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Un buon manuale d’uso
Le regole impongono al rivenditore l'obbligo di fornire all'acquirente, insieme alle merci, il manuale d’uso Cisco Systems RT31P2. La mancanza del manuale d’uso o le informazioni errate fornite al consumatore sono la base di una denuncia in caso di inosservanza del dispositivo con il contratto. Secondo la legge, l’inclusione del manuale d’uso in una forma diversa da quella cartacea è permessa, che viene spesso utilizzato recentemente, includendo una forma grafica o elettronica Cisco Systems RT31P2 o video didattici per gli utenti. La condizione è il suo carattere leggibile e comprensibile.
Che cosa è il manuale d’uso?
La parola deriva dal latino "instructio", cioè organizzare. Così, il manuale d’uso Cisco Systems RT31P2 descrive le fasi del procedimento. Lo scopo del manuale d’uso è istruire, facilitare lo avviamento, l'uso di attrezzature o l’esecuzione di determinate azioni. Il manuale è una raccolta di informazioni sull'oggetto/servizio, un suggerimento.
Purtroppo, pochi utenti prendono il tempo di leggere il manuale d’uso, e un buono manuale non solo permette di conoscere una serie di funzionalità aggiuntive del dispositivo acquistato, ma anche evitare la maggioranza dei guasti.
Quindi cosa dovrebbe contenere il manuale perfetto?
Innanzitutto, il manuale d’uso Cisco Systems RT31P2 dovrebbe contenere:
- informazioni sui dati tecnici del dispositivo Cisco Systems RT31P2
- nome del fabbricante e anno di fabbricazione Cisco Systems RT31P2
- istruzioni per l'uso, la regolazione e la manutenzione delle attrezzature Cisco Systems RT31P2
- segnaletica di sicurezza e certificati che confermano la conformità con le norme pertinenti
Perché non leggiamo i manuali d’uso?
Generalmente questo è dovuto alla mancanza di tempo e certezza per quanto riguarda la funzionalità specifica delle attrezzature acquistate. Purtroppo, la connessione e l’avvio Cisco Systems RT31P2 non sono sufficienti. Questo manuale contiene una serie di linee guida per funzionalità specifiche, la sicurezza, metodi di manutenzione (anche i mezzi che dovrebbero essere usati), eventuali difetti Cisco Systems RT31P2 e modi per risolvere i problemi più comuni durante l'uso. Infine, il manuale contiene le coordinate del servizio Cisco Systems in assenza dell'efficacia delle soluzioni proposte. Attualmente, i manuali d’uso sotto forma di animazioni interessanti e video didattici che sono migliori che la brochure suscitano un interesse considerevole. Questo tipo di manuale permette all'utente di visualizzare tutto il video didattico senza saltare le specifiche e complicate descrizioni tecniche Cisco Systems RT31P2, come nel caso della versione cartacea.
Perché leggere il manuale d’uso?
Prima di tutto, contiene la risposta sulla struttura, le possibilità del dispositivo Cisco Systems RT31P2, l'uso di vari accessori ed una serie di informazioni per sfruttare totalmente tutte le caratteristiche e servizi.
Dopo l'acquisto di successo di attrezzature/dispositivo, prendere un momento per familiarizzare con tutte le parti del manuale d'uso Cisco Systems RT31P2. Attualmente, sono preparati con cura e tradotti per essere comprensibili non solo per gli utenti, ma per svolgere la loro funzione di base di informazioni e di aiuto.
Sommario del manuale d’uso
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Pagina 1
PHONE ADAPTER Administration Guide August 2004 Linksys PAP2 and RT31P2 © 2004 Linksys Proprietary (See C opyright Notice on Page 2) 1[...]
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Pagina 2
Disclaimer – Please Read: This document contains i mplementation example s and techniques using Lin ksys and, in some instance s, other company’s technology and pro ducts and is a recommendation only and doe s not constitute any legal arrangement b etween Linksys and the reader, either written or implied. The conclu sions reached and recommenda[...]
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Pagina 3
Table of Contents 1. Introduction .................................................................................................................................... 6 1.1. The Session Initiation Protocol ............................................................................................. 6 1.1.1. Components of a SIP Network .............[...]
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Pagina 4
4.9. Progress Tone and Ring Configuration .............................................................................. 67 4.9.1. Distinctive Ring and Other Ring Settings ......................................................................................67 4.9.2. Progress Tones ...................................................................[...]
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Pagina 5
7.2.1.3. SIP Support in Network Address Translatio n Networks – NAT ............................................96 7.2.2. Codec Name Assignment ..............................................................................................................96 7.2.3. Secure Calls .......................................................................[...]
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Pagina 6
1. Introduction This guide describes ba sic administrati on and use of the Linksy s Technology PHO NE ADAPTER phone adapter – an intelligent low-density Voice ov er IP (VoIP) gateway. The PHONE ADAPTER enables carri er class resi dential and business IP T elephony services deliv ered over broadban d or high-speed Internet conne ctions. By intelli[...]
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Pagina 7
• SIP enables the implementation of intelligent endpoint s to support scalable advance d services. In a nutshell, SIP is a dist ributed signaling p rotoc ol (as opposed to a centrali zed protocol such as SS7, MGCP or MEGACO/H.248 ). With a distributive proto col, the intelligence does not necessarily reside on a central se rver, but can be b uilt[...]
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Pagina 8
1.1.1. Components of a SIP Network SI P Proxy Server PSTN Ga t eway Route r NAT Prov is i oni ng Se r v er PC PC Ap p lic at i on Ser v er PSTN IS P Ga t eway PST G N a t eway P Net r iva t e IP wor k PSTN Su b sc ri b e r Se r vice Dat ab ase Provider Bill ing Se r v er Doma i n Ap p lic a t i on Se r v er Br oad b and Mod e m S u bsc ri b er Do m[...]
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Pagina 9
gateways, etc. The default router in this case is the IP addre ss of the LAN interface of the router itself. • Performs NAT on packets sent from the privat e network to the public netwo rk. This is an important feature su ch that recipients of t he private packe ts will perceive them as originated from a public IP address (t he router’s WAN int[...]
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Pagina 10
ADAPTER unit is shipped from the factory, it c ontai ns a default common Profile and is consid ered Unprovisioned. To save cost s and expedite deliv ery, however, it is very desirable that an e PHONE from the provisioning serv er is very scalable and flexible. Using this provisio ning method, a TER units can be provisioned simultaneou sly and updat[...]
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Pagina 11
• The SIP signaling me ssages – The SIP messag es exchanged between th e SIP proxy server and the PHONE ADAPTER should be encrypted with a secret key. This can be achieved, for instance, by transporting S I P over TLS. • RTP packets – The RTP payload ex changed between SIP user agent s can be encrypted with a secret key to protect against e[...]
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Pagina 12
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 hand set, the PHONE ADAPTER provides dial tone and is ready to collect diali ng informat ion via DTMF digits from a touch tone telephone. While it is possible to support ove rlapped dialin g within the context of SIP, [...]
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Pagina 13
1.1.4.2.3. Voice Mail Message Waiting Indication Service Providers may provide voice mail service to their sub scribers. When voice mail is av ailable for a subscriber, a notification message will be sent from the Vo ice Mail server to the PHONE ADAPTER. The PHONE ADAPTER indicate s that a mess age is waiting by, playing stuttered dial tone (or oth[...]
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Pagina 14
telephone number to forward calls to. The PHO NE AD APTER provides audio instructions to prom pt the user for a forwarding numbe r and confirms t hat the requested service has b e en activated. Call FWD – Unconditional All calls are immediately forwarded to the desig nated forwarding numbe r. The PHONE ADAPTER will not ring or provide call waitin[...]
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Pagina 15
If the service provider is offering originat ion and/or termination on endpoint equipment then it is very likely that the softswitch chosen for network ope r ations will support multiple PSTN and VoIP signaling protocols. The table below lists the most commonly accepted, de-facto standards used whe n implementing a VoIP signaling scheme based on th[...]
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Pagina 16
address/port to the correspondin g private source address/port. T h ese characteristics of a NA T can be exploited by an PHONE ADAPTER to l et external enti ties send SIP messag es and RTP packets to it when it is installed on a priv ate network. 1.2.2. VoIP-NAT Interworking In the case of SIP, the addresses where messag es/d ata should be sent to [...]
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Pagina 17
Several factors that contribute to Voice Quality are described bel ow. Audio compression algorithm – Speech signals are sampled, quantized and compressed before they are packetized and transmitted to the other end. For IP Telep hony, speech signals are usually sampled at 8000 samples per second with 12-16 bits per sample. The compression algorith[...]
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Pagina 18
Figure 2 – PAP2 Front Figure 3 – PAP2 Back Figure 3 – RT31P2 Front Fig ure 4 – RT31P2 Back The PAP2 PHONE ADAPTER has the following interfac es for networking, po wer and visual status indication: 1. Two (2) RJ-11 Type Analog Telephone Ja ck Interfaces (Figure 3 , above): These interfaces accept standard RJ-11 telephone connectors. An Analo[...]
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Pagina 19
2.1. Phone Adapter LED Status LED Color(s) Acti vity Description Off Power OFF Blue On Power On / Device Ready Blue Blinking Booting / System Self-Test / Firmware upgrade Power Blue Red On POST (Power On Self Test) failure (not boota b le) or Device m alfunction Off No Connection on Ether net Blue On Ethernet Co nnection esta blished Ethernet Blue [...]
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Pagina 20
Please check to make sure that y ou have the following package contents: 1. Linksys Phone Adapter Unit or Linksys Broadban d Route r (RT31P2) 2. Ethernet Cable 3. 5 Volt (PAP2) or 12 Volt (RT31P2) Power Adapter 4. CD with User Guide You will also need: 1. One or Two Analog Touch Tone Telephones (or F ax Machine) 2. Access to an IP Network via an Et[...]
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Pagina 21
The PHONE ADAPTER al so provides a Web Interfac e with two-level access (u ser-level and admin- level) to configuration parameters. Fo r standalone PHONE ADAPTERS (whi ch contain no router or NAT functionality), an IVR (Interactive Voice Respon se) interfac e is also available for configuring basic networking. 3.1. Configuration Profile Formats The[...]
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Pagina 22
respectively. If neither mark is present, the paramet er is made inaccessible to the use r from the web interface. Note that this syntax has no effe ct on the admin-level access to the parameters. When using the SPC, a service provide r is giv en full control over which p arameters become inaccessible, read-only, or read -write fo llowin g provi si[...]
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Pagina 23
# 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 acce ss settings for the PHONE ADAPTER can be obtained from the profile comp iler[...]
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Pagina 24
An encrypted CFG file requires either a passwor d (or quoted pass-phrase ) or a hex-st ring. The following lines illustrate command-line invocatio ns 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 phra[...]
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This utility generates 8-bytes of salt (which is pr epended to the encrypted configuration file), and then calculates an Initialization Vector (IV) a nd an 25 6-bit encryption key using the key phrase prov ided on the command line. The TA recognizes th e leading char acters "Salted__" as a hint to find the salt and decrypt the configurati[...]
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Pagina 26
Within this directory, the Apache module mod_ssl ve ri fies the clien t certificate, a nd verifies that the MAC address in the certificate corres ponds the config uration file it is req uesting. Either this directory must contain a configuration file, or a CGI applicatio n needs to generate the ap p ropriate config file if that MAC address is confi[...]
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Pagina 27
NOTE : Pressing the “ Undo All Changes ” has no effect on the P HONE ADAPTER; it will only reset the values on the web page. 3.3.2. Administration Privileges The PHONE ADAPTER support s two levels of administ ration privil eges: Administrator and User, both privileges can be password protected. Important note : by factory default, there are no [...]
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Pagina 28
http://<PAP2-ip-addr>/admin/ upgrade?[protocol://][server-nam e[:port]][/firmware-pathname ] If no protocol is specified, TFTP is assumed. Note: Only TFTP is supported in the current relea se. If no server-name is specified, the host that requests the URL is us ed as server-name. If no port specified, default port of the protocol is used. (69[...]
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3.5. Configuration vi a the IVR (PAP2 only) Administrators and/or users can che ck (read) and se t (write) basic network configuration settings via a touchtone telephone conne cted to one of the RJ-11 phone ports of the PAP2 model PHONE ADAPTER. Please Note: Service Providers offering service using the PHONE AD APTER may restrict, protect or turn o[...]
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Pagina 30
3. After one minute of inactivity, the unit times ou t. The user will need to re-enter the co nfiguration menu from the beginning by pressing * * * * . 4. If, while entering a value (like an IP address) and you decide to exit wit hout entering any change s, you may do so by pressing the * (star) key twice w ithin a half second window of time. Other[...]
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PHONE ADAPTER. Set Network Mask 121 Enter value using numbers on the telephone key pad. Use the * (star) key when entering a decimal point. DHCP must be “Disabled” otherwise you will hear, “Invalid Option,” if you try to set this value. Requires Pass w ord Check Static Gateway IP Address 130 None IVR will announce the current gateway IP add[...]
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User Factory Reset of Unit WA RN ING : ALL “Us er-Change able” NON - DEFAULT SETTINGS WILL BE LOST! This might include ne twork and service provider data. 877778 Enter 1 to confirm Enter *(star) to cancel operation PHONE ADAPTER will prompt for confirmation. After confirming, you will hear “Option Successful.” Hang-up. Unit will reboot and [...]
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Pagina 33
wildcard characters. It can contain up to 39 cha racters. Examples: “1408*, 1510* ”, “1408123????, 555?1 ”. • RscTmplt – A template of SIP Re sponse 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 caden ce parameters of a sign[...]
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Pagina 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 frequ ency, level and cadence param eters of a call progress tone. May contai n up to 127 characters. Syntax: FreqScript;Z 1 [;Z 2 ]. The section Zi is similar to the S i section in a CadScript e xcept th[...]
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Pagina 35
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 – Scriptin g syntax used to define configuration resync and firmware upgrade rules. Refer to the provisioning di scussi on in the next section fo r a [...]
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Pagina 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). The func tio[...]
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active call is in progress. T he 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 re try with a delay indica[...]
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Pagina 38
These strings each support s one level of macro expansion, using a small set of variabl es. 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 def ault values a re assumed for the unspecified terms. The filepath portion[...]
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Pagina 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 [":" port]]] "/"[...]
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In addition, the contents of the general purpose parameters, GPP_A, through GPP_P, are available as macro variables A through P, respecti vely. A secondary set of general purpo se 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 access[...]
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Pagina 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 indi vidualizes each PHONE ADAPTER into fetching a unique CFG file) could be pre configured during manuf acturi ng. Fo[...]
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Pagina 42
string supports one level of macro su bsti tution, 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 Pa rameters, usable by bot h the provisioning and the upgrade logic. Each[...]
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Log Resync Request Msg Syslog message generated when attempting a resync ProfileMsg See provisioning discussion section Log Resync Success Msg Syslog message generated after a successful re sync ProfileMsg See provisioning discussion section Log Resync Failure Msg Syslog message generated after a failed resync ProfileMsg See provisioning discussion[...]
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Pagina 44
The firmware file must be requested by the PHONE ADAPTER and cann ot be pushed from an upgrade server (although a servi ce provider can e ffectively push a new firm ware load by trig gering the request operation remotely via the CFG file ). The functionality is cont rolled by the Upgrade_Enable param eter. The parameter enables the functionality en[...]
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The upgrade will fail if the new firmware load doe s not satisfy the upgrade rule condition that suggested the URL. This alleviates the possibility of infinite upgrade loops, in ca se the devi ce has been misconfigured. The rule syntax is the same as for the P rofile_Rule described in a previous section, except that there are no supported optional [...]
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string supports one level of macro su bsti tution, with the same variables as for the Upg rad e_ Rule above. An empty string does not generate a syslog message. Parameter Name Description Typ e Default Upgrade Enable Master ena bl e for firmware upgrade operations Bool Yes Upgrade Error Retry Delay Retry interval following upgrade failure Time0 360[...]
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Protect IVR Factory Reset Bool No Admin Password The password for administrator Str63 User Password The password for User Str63 4.3. Basic Networking Configuration Configuration parameters in this list are used fo r setting up basic network conn e ctivity. In general, many of these parameters are set automatically (for example, using DHCP) or are c[...]
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- Parallel DNS query mode: PHONE ADAPTER will se nd 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 a re al so logge d to[...]
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Pagina 49
parameter is useful only if the primary an d backu p proxy server list is provided to the PHONE ADAPTER via DNS SRV record lookup on the server name. (Using multiple DNS A re cord per server name does not allow the notion of priority and so all hosts will be considere d at the same priority and the PHONE ADAPTER will not attempt to fall back after [...]
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Pagina 50
ADAPTER. Also, set the Substitute_VIA_Addr and NAT_Mapping_Enable p ara meters. Follow the instructions of the NAT software to configure st atic NAT mappings between the external address and ports (EXT_SIP_Port, EXT_RTP_Port_Min) and the internal add re ss a nd ports (SIP_Port, RTP_Port_Min). Set the RTP_Port_Max param eter to a smaller nu mber (fo[...]
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Ext SIP Port External port to subs titute for the actual SIP port of the unit in all outgoing SIP messages. If “0” is specified, no SIP port substitution is performe d. Port 0 Ext RTP Port Min External port mapping of <RTP Port Min>. If this value is non-zero, the RTP port number in all outgoing SIP messages is substituted by the correspo[...]
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The administrator can sele ct a method f or convey ing DTMF and ho okfla sh on a per-line basis. In addition, the administrator can also confi gure the MIME type (Content-Type header) u se d wh en conveying DTMF or hookflash in SIP INFO mess ages. The MIME type is set once for both line s. DTMF Tx Method Method to transmit DTMF signals to the far e[...]
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FAX CED Detect Enable Enable det ect ion of FAX tone. Bool Yes FAX CNG Detect Enable Bool Yes FAX Passthru Codec Codec to use for fax passthru {G711u, G711a} G711u FAX Codec Symmetric Force unit to use symmetric codec duri ng FAX passthru Bool Yes FAX Passthru Method Choices: None / NSE / ReINVITE Choice NSE FAX Process NSE Bool Yes Release Unused [...]
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G726r40 Codec Name G726-40 Codec name used in SDP Str31 G726-40 G729a Codec Nam e G729a Codec name u sed in SDP Str31 G729a G729b Codec Nam e G729b Codec name u sed in SDP Str31 G729ab G723 Codec Name G723 Codec name used in SDP Str31 G723 Notes: 1. PHONE ADAPTER uses the configured codec na mes in its out bound SDP 2. PHONE ADAPTER ignores the cod[...]
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- 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 oth er. The public ke y of the signing agent is pre-configured into the PHONE ADAPTER’s by the administrator and will be used by the PHONE ADAPT[...]
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Pagina 56
e3VgYxWCQNa335YCnDsenASeBxuMIEaBCYd1l1f VEodJZ OGwXwfAde0MhcbD0 kj7LVlzcsTyk2TZ YTccnZ75TuTjj13qvYs= 5nEtOrkCa84/mEwl3D9tSvVLyliw Q+u/Hd+C8u5SNk7hsAUZaA9TqH8Iw0J/IqSrsf6scsmundY5 j7Z5m K5J9uBxSB8t8vamFGD0pF4zhNtbrVvIXKI9kmp4v ph1C5jzO9gDfs3MF+zjyYrV UFdM+pXtDBxmM+f GUfrpAuXb7/k= - user-name is the name of the subscriber, su ch as “Joe Smith”. M[...]
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Prefer G723 Code Dialing code will make this codec the preferred codec for the associated ca ll. ActCode *01723 Force G723 Code Dialing code w ill make this codec the only codec that can be used for the associate d call. ActCode *02723 Prefer G726r16 Code Dia ling code will make this codec the preferred codec for the associated ca ll. ActCode *0172[...]
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CWCID Serv Enable Call Waiting Caller ID Service Bool Yes Call Return Serv Enable Call Return Service Bool Yes Call Back Serv Enable Call Back Service Bool Yes Three Way Call Serv 1 Enable Three Way Calling Service Bool Yes Three Way Conf Serv 1,2 Enable Three Way Conference Se rvice Bool Yes Attn Transfer Serv 1,2 Enable Attended Call Transfer Ser[...]
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Cfwd Last Deact Code Cancel call forward last ActCode *83 Block Last Act Code Block the last inbound call ActCode *60 Block Last Deact Code Cancel blocking of the last inbound call ActCode *80 Accept Last Act Code Accept the last outbound call. Let it ring through when DND or Call Forward All is in effect ActCode *64 Accept Last Deact Code Can cel [...]
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Pagina 60
Secure Call Setting If yes, all outbound calls are secure calls by default Bool No 4.7.2. Call Forwarding Implemented internally The PHONE ADAPTER supports local call forwardi ng services (Call Forward All, Call Forward Busy, Call Forward No Answer, and Selective Call Fo rwarding for up to 8 numbers). Parameter Name Description Typ e Default Cfwd A[...]
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One or more *code can be configured into this param eter, 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 (1 st or 2n d dial tone). Enter *code (and the following target number accord ing to current dial plan) entered at the dial tone triggers the PHONE ADAPTER to call the ta[...]
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The dial plan functionality is regulated by the following configura bl e parameters: • Interdigit_Long_Timer • Interdigit_Short_Timer • Dial_Plan ([1] and [2]) • Enable_IP_Dialing Other timers are configurable via param eters, but do not directly pertain to the dial plan itself. They are discussed elsewhere i n this document. Interdigit Lon[...]
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Any one of a set of terminating events triggers the PHONE ADAPTER to either accept the user-di aled sequence, and transmit it to initiate a call, or else reject it as invalid. The terminating events are: • No candidate sequence s remain: the number is reject ed. • Only one candidate sequence remains, and it ha s been matched completely : the nu[...]
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Interdigit Timer Master Overri de: The long and short interdigi t timers ca n be changed in the dial plan (affecting a specific lin e) by preceding the entire plan with the followi ng syntax: • Long interdigit timer: ‘L’ ‘:’ delay-value ‘,’ • Short interdigit timer: ‘S’ ‘:’ delay-value ‘,’ Thus, “L=8,( . . . )” would[...]
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( 1 xxx xxxxxxx ) The following also allows 7-digit US-style dialing, an d automatically inserts a 1 + 212 (lo cal 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. In either case, an[...]
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( 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|1 xxx[2-9]xxxxxx|x xxxxxxxxxxx.)” Dial Plan Entry Functionality *xx Allow arbitrary 2 digit star code [3469]11 Allow x11 sequen ces 0 Operator 00 Int’l Operator [2-9]xxxxxx US "local" number 1xx[...]
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4.9. Progress Tone and Ring Configuration The progress tones and ring tones on the PHONE ADAPTER are extremely configurable. The re are 18 configurable call progress ton e s, 8 configurable ringing cadenc es , and 8 configurable call waiting cadences. Progress tone s and Ring cadences are configured usi ng FreqScipts and CadScrip ts respectively (d[...]
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Ring8 Name Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 8 for the inbound call Str31 Bellcore-r8 Cfwd Ring Splash Len 2 Duration of ring splash when a call i s forwarded (0 – 10.0s) Time3 0 Cblk Ring Splash Len 2 Duration of ring splash when a call i s blocked (0 – 10.0s) Time3 0 VMWI Ring Splash Len Duration of ring spl[...]
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CWT8 Cadence Cadence script for distinctive CWT 8 CadScript 2.3(..3/2) Ring Waveform Waveform for the ringing signal {Sinusoid, Trapezoid} Sinusoid Ring Frequency Frequency of the ringing signal. Valid values are 10 – 100 (Hz) Uns8 25 Ring Voltage Ringing voltage. 60-90 (V) Uns8 70 CWT Frequency Frequency script of the call waiting tone. All dist[...]
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Confirm Tone This should be a brief tone to notify the user that the last input value has been accepted. ToneScript 600@- 16;1(.25/.25/1)" SIT1 Tone An alternative to <Reorder Tone> played when an error occurs while making an outbound call. The RSC to trigger this tone is configurable (see Section ???) ToneScript 985@-16,1428@- 16,1777@-[...]
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Max Redirection Number of times to allow an INVITE to be redirected by a 3xx response to avoid a n infinite loop. Note: This parameter curren tly has no effect: there is no limit on number of redirec tion. Uns8 5 Max Auth Maximum number of times a request may be challenged (0-255 ) Uns8 2 SIP User Agent Name User-Agent Header to be used by the unit[...]
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value is larger than this, then the maximum value is used Reg Retry Intvl Interval to wait before the PHONE ADAPTER retries registration agai n after encountering a failure condition during last regist ration Time0 30 Reg Retry Long Interval When Registration fails with a SIP response code that does no match <Retry Reg RSC>, the PHONE ADAPTER[...]
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carries no RR. The SDES contains CNAME, NAME , and TOOL identifiers. The CNAME is set to <User ID>@<Proxy>, NAME is set to <Dis play Name > (or “Ano nymous” if user blocks caller ID), and TOOL is set to the Verdor/Hardware-platfo rm-software-version (su ch as Linksys/PHONE ADAPTER2000-1.0.31(b)). The NTP timestam p used in the[...]
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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 REGISTER, exclude NOTIFY, … Choice none Network Jitter Level 4 settings are av ailable: very high, high, medium, low. T[...]
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• IVR can still be used on an SAS line, but the user needs to follow some simp l e steps: a) Connect a phone to the port and make sure the phone is on-hook, b) po wer on the PHONE ADAPTER and c) pick up handset and press * * * * to invoke IVR in t he usual way. The idea behind this is that if the PHONE ADAPTER boot s up and finds that the SAS lin[...]
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CPC Delay 3,4 Delay in seconds after call er 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) 2 CPC Duration 3,4 Duration in seconds for which the tip-to -ring voltage is removed after the caller hangs up. After that tip-to-ring voltage [...]
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DTMF Playback Level Local DTMF playback level in dBm (up to 1 decimal place) PwrLevel -10.0 DTMF Playback Length Local DTMF playback duration in ms Time3 .1 Detect ABCD Enable local detection of DTMF ABCD Bool Yes Playback ABCD Enable local playback of OOB DTMF ABCD Bool Yes Caller ID Method The following choices are available: • Bellcore (N.Amer[...]
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Polari ty Reversal First Ring CAS (DTAS) DTMF/ FSK Polari ty Reversal CAS (DTAS) FSK CAS (DTAS) Wait For ACK FSK First Ring FSK OSI FSK a) Bell core/ ETSI Onho ok Post- Ring FSK d) Bellcore Onhook FSK w/o Ring f) Bellcore/ETSI Offh ook FSK c) ETSI On hook Pre- Ring F SK/DTM F e) ETSI Onhook FS K w/o Ring DTMF b) ETSI Onhook P ost-Ring DTMF First Ri[...]
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5. Expected Feature Behavior The PHONE ADAPTER can be config ured to the custom requirements of the service provider, so that from the subscriber’s point of view, the service b ehav es exactly as the service provider wishe s – with varying degrees of control left with the end user. Thi s means that a service provi der can leverage the programma[...]
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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 availabl e, the PHONE ADAPTER supports the generation and pass through of Caller I[...]
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effect for the duration of the current call. 5.5. Calling Line Identification Restri ction (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 m ust be activated prior to dialing each call and is only in effect for the duration of each c[...]
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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 sup ports disabling of call waiting permanently or on a per call basis. User Action Required to Acti vate or Use To temporarily di[...]
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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 W aiting has been accepted. 5.8. Call-Waiting with Caller ID Service Description When the user is on the phone and h[...]
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instructions to listen to your messages. Expected Call and Network Behavior When voic e mail is available for a subscriber, a notification message will be sent from the Voice Mail server to the PHONE ADA PTER. When the user dials their own phon e number, the PHONE ADAPTER connects the subscriber their voice m ail system which can then connect them [...]
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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 Toucht one phone to send a call to any other phone, in side or outside thei r business, incl uding a wireless phones. User Action Required to [...]
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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 Acti vate or Use Press the switch hook or flash button on the phone to place the first party on hold Li[...]
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Call the first party in the normal manner Follow the directions for adding a thi rd party (see instructions a bove) 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). If you also have Call Transfer you can also hang up[...]
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lines are idle, the user hears a special ring. During the monitoring p rocess the user can continue to originate an d receive calls without affecting the Call Return on Busy reque st. Call Return on Busy requests can be canceled by dialing the deactivation code. User Action Required to Deacti vate or End Lift the receiv er Listen for dial tone Pres[...]
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5.18. Call FWD – Busy Service Description Calls are forwarded to the desi gnated forwarding number if the sub scriber’s line is busy because of the following; Prima ry line already in a call, primary and secondary line in a call or conference. User Action Required to Acti vate or Use Lift the receiver Listen for dial tone Press *__ Listen for d[...]
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Listen for dial tone and enter the tel ephone number you are forwarding your call to. Activation will be confirm ed with three short bursts of tone and your forwarding will be activated. Alternatively, the user c an activate this feature from a web browser interface. Note: The forward del ay is entered from the web interface. Default is 20s Expecte[...]
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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 int ermediate application server in the Service Provider’s domain). User Action Required to Activa te or Use Pick up the receiver Listen for [...]
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Expected Call and Network B ehavior Pick up the receiver Listen for dial tone Press single digit code a ssigned to the stored number (2-9) Press # to signal dialing complete The number is automatically dialed norm ally. User Action Required to Deactivate or End None 6. Troubleshooting 6.1. Call Statistics Reporting The following lists the statistic[...]
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Prompt, Confirmation, or Message-Waiting Encoder Encoder in use: G711u, G711a, G726-16/24/32/40, G7 29a, or G729ab Decoder Decoder in u se: G71 1u, G 711a, G7 26-16/24/32/40, G729a, or G729ab FAX Indicate whether FAX pass-through mo de has been initiated: Yes or No Type Indicate the call type: Inbound or Outbound Remote Hold Indicate whether the re[...]
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X40 General SIP Protocol Error (e.g., una cceptable codec in SDP in 200 and ACK messages, or times ou t while waiting for ACK) X60 Dialed number invalid a ccording to given dial plan 6.5. Provisioning and Upgrade result codes The $PRVST and $UPGS T macro variables expand to integer codes which report the state of a resync or upgrade attempt. They a[...]
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410 Gone 412 Conditional Request Failed 413 Request Entity Too Large 414 Request-URI Too Long 415 Unsupported Media Type 416 Unsupported URI S chem e 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 484 Ad[...]
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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. DHCP Client – Dynamic Host Configuration Protocol (RFC 2131) 7.1.7. ICMP – Internet C[...]
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Negotiation of the optimal voice codec is sometim es dependent o n the PHONE ADAPTER device’ s ability to “match” a codec name with the far- end device/gat eway code c name. The PHONE ADAPTER allows the network admini strator to individually name the vari ous codecs that are supported such that the corre ct codec succe ssfully negotiate s wit[...]
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The PHONE ADAPTER may rel ay DTMF digits as o ut-of-band events to pre serve the fidelity of the digits. This can enhance the reliability of DTMF tran smission re quired by many IVR applications such as dial-up banking and ai rline information. 7.2.10. Call Progress Tone Generation The PHONE ADAPTER has configurab le call progress tones. Pa rameter[...]
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The PHONE ADAPTER can signal hook flash events to the remote party on a co nnected call. This feature can be used to provide advan ced mid-call serv ices with third-pa rty-call-control. Depending on the features that the service provider will offer using third-part y-call-control, the following three PHONE ADAPTER features may be disabled to correc[...]
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First Ring FSK a) Bellco re/ETSI Onhook Post-Ring F SK Polarity Reversal First Ring CAS (DTAS) DTMF/ FSK Polarity Reversal CAS (DTAS) FSK CAS (DTAS) Wait For ACK FSK OSI FSK d) Bellcore Onhook FSK w/o Ring f) Bellcore/ET I Offh ook FSK S c) ETSI Onhook Pre-Ring FSK/DTMF e) ET SI Onhook FS K w/o Ring DTMF b) ETSI Onhook Post- Ring DTMF First Ring PH[...]
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MSA CD Player, Ra di o, e tc . Lin e In PA1: 2.100 IP=192.168. User ID[1]=1001, User ID[2]=1002, Phone 1 Phone 2 Phone 1 IP N etwor k IP N etwor k Phone 2 SIP Por t[1]=5060 SIP Por t[2]=5061 2.168.2.200 D[1]=2001, SIP Por t[1]=5060 MOH application with a PHONE ADAPTER line configured as a S AS : examples are based on the setup as depicted in Figu r[...]
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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) dministration and Maintenance Features 4.1. Web Browser Administration and [...]
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Enable_Web_Admin_Access "Yes" ; Admin_Passwd "" ; User_Password ! "" ; # *** Internet Connection Type DHCP ! "Yes" ; " ; onfiguration Sequential bug_Level "0" ; # options: 0/1/2/3 "" ; condary_NTP_Server "" ; ; ; ; ; " ; ; ; .cfg" ; Profile_Rule_C "" ; MA[...]
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GPP_G "" ; GPP_H "" ; GPP_I "" ; GPP_J "" ; GPP_K "" ; GPP_L "" ; GPP_M "" ; ; P_O "" ; *** SIP Parameters x_Forward "70" ; ; ON" ; MF_Relay_MIME_Type "application/dtmf-relay" ; "application/hook-flash" ; move_Last_Reg "No" ;[...]
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G729b_Dynamic_Payload "99" ; E_Codec_Name "NSE" ; ne-event" ; 26r24_Codec_Name "G726-24" ; "G726-32" ; 26r40_Codec_Name "G726-40" ; rameters UN_Test_Enable "No" ; "" ; T_IP "" ; *** ne_Enable[1] "Yes" ; FY" ; T_Keep_Alive_Dest[1] "$PROXY" ;[...]
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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] &qu[...]
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G726-40_Enable[1] "Yes" ; ; ; # options: InBand/AVT/INFO/Auto ; # options: None/NSE/ReINVITE ; # options: None/AVT/INFO ; ; al_Plan[1] "(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)" ; "No" ; rd" ; # options: Forward/Reverse rd" ; # options: Forward/Reverse rd" ; # options: Forward/Reve[...]
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Secure_Call_Setting[1] "No" ; # *** Distinctive Ring Settings Ring1_Caller[1] ! "" ; ! "" ; ng3_Caller[1] ! "" ; ng7_Caller[1] ! "" ; ! "" ; # options: 1/2/3/4/5/6/7/8 options: 1/2/3/4/5/6/7/8 lk_Ring_Splash_Len[1] ! "0" ; ! ".5" ; WI_Ring_Policy[1] "New VM Availab[...]
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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_[...]
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DTMF_Tx_Method[2] "Auto" ; # options: InBand/AVT/INFO/Auto ; # options: None/NSE/ReINVITE ; # options: None/AVT/INFO ; ; al_Plan[2] "(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)" ; "No" ; rd" ; # options: Forward/Reverse rd" ; # options: Forward/Reverse rd" ; # options: Forward/Reverse[...]
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# *** 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" ; [...]
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CWT8_Cadence "2.3(.3/2 )" ; -r1" ; -r2" ; -r3" ; -r4" ; -r5" ; -r6" ; -r7" ; Ring8_Name "Bellcore-r8" ; # *** Ring and Call Waiting Tone Spec Ring_Waveform "Sinusoid" ; # options: Sinusoid/Trapezoid Ring_Frequency "25" ; Ring_Voltage "70" ; CWT_Frequency "440@[...]
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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_Co[...]
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CIDCW Call Waiting Caller ID CNG Comfort Noise Generation CPC Calling Party Control CPE Customer Premises Equipment CWCID Call Waiting Calle r ID CWT Call Waiting Tone D/A Digital to Analog Converter dB decibel dBm dB with respect to 1 milliwatt DHCP Dynamic Host C onfiguration Protocol DNS Domain Name Server DRAM Dynamic Random Access Memory DSL D[...]
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RTT Rou nd Trip Time SAS Streaming Audio Server SDP Session Description Protocol SDRAM Synchronous DRAM sec second s SIP Session Initiation Protocol SLIC Subscriber Line Interface Circuit SP Service Provider PAP2 Phone Adaptor Ports 2 (Li nksys Phone Adaptor) SSL Secure Socket Layer TFTP Trivial File Transfer Protocol TCP Transmission Control Proto[...]
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Circuits: The communication path(s) that carr y calls between two points on a network. Customer Premise Equipment: The o nly part of th e telecommunications system that the customer comes into direct contact with. Example of such pieces of equipment are: tel ephones, key systems, PBXs, voicemail systems and call accounting system s as well as wirin[...]
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© 2004 Linksys Proprietary (See C opyright Notice on Page 2) 117[...]