93: Enabling Logging And Debugging

    Prompt, Confirmation, or Message-Waiting Encoder Encoder in use: G711u, G711a, G726-16/24/32/40, G729a, or G729ab Decoder Decoder in use: G711u, G711a, G726-16/24/32/40, G729a, or G729ab Indicate whether FAX pass-through mode has been initiated: Yes or No Type Indicate the call type: Inbound or Outbound Remote Hold Indicate whether the remote end has placed the call on hold: Yes or No Call Back...

  • Page 94: Provisioning And Upgrade Result Codes

    General SIP Protocol Error (e.g., unacceptable codec in SDP in 200 and ACK messages, or times out while waiting for ACK) Dialed number invalid according to given dial plan 6.5. Provisioning and Upgrade result codes The $PRVST and $UPGST macro variables expand to integer codes which report the state of a resync or upgrade attempt.

  • Page 95: Summary Of Implemented Features And Specifications

    410 Gone 412 Conditional Request Failed 413 Request Entity Too Large 414 Request-URI Too Long 415 Unsupported Media Type 416 Unsupported URI Scheme 420 Bad Extension 421 Extension Required 423 Interval Too Brief 429 Provide Referrer Identity 480 Temporarily Unavailable 481 Call/Transaction Does Not Exist 482 Loop Detected 483 Too Many Hops...

  • Page 96: Ipv4 - Internet Protocol Version 4 (Rfc 791) Upgradeable To V6 (Rfc 1883)

    7.1.2. IPv4 – Internet Protocol Version 4 (RFC 791) upgradeable to v6 (RFC 1883) 7.1.3. ARP – Address Resolution Protocol 7.1.4. DNS – A Record (RFC 1706), SRV Record (RFC 2782) 7.1.5. DiffServ (RFC 2475) and ToS – Type of Service (RFC 791/1349) 7.1.6.

  • Page 97: Secure Calls

    Negotiation of the optimal voice codec is sometimes dependent on the PHONE ADAPTER device’s ability to “match” a codec name with the far-end device/gateway codec name. The PHONE ADAPTER allows the network administrator to individually name the various codecs that are supported such that the correct codec successfully negotiates with the far end the equipment.

  • Page 98: Call Progress Tone Generation

    The PHONE ADAPTER may relay DTMF digits as out-of-band events to preserve the fidelity of the digits. This can enhance the reliability of DTMF transmission required by many IVR applications such as dial-up banking and airline information. 7.2.10. Call Progress Tone Generation The PHONE ADAPTER has configurable call progress tones.

  • Page 99: Configurable Flash / Switch Hook Timer

    The PHONE ADAPTER can signal hook flash events to the remote party on a connected call. This feature can be used to provide advanced mid-call services with third-party-call-control. Depending on the features that the service provider will offer using third-party-call-control, the following three PHONE ADAPTER features may be disabled to correctly signal a hook-flash event to the softswitch: 1.

  • Page 100: Streaming Audio Server - Sas

    a) Bellcore/ETSI Onhook Post-Ring FSK First Ring b) ETSI Onhook Post-Ring DTMF First DTMF Ring c) ETSI Onhook Pre-Ring FSK/DTMF Polarity DTMF/ First Reversal (DTAS) Ring d) Bellcore Onhook FSK w/o Ring e) ETSI Onhook FSK w/o Ring Polarity Reversal (DTAS) f) Bellcore/ETSI Offhook FSK Wait For...
  • Page 101 PA1: IP=192.168.2.100 User ID[1]=1001, SIP Port[1]=5060 User ID[2]=1002, SIP Port[2]=5061 Phone 1 Phone 2 Line CD Player, Radio, etc. Network Network PA2: IP=192.168.2.200 User ID[1]=2001, SIP Port[1]=5060 User ID[2]=2002, SIP Port[2]=5061 Phone 1 Phone 2 Example configuration for MOH application with a PHONE ADAPTER line configured as a SAS SAS Configuration Examples: The following configuration examples are based on the setup as depicted in Figure.

  • Page 102: Security Features

    SAS Enable[2] = yes On PHONE ADAPTER 2: SAS Enable[1] = no MOH Server [1] = 1002 SAS Enable[2] = no MOH Server [2] = 1002 7.3. Security Features 7.3.1. Multiple Administration Layers (Levels and Permissions) 7.3.2. HTTP Digest – Encrypted Authentication via MD5 (RFC 1321) 7.3.3.
  • Page 103 Enable_Web_Admin_Access "Yes" ; Admin_Passwd "" ; User_Password ! "" ; # *** Internet Connection Type DHCP ! "Yes" ; Static_IP ! "" ; NetMask ! "" ; Gateway ! "" ; # *** Optional Network Configuration HostName ! "" ; Domain ! ""...
  • Page 104 GPP_G "" ; GPP_H "" ; GPP_I "" ; GPP_J "" ; GPP_K "" ; GPP_L "" ; GPP_M "" ; GPP_N "" ; GPP_O "" ; GPP_P "" ; GPP_SA "" ; GPP_SB "" ; GPP_SC "" ; GPP_SD ""...
  • Page 105 G729b_Dynamic_Payload "99" ; NSE_Codec_Name "NSE" ; AVT_Codec_Name "telephone-event" ; G711u_Codec_Name "PCMU" ; G711a_Codec_Name "PCMA" ; G726r16_Codec_Name "G726-16" ; G726r24_Codec_Name "G726-24" ; G726r32_Codec_Name "G726-32" ; G726r40_Codec_Name "G726-40" ; G729a_Codec_Name "G729a" ; G729b_Codec_Name "G729ab" ; G723_Codec_Name "G723" ; # *** NAT Support Parameters Handle_VIA_received "No"...
  • Page 106 Proxy[1] "" ; Use_Outbound_Proxy[1] "No" ; Outbound_Proxy[1] "" ; Use_OB_Proxy_In_Dialog[1] "Yes" ; Register[1] "Yes" ; Make_Call_Without_Reg[1] "No" ; Register_Expires[1] "3600" ; Ans_Call_Without_Reg[1] "No" ; Use_DNS_SRV[1] "No" ; DNS_SRV_Auto_Prefix[1] "No" ; Proxy_Fallback_Intvl[1] "3600" ; Voice_Mail_Server[1] "" ; # *** Subscriber Information Display_Name[1] ""...
  • Page 107 G726-40_Enable[1] "Yes" ; FAX_Codec_Symmetric[1] "Yes" ; DTMF_Tx_Method[1] "Auto" ; # options: InBand/AVT/INFO/Auto FAX_Passthru_Method[1] "NSE" ; # options: None/NSE/ReINVITE Hook_Flash_Tx_Method[1] "None" ; # options: None/AVT/INFO FAX_Process_NSE[1] "Yes" ; Release_Unused_Codec[1] "Yes" ; # *** Dial Plan Dial_Plan[1] "(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)" ; Enable_IP_Dialing[1] "No" ; # *** FXS Port Polarity Configuration Idle_Polarity[1] "Forward"...
  • Page 108 Secure_Call_Setting[1] "No" ; # *** Distinctive Ring Settings Ring1_Caller[1] ! "" ; Ring2_Caller[1] ! "" ; Ring3_Caller[1] ! "" ; Ring4_Caller[1] ! "" ; Ring5_Caller[1] ! "" ; Ring6_Caller[1] ! "" ; Ring7_Caller[1] ! "" ; Ring8_Caller[1] ! "" ; # *** Ring Settings Default_Ring[1] ! "1"...
  • Page 109 Outbound_Proxy[2] "" ; Use_OB_Proxy_In_Dialog[2] "Yes" ; Register[2] "Yes" ; Make_Call_Without_Reg[2] "No" ; Register_Expires[2] "3600" ; Ans_Call_Without_Reg[2] "No" ; Use_DNS_SRV[2] "No" ; DNS_SRV_Auto_Prefix[2] "No" ; Proxy_Fallback_Intvl[2] "3600" ; Voice_Mail_Server[2] "" ; # *** Subscriber Information Display_Name[2] "" ; User_ID[2] "" ; Password[2] ""...
  • Page 110 DTMF_Tx_Method[2] "Auto" ; # options: InBand/AVT/INFO/Auto FAX_Passthru_Method[2] "NSE" ; # options: None/NSE/ReINVITE Hook_Flash_Tx_Method[2] "None" ; # options: None/AVT/INFO FAX_Process_NSE[2] "Yes" ; Release_Unused_Codec[2] "Yes" ; # *** Dial Plan Dial_Plan[2] "(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)" ; Enable_IP_Dialing[2] "No" ; # *** FXS Port Polarity Configuration Idle_Polarity[2] "Forward"...
  • Page 111 # *** Distinctive Ring Settings Ring1_Caller[2] ! "" ; Ring2_Caller[2] ! "" ; Ring3_Caller[2] ! "" ; Ring4_Caller[2] ! "" ; Ring5_Caller[2] ! "" ; Ring6_Caller[2] ! "" ; Ring7_Caller[2] ! "" ; Ring8_Caller[2] ! "" ; # *** Ring Settings Default_Ring[2] ! "1"...
  • Page 112 CWT8_Cadence "2.3(.3/2)" ; # *** Distinctive Ring/CWT Pattern Names Ring1_Name "Bellcore-r1" ; Ring2_Name "Bellcore-r2" ; Ring3_Name "Bellcore-r3" ; Ring4_Name "Bellcore-r4" ; Ring5_Name "Bellcore-r5" ; Ring6_Name "Bellcore-r6" ; Ring7_Name "Bellcore-r7" ; Ring8_Name "Bellcore-r8" ; # *** Ring and Call Waiting Tone Spec Ring_Waveform "Sinusoid"...

  • Page 113: Acronyms

    Dist_Ring_Act_Code "*26" ; Dist_Ring_Deact_Code "*46" ; Speed_Dial_Act_Code "*74" ; Secure_All_Call_Act_Code "*16" ; Secure_No_Call_Act_Code "*17" ; Secure_One_Call_Act_Code "*18" ; Secure_One_Call_Deact_Code "*19" ; Referral_Services_Codes "" ; Feature_Dial_Services_Codes "" ; # *** Outbound Call Codec Selection Codes Prefer_G711u_Code "*017110" ; Force_G711u_Code "*027110" ; Prefer_G711a_Code "*017111"...
  • Page 114 CIDCW Call Waiting Caller ID Comfort Noise Generation Calling Party Control Customer Premises Equipment CWCID Call Waiting Caller ID Call Waiting Tone Digital to Analog Converter decibel dB with respect to 1 milliwatt DHCP Dynamic Host Configuration Protocol Domain Name Server DRAM Dynamic Random Access Memory Digital Subscriber Loop...

  • Page 115: Glossary

    Round Trip Time Streaming Audio Server Session Description Protocol SDRAM Synchronous DRAM seconds Session Initiation Protocol SLIC Subscriber Line Interface Circuit Service Provider PAP2 Phone Adaptor Ports 2 (Linksys Phone Adaptor) Secure Socket Layer TFTP Trivial File Transfer Protocol Transmission Control Protocol User Agent Micro-controller User Datagram Protocol...
  • Page 116 Circuits: The communication path(s) that carry calls between two points on a network. Customer Premise Equipment: The only part of the telecommunications system that the customer comes into direct contact with. Example of such pieces of equipment are: telephones, key systems, PBXs, voicemail systems and call accounting systems as well as wiring telephone jacks.
  • Page 117 © 2004 Linksys Proprietary (See Copyright Notice on Page 2)

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