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PHONE ADAPTER Administration Guide August 2004 Linksys PAP2 and RT31P2 © 2004 Linksys Proprietary (See C opyright Notice on Page 2) 1.
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.
Table of Contents 1. Introduction .................................................................................................................................... 6 1.1. The Session Initiation Protocol .............................................
4.9. Progress Tone and Ring Configuration .............................................................................. 67 4.9.1. Distinctive Ring and Other Ring Settings ...............................................................................
7.2.1.3. SIP Support in Network Address Translatio n Networks – NAT ............................................96 7.2.2. Codec Name Assignment .........................................................................................................
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.
• 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.
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.
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.
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.
• 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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
# 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” ; .
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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 ].
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.
• 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).
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.
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.
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-str.
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.
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.
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.
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 Profi.
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 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.
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.
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.
- 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.
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.
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).
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.
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.
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 us.
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.
- 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.
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”.
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.
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 C.
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.
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).
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).
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.
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.
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,( .
( 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.
( 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.
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.
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.
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.
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.
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.
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 regis.
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.
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.
• 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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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).
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.
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.
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.
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).
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.
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 Indi.
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.
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 Inter.
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.
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 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.
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.
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.
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.
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.
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 "" ; ; ; ; ; " ; ; ; .
GPP_G "" ; GPP_H "" ; GPP_I "" ; GPP_J "" ; GPP_K "" ; GPP_L "" ; GPP_M "" ; ; P_O "" ; *** SIP Parameters x_Forward &qu.
G729b_Dynamic_Payload "99" ; E_Codec_Name "NSE" ; ne-event" ; 26r24_Codec_Name "G726-24" ; "G726-32" ; 26r40_Codec_Name "G726-40" ; rameters UN_T.
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] ".
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.
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" ; ! ".
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_Wit.
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.
# *** Distinctive Ring Settings Ring1_Caller[2] ! "" ; Ring2_Caller[2] ! "" ; Ring3_Caller[2] ! "" ; Ring4_Caller[2] ! "" ; Ring5_Caller[2] ! "" ; Rin.
CWT8_Cadence "2.3(.3/2 )" ; -r1" ; -r2" ; -r3" ; -r4" ; -r5" ; -r6" ; -r7" ; Ring8_Name "Bellcore-r8" ; # *** Ring and Call Waiting Tone Spec Rin.
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" ; S.
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 .
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 P.
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.
© 2004 Linksys Proprietary (See C opyright Notice on Page 2) 117.
Un point important après l'achat de l'appareil (ou même avant l'achat) est de lire le manuel d'utilisation. Nous devons le faire pour quelques raisons simples:
Si vous n'avez pas encore acheté Cisco Systems Linksys PAP2 c'est un bon moment pour vous familiariser avec les données de base sur le produit. Consulter d'abord les pages initiales du manuel d'utilisation, que vous trouverez ci-dessus. Vous devriez y trouver les données techniques les plus importants du Cisco Systems Linksys PAP2 - de cette manière, vous pouvez vérifier si l'équipement répond à vos besoins. Explorant les pages suivantes du manuel d'utilisation Cisco Systems Linksys PAP2, vous apprendrez toutes les caractéristiques du produit et des informations sur son fonctionnement. Les informations sur le Cisco Systems Linksys PAP2 va certainement vous aider à prendre une décision concernant l'achat.
Dans une situation où vous avez déjà le Cisco Systems Linksys PAP2, mais vous avez pas encore lu le manuel d'utilisation, vous devez le faire pour les raisons décrites ci-dessus,. Vous saurez alors si vous avez correctement utilisé les fonctions disponibles, et si vous avez commis des erreurs qui peuvent réduire la durée de vie du Cisco Systems Linksys PAP2.
Cependant, l'un des rôles les plus importants pour l'utilisateur joués par les manuels d'utilisateur est d'aider à résoudre les problèmes concernant le Cisco Systems Linksys PAP2. Presque toujours, vous y trouverez Troubleshooting, soit les pannes et les défaillances les plus fréquentes de l'apparei Cisco Systems Linksys PAP2 ainsi que les instructions sur la façon de les résoudre. Même si vous ne parvenez pas à résoudre le problème, le manuel d‘utilisation va vous montrer le chemin d'une nouvelle procédure – le contact avec le centre de service à la clientèle ou le service le plus proche.
