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LINKSYS SPA941 Manual

1. Parameter Name Description Type Default Prefer G711u Code Dialing code will make this codec the preferred ActCode 017110 codec for the associated call Force G711u Code Dialing code will make this codec the only codec that ActCode 027110 can be used for the associated call Prefer G711a Code Dialing code will make this codec the preferred ActCode 017111 codec for the associated call Force G711a Code Dialing code will make this codec the only codec that ActCode 027111 can be used for the associated call 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 codec that ActCode 02723 can be used for the associated call Prefer G726r16 Code Dialing code will make this codec the preferred ActCode 0172616 codec for the associated call Force G726r16 Code Dialing code will make this codec the only codec that ActCode 0272616 can be used for the associated call Prefer G726r24 Code Dialing code will make this codec the preferred ActCode 0172624 codec for the associated call Force G726r24 Code Dialing code will make this codec the only codec that ActCode 0272624 can be used for the associated call Prefer G726r32 Code Dialing code will make this codec the preferred ActCode 0172632 codec for the associated call Force G726r32 Code Dialing code
2. Parameter Name Description Type Default SIP TOS DiffServ Value TOS DiffServ field value in UDP IP Packets carrying a Byte Oxb8 SIP Message RTP TOS DiffServ TOS DiffServ field value in UDP IP Packets carrying a Byte Oxb8 Value RTP data Network Jitter Level 4 settings are available very high high medium low Choice High This parameter affects how jitter buffer size is adjusted in the SPA Jitter buffer size is adjusted dynamically The minimum jitter buffer size is 30 ms or 10 ms current RTP frame size which ever is larger for all jitter level settings But the starting jitter buffer size value is larger for higher jitter levels This parameter controls the rate at which to adjust the jitter buffer size to reach the minimum If the jitter level is set to high then the rate of buffer size decrement is slower more conservative else faster more aggressive Call Feature Settings Parameter Name Description Type Default Blind Attn Xfer Enable MOH Server The User ID or URL of the auto answering SAS to Str127 contact for MOH services Examples 5000 1001 music linksys com 66 12 123 15 5061 Note When only a user id is given the current proxy or outbound proxy will be contacted as in the making of a 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 70 LINKSYS A Division of Cisco Systems Inc regular outbo
3. e Provisioning Overview Provisioning This section contains information regarding the steps that a network administration should take when setting up a provisioning system for large numbers of IP telephones and or terminal adaptors An additional reference document is the Linksys SPA Provisioning Guide which provides detailed information on provisioning requirements Provisioning Capabilities The SPA941 provides for secure provisioning and remote upgrades The following provides an overview of the basic functionality and requirements for provisioning IP telephones 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 81 LINKSYS A Division of Cisco Systems Inc e Provisioning is achieved through configuration profiles requested by the device from a provisioning server via TFTP HTTP or HTTPS e No end user intervention is required to initiate or complete a profile update or firmware upgrade The SPA941 is programmed to resync with the server on power up and periodically thereafter e Remote upgrade is also achieved via TFTP HTTP or HTTPS e General purpose parameters are provided as an additional aid to service providers in managing the provisioning process These parameters are often used to hold encryption keys file path locations or provisioning state e The SPA can be configured to resync its internal configuration state to a remote profile periodically and on power up
4. A Division of Cisco Systems Inc Only one audio device can be selected at anytime The speaker is selected by pressing the SPEAKER button The headset is selected by pressing the HEADSET button When the SPEAKER or HEADSET is selected the corresponding LED will be steady GREEN The handset is selected whenever it goes from the on hook to the off hook position Hence selecting or turning on of any of the audio device is equivalent to an off hook action while turning off of any of the audio devices while it is equivalent to an on hook hang up operation When there are no active calls all audio devices are deselected When any one of the calls becomes active the handset will be selected automatically as the audio device if it is off hook else the speaker or the headset will be selected according to configuration the user can configure whether speaker or the headset device has higher preference A line is selected by pressing the corresponding Line Key To make or receive calls the user must select a call appearance Line Key or an audio device If the user selects a call appearance then an audio device is selected automatically according to user s preference setting If the user selects an audio device an idle call appearance is selected automatically in the order L1 L2 L3 and L4 The user may switch between audio devices while the call in any of the active states Exception When a new call appearance is selected automatically wh
5. A Division of Cisco Systems Inc foregoing this warranty does not cover any defect resulting from i any design or specification supplied by an entity other than Linksys ii non observance of technical operating parameters e g exceeding limiting values or iii misuse abuse abnormal conditions or alteration by anyone other than Linksys Replacement Repair Refund After the receipt of an RMA Return Materials Authorization request Linksys will attempt to refund repair or replace this device To receive an RMA number for this device contact the party from whom it 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 9 LINKSYS A Division of Cisco Systems Inc Network and Service Configuration Settings This unit may have been supplied by or sponsored by a telephone service provider If so the service provider or network administrator who supplied the unit may have provided a pre configuration of the network and service settings Depending on the configuration policy end user access to the local configuration settings may be restricted or inaccessible Therefore some of the network and service configuration setting instructions described in the following sections may not be available on all units e SPA941 Overview The SPA941 is a SIP based IP telephone offered by Linksys a division of Cisco Systems The SPA941 supports up to four phones lines of operation It has a pi
6. If enabled with lt STUN Enable gt yes and a valid lt STUN Servers the SPA will perform a NAT type discovery operation when first power on by contacting the configured STUN server The result of the discovery will be reported in a Warning header in all subsequent REGISTER requests Bool No 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 57 LINKSYS A Division of Cisco Systems Inc Warning 399 spa lt stun type gt where lt stun type gt is one of the following Unknown NAT Type STUN Server Not Reachable STUN Server Not Responding Open Internet Detected Symmetric Firewall Detected Full Cone NAT Detected Restricted Cone NAT Detected Symmetric NAT Detected If the SPA detects Symmetric Nat or Symmetric Firewall Nat Mapping will be disabled that is no substitution of IP address and port with external IP address an nat mapped port Ext IP If this parameter is specified the SPA will assume this IP IP address when generating SIP messages and SDP if NAT Mapping is enabled for that Line However results of STUN and VIA received parameter processing if available will supersede this statically configured value Ext RTP Port Min External port mapping of lt RTP Port Min gt If the Port parameter is specified and non zero the SPA will use this value to compute the corresponding external RTP port
7. Templates are compared in the order given The first not the closest match is selected The parameter name must match exactly If more than one definition for a parameter is given in a configuration file the last such definition in the file is the one that will take effect in the SPA Aparameter specification with an empty parameter value forces the parameter back to its default value To specify an empty string instead use the empty string as the parameter value Data Types Unsn Unsigned n bit value where n 8 16 or 32 It can be specified in decimal or hex format such as 12 or 0x18 as long as the value can fit into n bits Sign Signed n bit value It can be specified in decimal or hex format Negative values must be preceded by a sign A sign before positive value is optional Int A generic integer value The range depends on the parameter Strn A generic string with up to n non reserved characters Floatn A floating point value with up to n decimal places Timen Time duration in seconds with up to n decimal places Extra decimal places specified are ignored PwrLevel Power level expressed in dBm with 1 decimal place such as 13 5 or 1 5 dBm Bool Boolean value of either yes or no 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 45 LINKSYS A Division of Cisco Systems Inc a b c A choice among a
8. Parameter Name Description Type Default SIP T1 RFC 3261 T1 value RTT Estimate Range 0 64 Time3 5 sec SIP T2 RFC 3261 T2 value Maximum retransmit interval Time3 4 for non INVITE requests and INVITE responses Range 0 64 sec SIP T4 RFC 3261 T4 value Maximum duration a message Time3 5 will remain in the network Range 0 64 sec SIP Timer B INVITE time out value Range 0 64 sec Time3 32 SIP Timer F Non INVITE time out value Range 0 64 sec Time3 32 SIP Timer H INVITE final response time out value Range 0 Time3 32 64 sec SIP Timer D ACK hang around time Range 0 64 sec Time3 32 SIP Timer J Non INVITE response hang around time Range 0 Time3 32 64 sec INVITE Expires INVITE request Expires header value in sec 0 Time 180 do not include Expires header in INVITE Range 0 2 il RelNVITE Expires RelNVITE request Expires header value in sec 0 Time 30 do not include Expires header in the request 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 54 LINKSYS A Division of Cisco Systems Inc Range 0 271 1 sender report on an active connection Reg Min Expires Minimum registration expiration time allowed from Time0 1 the proxy in the Expires header or as a Contact header parameter If proxy returns
9. Prefer G729a 01729 Set preferred codec to G729a for next outbound call Force G711u 027110 Force to use G71 1u for next outbound call Force G711a 027111 Force to use G711a for next outbound call Force G723 02723 Force to use G723 for next outbound call Proprietary See Copyright Notice on Page 2 34 LINKSYS A Division of Cisco Systems Inc e Force G726r16 e Force G726r24 0272616 Force to use G726r16 for next outbound call 0272624 Force to use G726r24 for next outbound call e Force G726r32 0272632 Force to use G726r32 for next outbound call e Force G726r40 0272640 Force to use G726r40 for next outbound call e Force G729a 02729 Force to use G729a for next outbound call pe ee opge se When off hook dialing the star code is processed by the phone as soon as they are entered and recognized by the phone When on hook dialing the user can pre dial multiple star code before the target number such as 67 017110 the star codes will be handled one by one in the order they are entered before the target number is dialed out There is SK code in the dialing soft key to assist the user on what code is available Star code can be included in a directory entry and redial list In addition there are two special codes e Referral Services Codes This is a list of separated star codes such as 79 199 09 where entering each member triggers a blind
10. lt dialed subsequence transmitted subsequence gt So for example lt 8 1650 gt xxxxxxx would match 85551212 and transmit 16505551212 Intersequence Tones An outside line dial tone can be generated within a sequence by appending a character between digits Thus the sequence 9 1xxxxxxxxxx sounds an outside line dial tone after the user presses 9 until the 1 is pressed Number Barring A sequence can be barred rejected by placing a character at the end of the sequence Thus 1 900xxxxxxx automatically rejects all 900 area code numbers from being dialed 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 would set the interdigit long timeout to 8 seconds for the line associated with this dial plan And L 8 S 4 would override both the long and the short timeout values Local Timer Overrides The long and short timeout values can be changed for a particular sequence starting at a particular point in the sequence The syntax for long timer override is L delay value Note the terminating space character The specified delay value is measured in seconds Similar
11. 1 2 Reorder Tone Played when an outbound call has failed or ToneScript 480 19 620 after the far end hangs up during an 19 10 25 25 14 2 established call Off Hook Warning Played when the subscriber does not place the ToneScript 480 Tone handset on the cradle properly 10 620 0 10 125 1 25 1 2 Ring Back Tone Played for an outbound call when the far end is ToneScript 440 19 480 ringing 19 2 4 1 2 Confirm Tone This should be a brief tone to notify the user ToneScript 600 that the last input value has been accepted 16 1 25 25 1 SIT1 Tone An alternative to lt Reorder Tone gt played when ToneScript 985 16 1428 an error occurs while making an outbound call 16 1777 The RSC to trigger this tone is configurable 16 20 380 0 1 380 0 see Section 2 380 0 3 0 4 0 SIT2 Tone See lt SIT1 Tone gt ToneScript 914 16 1371 16 1777 16 20 274 0 1 274 0 2 380 0 3 0 4 0 SIT3 Tone See lt SIT1 Tone gt ToneScript 914 16 1371 16 1777 16 20 380 0 1 380 0 2 380 0 3 0 4 0 SIT4 Tone See lt SIT 1 Tone gt ToneScript 985 16 1371 16 1777 16 20 380 0 1 274 0 2 380 0 3 0 4 0 MWI Dial Tone This tone is played instead of lt Dial Tone gt ToneScript 350 19 440 when there are unheard messages in the 19 2 1 1 1 2 10 0 subscriber s mail box 1 2 Cfwd Dial Tone Special dial tone played when call forward all is ToneScript 350 19 440 2003 2005 Links
12. 4 Cadence 8 Cadence 8 script CadScript 60 4 2 3 2 8 4 Control Timer Values sec Parameter Name Description Type Default Reorder Delay Delay after far end hangs up before reorder tone is Time 5 played 0 plays immediately inf never plays Range 0 255 sec Call Back Expires Expiration time in sec of a call back activation Timed 1800 Ragne 0 65535 sec Call Back Retry Intvl Call back retry interval in sec Range 0 255 sec Timed 30 Call Back Delay Delay after receiving the first SIP 18x response Time3 0 5 before declaring the remote end is ringing If a busy response is received during this time the SPA still considers the call as failed and keeps on retrying VMWI Refresh Intvl Interval between VMWI refresh to the CPE Time3 0 5 Interdigit Long Timer Long timeout between entering digits when dialing Time0 10 Range 0 64 sec 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 60 LINKSYS A Division of Cisco Systems Inc Interdigit Short Timer Short timeout between entering digits when dialing Range 0 64 sec Timed 3 Notes 1 The interdigit timer values are used as defaults when dialing The Interdigit_Long_Timer is used after any one digit if all valid matching sequences in the dial plan are incomplete as dialed The Interdigit_Shor
13. Additional triggers are available including SIP NOTIFY messages registration state and provisioning state e 256 bit symmetric key encryption of profiles e Supports targeted end point unique key less profile encryption e The SPA941 supports a secure first time provisioning mechanism using SSL functionality For this purpose each SPA941 is loaded with a unique client certificate at manufacturing time The certificate identifies each device by MAC address Serial Number and Product Name This feature can be used in conjunction with a properly configured HTTPS provisioning server to provide for an additional level of security in provisioning units e Profile resyncs and firmware upgrades are attempted only when the SPA is idle as they may trigger a software reboot as a result of changes in parameter values A configurable override delay is available to force a resync after a predetermined grace period Configuration Profile The SPA941 configuration profile can take one of two forms a compact binary file or an XML text file By convention the profile is named with the extension cho e g spa941 cfg In the case of a binary profile the Linksys Profile Compiler tool SPC is provided by Linksys for compiling a plain text file containing parameter value pairs into a properly formatted and optionally encrypted CFG file The spc tool is available from Linksys for the Win32 environment spc exe and the Linux i386 elf environment spc
14. Calls with Calling Name and Number URI IP Dialing Support Vanity Numbers Configurable Dial Plans with Auto Completion of dialed number Do Not Disturb callers hear line busy tone On Hook Default Audio Configuration Speakerphone and Headset Multiple Ring Tones with Selectable Ring Tone per Extension Date and Time with Intelligent Daylight Savings Support Call Timer Call Duration and Start Time Stored in Call Logs Name and Identity Text Displayed at Start Up Distinctive Ringing Based on Calling and Called Number Ten User Downloadable Ring Tones Ring Tone Generator Free from www linksys com Speed Dialing Configurable Dial Numbering Plan Support per Line DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy Syslog Debug Report Generation and Event Logging SecureCall Encrypted Voice Communication Support Built in Web Server for Administration and Configuration with Multiple Security Levels Automated Provisioning Multiple Methods Up to 256 Bit Encryption HTTP HTTPS TFTP e Optionally Require Admin Password to Reset Unit to factory Defaults Feature requires support by call server Navigating the SPA941 Graphical User Interface Note Although this guide provides an overview of how to navigate the SPA941 Graphical User Interface GUI its primary focus is to describe the administrative functions available via the phone s web interface For more detailed discussion of the phone s user interface
15. Copyright Notice on Page 2 19 LINKSYS A Division of Cisco Systems Inc When the call back number rings or answers the call appearance will ring also like a normal incoming call the ring to be used is the default ring for the corresponding extension for that Line key If the call back party answers the call before the local user does the SPA941 sends holding tone to the call back party If the local user picks up the call back Line key first he will hear ring back tone like a regular outbound call Message Waiting Indication MWI The SPA941 indicates message waiting in 2 ways 1 Turning on the red MWI light LED with an envelope icon right above LED on the phone 2 Playing stutter tone instead of regular dial tone The MWI light only applies to new message indication received on Extension 1 It will not be turned on or off due to new message indication received on the other Extensions 2 4 On the other hand the SPA941 can be configured to play a stutter tone on the individual call appearance according to the message waiting state of the corresponding Extension X Accessing Voice Mail The SPA941 has a VM retrieval key for quick access to voice mail If a voice mail access number or URL is not specified pressing the key brings up a screen where the user can enter the appropriate information This information can also be entered from the web page or downloaded via provisioning The SPA941 will attempt to call the
16. Name Description Type Default Ring 1 Ring tone script for Ring 1 RingScript n Classic 1 w 3 c 1 Ring 2 Ring tone script for Ring 2 RingScript n Classic 2 w 3 c 2 Ring 3 Ring tone script for Ring 3 RingScript n Classic 3 w 3 c 3 Ring 4 Ring tone script for Ring 4 RingScript n Classic 4 w 3 c 4 Ring 5 Ring tone script for Ring 5 RingScript n Simple 1 w 2 c 1 Ring 6 Ring tone script for Ring 6 RingScript n Simple 2 w 2 c 2 Ring 7 Ring tone script for Ring 7 RingScript n Simple 3 w 2 c 3 Ring 8 Ring tone script for Ring 8 RingScript n Simple 4 w 2 c 4 Ring 9 Ring tone script for Ring 9 RingScript n Simple 5 w 2 c 5 Ring 10 Ring tone script for Ring 10 RingScript n THx w 4 c 7 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 68 LINKSYS A Division of Cisco Systems Inc Extension 1 4 Parameters Each Extension has its own set of Ext configuration parameters In a configuration profile the parameters listed in this section must be appended with n where n 1 2 3 4 to indicate which extension the parameter belongs to General Parameter Name Description Type Default Line Enable Enable this extension for service Bool Yes Share Ext Indicates whether this extension is to be shared with Bool Yes other stations If the extension is not shared then a call appearance assigned to this extension is not shared regardless the setting of lt Shar
17. Register Enable periodic registration with the lt Proxy gt This Bool Yes parameter is ignored if lt Proxy gt is not specified Make Call Without Reg Allow making outbound calls without successful Bool No dynamic registration by the unit If No dial tone will not play unless registration is successful Ans Call Without Reg Allow answering inbound calls without successful Bool No dynamic registration by the unit Register Expires Expires value in sec in a REGISTER request SPA will Time 3600 periodically renew registration shortly before the current registration expired This parameter is ignored if lt Register gt is no Range 0 2 1 sec Use DNS SRV Whether to use DNS SRV lookup for Proxy and Bool No Outbound Proxy 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 71 LINKSYS A Division of Cisco Systems Inc will retry from the highest priority proxy or outbound proxy servers after it has failed over to a lower priority server This parameter is useful only if the primary and backup proxy server list is provided to the SPA via DNS SRV record lookup on the server name Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the SPA will not attempt to fall back after a fail over DNS SRV Auto Prefix If enabled the
18. Return 69 call the last caller regardless which extension e Call Back 66 Periodically redial the last busy number every 30s by default until it rings or until the trial expires 30 min by default regardless which extension Only 1 call back operation can be ordered at a time A new order will automatically cancel the last order e Cancel Call Back 86 Cancel the last call back operation e Call Forward All 72 Call forward all inbound calls Applicable to primary extension only 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 33 LINKSYS A Division of Cisco Systems Inc 2003 2005 Linksys a Division of Cisco Systems Cancel Call Forward All 73 Cancel call forward all Applicable to primary extension only Call Forward Busy 90 Call forward on busy Applicable to primary extension only Cancel Call Forward Busy 91 Cancel call forward on busy Applicable to primary extension only Call Forward No Answer 92 Call forward if no answer Applicable to primary extension only Cancel Call Forward No Answer 93 Cancel call forward no answer Applicable to primary extension only Block Caller ID Always 67 Block caller ID on all outbound calls Apply to all extensions Block Caller ID Per Call 81 Block caller ID on the next outbound call on the current call appearance only Unblock Caller ID Always 68 Unblock c
19. SPA will automatically prepend the Proxy Bool No or Outbound Proxy name with sip _udp when performing a DNS SRV lookup on that name Proxy Fallback Intvl This parameter sets the delay sec after which the SPA TimeO 3600 Subscriber Information used for SIP authentication Else the pair lt User ID gt and lt Password gt are used Parameter Name Description Type Default Display Name Subscriber s display name to appear in caller id Str23 User ID Subscriber s user id Usually a E 164 number Str47 Password Subscriber s a c password Str23 Auth ID Subscriber s authentication ID Str39 Use Auth ID If set to yes the pair lt Auth ID gt and lt Password gt are Bool No Mini Certificate Base64 encoded of Mini Certificate concatenated with the 1024 bit public key of the CA signing the MC of all subscribers in the group Str508 Empty SRTP Private Key Base64 encoded of the 512 bit private key per subscriber for establishment of a secure call Str88 Empty Notes 1 If proxy responded to REGISTER with a smaller Expires value the SPA will renew registration based on this smaller value instead of the configured value If registration failed with an Expires too brief error response the SPA will retry with the value given in the Min Expires header in the error response 2 MOH Notes e The remote party must indicate that it can receive audio while hold
20. b c IP IP Address in the form of x x x x where x between 0 and 255 For example 10 1 2 100 Port TCP UDP Port number 0 65535 It can be specified in decimal of hex format UserID User ID as appeared in a URL up to 63 characters FQDN Fully Qualified Domain Name such as sip linksys com 5060 or 109 12 14 12 12345 It can contain up to 63 characters Phone A phone number string such as 14081234567 69 72 345678 or a generic URL such as 1234 10 10 10 100 5068 or jsmith linksys com It can contain up to 39 characters ActCode Activation code for a supplementary service such as 69 It can contain up to 11 characters PhTmplt A phone number template Each template may contain 1 or more patterns separated by a White space at the beginning of each pattern is ignored and represent wildcard characters It can contain up to 39 characters Examples 1408 1510 14081237 55571 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 Syntax S S2 where S D on off 1 0n 2 off 2 on off Jon oft Jon soft done eioft ell and is known as a section oni and oft are the on off duration in seconds of a segment andi 1 or 2 andj 1 to 6 D is the total duration of
21. eegen E Ee eebe EE tetduevtucvensecthcdeezedeutawceeste 82 19 PROVISIONING FLOW eut a ants ated eed evidential ete eae a 84 Pre PrOVISIONING ME 84 Provisioning Propao dees SEANCE 84 14 FIRMWARE UPGRADE p manana a a eaaa a aa a a a aaa 85 PreOMIUM el TE 86 15 FUNCTIONAL URLS FOR UPGRADES REBOOT AND RESYNCH sssssssssssssnesisnssrnrnssrnrrnsrnernnne 86 Upgrade URL isisisi isea E aaa aAa TEE a a 86 Resyni TE 86 Reboot URL ege aiaa TEESE EEA R a 87 16 PERFORMANCE DREPORTINGFEATURES 87 Call Statistics and Reporting cccccescceseeeeeseeeeeseeeeeneseseeeeeseeeeeeaeeesaaesasneeeneeeeseaesesaesasneeeeseesesnaeseseeeeneeees 87 Report Generation and Event Logging cseecceseeeeseeeeseeeeneeeeeeeeeeseaeeenseeeeeeeeesaesaseeeeneeeeseaeseseaeeaseaeeees 88 Error and Log Reporting ceceecccceeeeeeeeeeeeeeeneeeee se eeeeeeeeeeseeeseseeeeeseseeeeeseseeeeeseeseeeeeseseneeeseseenenseseeneneeseenenes 89 Syslog and Debug Server RecordS cccseccesceseeeeeeeeeeeeeeesnaeeeseeeeseaesaseaeeasneeseeeeeseaesaseeeenseeeseaeseseneeeneeees 89 17 TROUBLESHOOTING E 89 Phone does not turn On Or DOOU Up ieee eirean ea eee ERARE EA EE E ES anaa anra ae AAEREN EEEE 89 Phone will not make or receive Calls 2 2 cccceeecccceceeeceeeeeeeceeeeeeeceeeeeeeaeeeeseeeeeeeeeaeesneeceeeseeeaeeeseeeageneenees 90 Galls with Poor Voice QUA EE ebe a aaa a a ghia a ea A a aaa aaa cess LEE e aea iaia 91 18 CUSTOMER SUPPORT METHODS riitit iea anded i
22. encoded as a fixed 12B string 000000010134 The tool generates the lt Mini Certificate gt and lt SRTP Private Key gt parameters that can be provisioned to the SPA For Example gen_mc ca_key Joe Smith 14085551234 00 00 00 1 1 34 Produces lt Mini Certificate gt Sm9IllIFNtaxX ROAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAXNDAANT U1 MTIZNAAAAAAAMDAWMDAW MDEwMTMO00OvJakde2vVMF3Rw4pPXL7IlAglagMpbLSAG2 YISqt1 98Cp9rP xMGFfoPmDKGx6JFtk Q5sxLcuwgxpxpxkeXvpZkIYlpsb28L4Rhg5qZA Gqj1 hDFCmG dffZ9SJhxES767 GOJIS N8IQBLrOAuem 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 24 LINKSYS A Division of Cisco Systems Inc otknSjjjOy8c 1ITCd2t44MhOvmwNg4fDck2YdmTMBR851 6xJt4 uQ LJQIni2kwqlm7scDvIl5k232EvwwVtCKO AYa4eWd6fQOpiESCO9CCYIaYU1 X5 JUU EBZmi3AmcqE9U1LxEOGwopaGyGOh3VyhkKgi6JaVtQZt87P iJINKW8XQj38B9Qqe3VgYxWCQNa335Y CnDsenASeBxuMIEaBC Yd 111 fVEodJZOGwXwfAdeOMhcbD0Okj 7LVIzesTyk2TZYTccnZ75TuTjj13qvYs lt SRTP Private Key gt b DWc96X4Y QraCnYzl5en1 ClUhVQQaqrvcr6Qd 8R52IEvJjOw e Kim4xiiFEPaKmU8UbooxKG36SEdKus p0OAQ e Memory Features and other User Accessible Settings Call Logs There are three call logs maintained for each VoIP interface e Redial List Each Redial List entry is added when dialing is completed regardless the number is correct or not or the call is successful or not e Answered Calls An Answered Call is logged when the coming call is answered e Missed C
23. entity in the public network and stores the mapping discovery results returned by the server Communicate the NAT mapping information to the external SIP entities If the entity is a SIP Registrar the information should be carried in the Contact header that overwrites the private address port information If the entity is another SIP UA when establishing a call the information should be carried in the Contact header as well as in the SDP embedded in SIP message bodies The VIA header in outbound SIP requests might also need to be substituted with the public address if the UAS relies on it to route back responses Extend the discovered NAT mappings by sending keep alive packets Since the mapping is only alive for short period the SPA continues to send periodic keep alive packets through the mapping to extend its validity as necessary Note NAT Mapping does not have to applied globally on the phone It can be enabled or disabled per Extension Some service providers offer their own solution of NAT traversal NAT mapping should be disabled on the SPA941 for extensions configured for these service providers e Data Networking Features Supported The SPA941 supports the following Data Networking Features 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 41 LINKSYS A Division of Cisco Systems Inc MAC Address IEEE 802 3 IPv4 Internet Protocol Version 4 RFC 791 upgradeable to v6 RF
24. is specified TFTP protocol is assumed Note Only TFTP is supported in the current release e f no server name is specified the host that requests the URL is used as server name e f no port specified default port of the protocol is used 69 for TFTP e The profile path is the path to the new profile to resync with For example http 192 168 2 217 upgrade tftp 192 168 2 251 spaconft scf Reboot URL Through the Reboot URL the user can reboot the SPA Note Upon request the SPA will reboot only when it is idle The Reboot URL is http lt spa ip addr gt admin reboot e Performance Reporting Features Call Statistics and Reporting The following lists the statistics collected by the SPA during normal operation These statistics are presented in the SPA web page under the Info tab Line status is reported for each line 1 and 2 Each line maintains up to 2 calls Call 1 and 2 System Status Current Time Current time and date E g 10 3 2003 16 43 00 Elapsed Time Total time elapsed since last reboot E g 25 days and 18 12 36 Broadcast Pkts Sent Total number of broadcast packets sent Broadcast Pkts Recv Total number of broadcast packets received Broadcast Bytes Sent Total number of broadcast bytes sent Broadcast Bytes Recv Total number of broadcast bytes received and processed Broadcast Packets Dropped Total number of broadcast packets received but not processed Broadcast B
25. item a number is displayed on the left side of each menu entry that indicates its position in the current menu list A menu item can be directly accessed by typing its number on the keypad even if the item isn t currently visible on the display For instance when the menu key is pressed to display the top level menus items one through four are displayed Item nine can be immediately entered by pressing the 9 key on the keypad without having to scroll down to that item Similarly the fifteenth item in a menu can be entered by pressing the 1 key followed by the 5 key with less than a two second pause between the key presses Any leading 0 entries are ignored Some of the entries in the menus such as those in the Status menu are only informational these entries cannot be changed from within the menus Typically only the cancel soft key is active when informational menu entries are displayed The changeable menu entries can be modified by pressing the select soft key while the entry is highlighted The soft keys change to display the list of choices to modify the selected phone feature such as add paste edit and delete The modified entry can be saved by pressing the save soft key or discarded by pressing the cancel soft key A menu item s display string may be wider than what can be displayed When the user highlights a wide item the SPA941 automatically scrolls the contents horizontally from right to left so that the user can view the
26. linux i386 static In the case of an XML text profile the XML file itself is fed to the SPA941 directly without prior compilation into a binary form The XML text profile function is an advanced feature described in the SPA Provisioning Guide The syntax of the source plain text file accepted by the Linksys Profile Compiler is a series of parameter value pairs with the value enclosed in double quotes Each parameter value pair is followed by a semicolon thus Parameter_Name parameter value For example 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 82 LINKSYS A Division of Cisco Systems Inc Proxy 1 ep service com Register_Expires 1 900 A sample source plain text profile can be generated from the SPC tool with the following command line invocation spc exe sample profile sample txt The sample txt can then be edited as appropriate for the administrator s network and compiled into a binary profile with this SPC command line invocation spc exe sample txt sample cfg If no quoted value is specified for a parameter or if a parameter specification is missing entirely from the source plain text file the value of the parameter will remain unchanged in the SPA941 when the SPA941 resyncs to the compiled binary profile The syntax also controls the parameter s user level access when using the built in web interface to the SPA An optional exclamation po
27. of a connection and use the result in the SDP for establishing the connection However results of STUN if available will supersede this statically configured value NAT Keep Alive Intvl This is the interval at which keep alive messages are sent Uns16 15 by the SPA to preserve a NAT mapping It controls the interval for sending keep alive messages for the SIP signaling ports and also for the RTP ports 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 58 LINKSYS A Division of Cisco Systems Inc Regional Parameters Call Progress Tones Parameter Name Description Type Default Dial Tone Played when prompting the user to enter a ToneScript 350 19 440 phone number 19 10 0 1 2 Second Dial Tone An alternative to lt Dial Tone gt when user tries ToneScript 420 19 520 to dial a Three way call 19 10 0 1 2 Outside Dial Tone An alternative to lt Dial Tone gt usually used to ToneScript 420 16 10 0 1 prompt the user to enter an external phone number versus an internal extension This is triggered by a character encountered in the dial plan Prompt Tone Played when prompting the user to enter a call ToneScript 520 19 620 forward phone number 19 10 7 0 1 2 Busy Tone Played when a 486 RSC is received for an ToneScript 480 19 620 outbound call 19 10 5 5
28. offset to apply to form the local Str9 time A sign can be added at the beginning to indicate the offset should be subtracted from the current time to form the local time otherwise the offset is added Daylight Saving Time Rule If a rule is specified the SPA will automatically Daylight Saving adjust for daylight saving time See Daylight Time Rule Saving Time Section on syntax of this rule DTMF Playback Level Local DTMF playback level in dBm up to 1 PwrLevel 26 decimal place DTMF Playback Length Local DTMF playback duration in seconds up to 3 Time3 NW decimal places Phone Parameters General Parameter Name Description Type Default Station Name A name to identify this station reserved for future use Str31 Preferred Audio Specifies whether to use the headset or the speaker when Choice Speaker Device pressing a Line key to make a new call or to answer an incoming call Choices Speaker Headset Voice Mail Number A phone number or URL to check voice mail Str79 Text Logo To set the text logo to display when the phone boots up Default Str32 is blank which will show Linksys Up to 2 lines of text can be setup each line should contain less than 32 characters A newline character n must be inserted between the 2 lines and escaped with 0a For example Super 0aTelecom Line Key n n 1 4 Parameters in this section must be appended with n in the configuration profile Parameter Name Description Type Defau
29. packet except the last one contains a SR Sender Report and a SDES Source Description The last RTCP packet contains an additional BYE packet Each SR except the last one contains exactly 1 RR Receiver Report the last SR carries no RR The SDES contains CNAME NAME and TOOL identifiers The CNAME is set to lt User ID gt lt Proxy gt NAME is set to lt Display Name gt or Anonymous if user blocks caller ID and TOOL is set to the Verdor Hardware platform software version such as Linksys SPA2000 1 0 31 b The NTP timestamp used in the SR is a snapshot of the SPA s local time not the time reported by an NTP server If the SPA receives a RR from the peer it will attempt to compute the round trip delay and show it as the lt Call Round Trip Delay gt value ms in the Info section of SPA web page Dynamic Payload Types Parameter Name Description Type Default AVT Dynamic Payload AVT DTMF tones dynamic payload type Uns8 101 INFOREQ Dynamic Payload If this value is not blank the SPA will include X Uns8 nt inforeq in the SDP to indicate support of INFO method for out of band DTMF transmission per Nortel MCS specification The recommended value to use is 111 G726r16 Dynamic Payload G726 16 dynamic payload type Uns8 98 G726r24 Dynamic Payload G726 24 dynamic payload type Uns8 97 G726r40 Dynamic Payload G726 40 dynamic payload type Uns8 96 G729b Dynam
30. phone ring tone menu the User 1 and 2 choices are replaced by the corresponding name of the ring tone or Not Installed if the user ring tone slots have not been downloaded into The user ring tone 1 or 2 can be downloaded using the phone web interface with the link http lt phone ip addr gt ringtone1 lt url gt where lt url gt syntax is tftp host port lt path gt Only tftp is supported If host is not specified the TFTP host is the web client default port is 69 To remove user ringtone 1 from the phone set the lt path gt to delete such as http lt phone ip addr gt ringtone1 delete The link is case sensitive Notes e For User Ring Tone 1 and 2 the cadence is fixed with the on time equals to the duration of the ring tone file and off time equals to 4s The total ring duration is fixed at 60s e The user ring tone names displayed on the phone GUI are extracted from the ring tone file header file e No need to reboot phone after a ring tone download e However if your phone is already in ring tone GUI menu while the new ring tone is downloaded you will need to exit and re enter the ring tone menu in order to see the new ring tone name on the menu Star Code to Activate Deactivate Certain Services THE SPA941 accepts star codes to activate or deactivate certain services on the phone Below is a list of services that are accessible with a star code with the default star code shown in parenthesis e Call
31. please refer to the SPA941 User Guide The user can invoke the phone s GUI by pressing the menu button and can navigate through the menus using the round directional rocker knob and the soft keys The menus are hierarchical The select soft key appears when a menu entry has a sub menu The sub menu is entered by pressing the select soft key and exited by pressing the cancel soft key The menus can be exited and the phone returned to it default display by pressing the menu button The phone has four soft keys but at times more than four functions can be performed with the soft keys When this occurs a small triangle appears at either the bottom right or the bottom left corner of the display This indicates that more soft key functions can be accessed by pushing the four way rocker knob right or left respectively 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 13 LINKSYS A Division of Cisco Systems Inc Up to four menu items can be displayed at one time The entries in each menu can be cycled through in a circular fashion by pressing the up and down arrows on the directional rocker knob Pressing the down arrow when the last entry in a menu is selected causes the cursor to move back to the top entry in the menu list Press the up arrow when the first menu entry is selected causes the cursor to move to the last entry in menu While the user is navigating the menus down to a leaf level
32. port number for RTP transmission and Port 16384 reception RTP Port Max Maximum port number for RTP transmission and Port 16482 reception RTP Packet Size Packet size in sec Valid values must be multiple of Time3 0 03 0 01s Range 0 01 0 16 Max RTP ICMP Err Number of successive ICMP errors allowed when Uns32 0 transmitting RTP packets to the peer before the SPA will terminate the call If value is set to 0 the SPA will ignore the limit on ICMP errors RTCP Tx Interval Controls the interval sec to send out RTCP Time0 0 Proprietary See Copyright Notice on Page 2 Range 0 255 s 2003 2005 Linksys a Division of Cisco Systems 55 LINKSYS A Division of Cisco Systems Inc Notes 1 2 3 Reorder or Busy Tone will be played by default for all unsuccessful response status code lt RTP Port Min gt and lt RTP Port Max gt should define a range that contains at least 4 even number ports such as 100 106 If inbound SIP requests contain compact headers SPA will reuse the same compact headers when generating the response regardless the settings of the lt Use Compact Header gt parameter If inbound SIP requests contain normal headers SPA will substitute those headers with compact headers if defined by RFC 261 if lt Use Compact Header gt parameter is set to yes During an active connection the SPA can be programmed to send out compound RTCP packet on the connection Each compound RTP
33. r LedScript Local Ringing LED pattern during the Local Ringing state when the call appearance is ringing Leaving this entry blank indicates the default value of c r p f LedScript Local Active LED pattern during the Local Active state where the call appearance is engaged in an active call Leaving this entry blank indicates the default value of c r LedScript Local Held LED pattern during the Local Held state where the call appearance is held by this station Leaving this entry blank indicates the default value of c r p s LedScript Remote Undefined LED pattern during the Remote Undefined state where the shared call state is undefined the station is still waiting for the state information from the application server Not applicable if the call appearance is not shared Leaving this entry blank indicates the default value of c r p d LedScript Remote Seized LED pattern during the Remote Seized state where the shared call appearance is seized by another station Not application if the call appearance is not shared Leaving this entry blank indicates the default value of c r p d LedScript Remote Progressing LED pattern during the Remote Progressing state where another station is attempting on this shared call appearance an outbound call that is progressing Not applicable if the call appearance is not shared Leaving this entry blank indicates the d
34. the URL of the firmware BIN file into the Upgrade_Rule parameter The SPA941 will respond by downloading the file and performing the upgrade Subsequently a new upgrade will not be attempted again until the contents of Upgrade Rule are modified to a different URL such as by changing the firmware file name Example Upgrade_Rule http prov service com spa941 spa941 00 09 01 bin Using the above strategy the SPA941 might attempt to upgrade again if it is redirected to a different file for example if the DNS entries change and the server name resolves to a new IP address To prevent spurious upgrades it is better to condition the upgrade on a specific target firmware version number The SPA941 can be directed to upgrade to a specific version by indicating the version number before the URL The following examples illustrate This will upgrade to 1 0 3 Upgrade_Rule 1 0 3 http call me com firmware spa941 01 00 03 bin This will upgrade to 2 1 4 d Upgrade_Rule 2 1 4 d Tftp 192 168 20 100 phones spa941 02 01 04 d bin Note that the downloaded file at the specified URL must match the version specified in the conditional expression Please refer to the Linksys SPA Provisioning Guide for a more exhaustive description of the available provisioning features in the SPA941 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 85 LINKSYS A Division of Cisco Sy
35. the section in seconds All durations can have up to 3 decimal places to provide 1 ms resolution The wildcard character stands for infinite duration The segments within a section are played in order and repeated until the total duration is played Examples Example 1 Normal Ring 60 2 4 Number of Cadence Sections 1 Cadence Section 1 Section Length 60 s Number of Segments 1 Segment 1 On 2s Off 4s Total Ring Length 60s Example 2 Distinctive Ring short short short long 60 2 2 2 2 2 2 1 4 Number of Cadence Sections 1 Cadence Section 1 Section Length 60s Number of Segments 4 Segment 1 On 0 2s Off 0 2s Segment 2 On 0 2s Off 0 2s Segment 3 On 0 2s Off 0 2s Segment 4 On 1 0s Off 4 0s 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 46 LINKSYS A Division of Cisco Systems Inc Total Ring Length 60s 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 Z The section Zi is similar to the S section in a CadScript except that each on off segment is followed by a frequency components parameter Zi Dj on 4 off 1 f 1 0ni 2 off 2 f 2 oni 9 offi s fi 3 Lon oft Ah Lonis off s fis Lonie offie fielll where fi j ni n2 n n n gt Nne and 1 lt nk lt 6 indicates which of the frequenc
36. to send back a response to the originator of the message since its private source IP address port is not usable When a packet is sent from a device on the private network to some address on the external network the NAT selects a port at the external interface from which to send the packet to the destination address port The private address port of the device the external address port selected by the NAT to send the packet and the external destination address port of the packet form a NAT Mapping The mapping is created when the device first sends a packet from the particular source address port to the particular destination address port and is remembered by the NAT for a short period of time This period varies widely from vendor to vendor it could be a few seconds or a few minutes or more or less While the mapping is in effect packets sent from the same private source address port to the same public destination address port is reused by the NAT The expiration time of a mapping is extended whenever a packet is sent from the corresponding source to the corresponding destination More importantly packets sent from that public address port to the external address port of the NAT will be routed back to the private address port of the mapping session that is in effect Some NAT devices actually reuse the same mapping for the same private source address port to any external IP address port and or will route packets sent to its external address port
37. transfer operation except the transfer target number in this case will be prepended by the corresponding star code whereas blind transfer itself contains the target number only without the star code Like blind transfer a referral service code can only be entered when the last call is placed on hold e Feature Dial Services Codes this is a list of separated star codes where entering each member will cause the next outbound call target number prepended by the corresponding star code Disabling Services Services handled locally by the phone can be disabled by the administrator in one or two ways a by directly disabling the service in the configurable profile or b by emptying out the code associated with the service Services that can be disabled or enabled by method a are Block Caller ID including blocking and unblocking per call and always Block Anonymous Calls Do Not Disturb Secure Call including enabling and disabling per call and always Call Forward All Call Forward Busy Call Forward No Answer Call Back Conference Call no star code 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 35 LINKSYS A Division of Cisco Systems Inc Attended Call Transfer no star code Blind Call Transfer Services that can be disabled or enabled by method b are any of the services that have a star code assignment in the SPA941 configurati
38. unused line There must be a spare line available to perform the transfer Once the new dial tone is heard enter 98 the phone will prompt the user to enter the phone number of the transfer target Enter the number to complete the blind transfer If the user s phone has more than one call on hold on the last call placed on hold will be the one transferred Call Back This service mimics the call back service offered by the PSTN The user can activate this service on a busy number such that he will be called back as soon as the busy number becomes available The SPA941 and other members in the SPA family implements this feature by repeatedly dialing the busy number periodically until the called party rings or answers or until the service order is canceled by the user or expires For that reason it is better referred to as the repeat dialing service To activate this service the user selects a Line key and enters 66 Call back activation code The SPA941 then uses this call appearance to call the last called number The retry period and expiration time are configurable To cancel the service the user picks up any line and enter 86 When call back service is active the corresponding Line key will blink in green The user can still use this line to make or answer calls when that happens the call back service is temporarily paused until the call appearance is idle again 2003 2005 Linksys a Division of Cisco Systems Proprietary See
39. voice mail number or URL if there is an idle call appearance available Muting Calls The user can mute the audio input into the phone when an audio device is switched on by pressing the MUTE key To un mute press the MUTE key again If no audio device is on pressing the MUTE key has no effect When switching from the speaker phone to the handset the phone is automatically un muted Shared Call Appearances The SPA941 supports shared call appearances in association with a Broadsoft application server An extension can be shared by two 2 or more stations Any call appearances on a shared extension is a shared call appearance At any given time each station sharing a call appearance can monitor the state of the call appearance A station can select a share call appearance to make a call only if the call appearance is not being used by another station All stations will ring on an inbound call to the shared call appearance extension Whoever picks up the call first will take the call When a call is placed on hold by one station it can be resumed from another station sharing the same call appearance Additional methods for shared call appearances per vendor application server implementation are being added to the SPA941 and will be available in future firmware releases Please contact sales linksys com to inquire about roadmap development schedules 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Pag
40. will cold boot the phone and restart DHCP Restart will warm boot the phone without restarting DHCP Both options require the user to confirm before proceeding Factory Reset Factory reset will reset all parameters to default value Personal directory and call logs will be cleared also This option require the user to confirm before proceeding 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 31 LINKSYS A Division of Cisco Systems Inc Password Protection The user can configure a password to the phone to protect against access to the following options Directory Call History Redial List Missed Calls and Answered Calls Call Forward Speed Dial Ring Tone Preferences Time Date Voice Mail Factory Reset Set Password Audio Volume Adjustment There are 4 kinds of audio volume to be adjusted e Ringer Volume on the speaker e Speaker e Handset e Headset To adjust the volume press the VOL key to invoke the volume adjustment screen Then use the UP DOWN keys to increase or decrease the volume The kind of volume to adjust is by context When no audio device is selected at the moment the Ringer volume will be adjusted Each UP DOWN key press will play a short burst of the current ring tone If one of the audio devices is selected then the volume adjustment applies to that device The screen will show why kind of volume is being adjusted with a pictorial representatio
41. 2 5 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 48 LINKSYS A Division of Cisco Systems Inc total time specifies the total number of seconds to play the ring tone before it times out Example 1 n Classic 1 w 3 c 1 Example 2 n Simple 1 w 2 c 1 Example 3 TBD LedScript A script that describes the color and blinking pattern of a Line Key LED Each script contains a number of fields separated by a semicolon White spaces are ignored Each field has the syntax lt field name gt lt field value gt The allowed field name and corresponding field values are listed below c o r gla This field sets the color of the LED The 4 choices are o Off r red g green a amber p n b s b f b d b u d This field sets the blinking pattern of the LED The 4 choices are nb no blink steady on or off sb slow blink 1s on and 1s off fb fast blink 100ms on and 100ms off ud user defined according to the contents of the u field u on off on off This is a user defined blinking pattern used only when p ud It consists of up to 4 pairs of on off duration in seconds with up to 2 deciaml places each value is separated by a forward slash Example 1 c r p sb Color is red and slow blink Example 2 c o LED is off Example 3 c g Color is green and steady on Example 4 c a p ud u 1 1 1 1 1 9 Color is amber and blink with th
42. AND OTHER USER ACCESSIBLE GETTINGS 25 Call OG Secs a AE EE E A hie eae Ra tee Sa ea eh ete aac bue cu suces ue etuveucunatetas 25 Personal Directory EE 26 Entering and Saving Settings ccccccccesceesseeeeseeeeeeeeeseeeseseneeeeeeeeseaesesaaesnseeeeeeeeescaesaseeeeneeeeeeseseseenenseaetes 27 Breletemeg ege e EENS 28 Speed Dial sis ee ee EE 28 Caller and Called Name Matching c ccsseceeseeeeeseeesneeeseeeeneeeeesnaesesneeeneeeeseaesesaesaseeeeeeeeeseaesesaeeeneeees 29 Dialing ASSISTANCE TE 29 Time Daleen ies check aca shed NEARE dees sda E dened cit cote duce EATA 29 Daylight Savino NL E 30 Checking A e EN Reboot and Restaltt raseria airaa aaraa NaN Ea ira Era Eaa er Ea aiai Eri AE ia ANET AA iana 31 Factory Reset yerno E TE E T ENOT A OORS 31 Password Protection etieEeeEeege ee EES ENEE EENS 32 Audio Volume Adjustment cceceee cece sees ee en teense en neee eee neeee eee geeeee ea neeeeeeseeeeeegseeeesdseeeeeegseceeeesaseeeenensesenens 32 Ring tel 32 Star Code to Activate Deactivate Certain Services ssccsscceseeseeseseeeseeeeeenessneesenseenesseessnensneessnensaees 33 RL RT 35 6 VOICE AND SIGNALING FEATURES 0 ceccceccceeeceeeceeeeeeeeeeeeeeaeecaeeseeeeeaeesaeesaaeseaesaeenaeseaeeeieesnees 36 SIPiProxy Dvpammie Hed updates eege 36 Re registration with Primary SIP Proxy Server scccssccsseeceseeesseeseseeeeneeeeeeeeeeeseaeenseeeeeseeescaesaseeeeseeeneas 36 Codec Name ASSignimenitiaisiccciseczfece
43. C 188 ARP Address Resolution Protocol DNS A Record RFC 1706 SRV Record RFC 2782 DiffServ RFC 2475 and ToS Type of Service RFC 791 1349 DHCP Client Dynamic Host Configuration Protocol RFC 2131 ICMP Internet Control Message Protocol RFC792 TCP Transmission Control Protocol RFC793 UDP User Datagram Protocol RFC 768 RTP Real Time Protocol RFC 1889 RFC 1890 RTCP Real Time Control Protocol RFC 1889 SRTP Secure Real Time Control Protocol RFC XXXX e Configuring and Provisioning Overview The SPA941 requires the administrator to ensure that two files are loaded correctly onto the device These files are e A firmware file e g SPA941 bin This file provides the operating system and call processing functions for the device Firmware upgrades are explained in Section 13 e A configuration file e g SPA941 cfg This file provides the specific parameter values for the administrator s network Many of the remaining sections explain the details on the configuration file i e the configurable parameters their values how to edit the parameters how to load the configuration file securely and remotely Both files can be loaded to the device remotely using TFTP HTTP or HTTPS The configuration file can be manually accessed and changed using the phone s web server and that web interface is explained in the next section Note In this document Configuration refers to the parameter val
44. Converter ANC Anonymous Call B2BUA Back to Back User Agent Bool Boolean Values Specified as yes and no or 1 and 0 in the profile CA Certificate Authority CDR Call Detail Record CID Caller ID CIDCW Call Waiting Caller ID CNG Comfort Noise Generation CPC Calling Party Control CPE Customer Premises Equipment CWCID Call Waiting Caller ID CWT Call Waiting Tone D A Digital to Analog Converter dB decibel dBm dB with respect to 1 milliwatt DHCP Dynamic Host Configuration Protocol DNS Domain Name Server DRAM Dynamic Random Access Memory DSL Digital Subscriber Loop DSP Digital Signal Processor DTAS Data Terminal Alert Signal same as CAS DTMF Dual Tone Multiple Frequency FQDN Fully Qualified Domain Name GW Gateway ITU International Telecommunication Union HTML Hypertext Markup Language HTTP Hypertext Transfer Protocol HTTPS HTTP over SSL ICMP Internet Control Message Protocol IGMP Internet Group Management Protocol ILEC Incumbent Local Exchange Carrier IP Internet Protocol ISP Internet Service Provider ITSP IP Telephony Service Provider IVR Interactive Voice Response LAN Local Area Network LBR Low Bit Rate LBRC Low Bit Rate Codec MC Mini Certificate MGCP Media Gateway Control Protocol 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 93 LINKSYS A Division of Cisco Systems Inc MOH Music On Hold MOS Mean Opinion Score 1 5 the higher the better ms M
45. Invalid Remote party hangs up or error while attempting outbound call o Busy the line is being used by another station shared line only Call State A Call Appearance State followed by the term Call For example a Ringing Call a Dialing Call Active Call State If the state is Dialing Calling Proceeding Connected or Invalid When the state of the Call is Active it is referred to as an Active Call Standby Call State the state is Ringing or Holding When the state of the Call is Standby it is referred to as a Standby Call Key Any one of the keys on the SPA941 keypad A Key has two states down when pressed and up when released Button a Key with an on and off state The on off state toggles when the key is pressed down where applicable There are 3 buttons on the SPA941 SPEAKER HEADSET and MUTE Each button has an associated LED that indicates the on off state SK abel Soft key with the given label such as SK select SK cancel SK dial SK conf SK xfer 0000000 0 Call Features Selecting Audio I O Device and Line There are three sets of audio I O devices available a Handset b Built in microphone and speaker and c External microphone and headset The speaker is also used for ringing For convenience where the context is clear these are referred to as b the speaker and c the headset 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 15 LINKSYS
46. LINKSYS A Division of Cisco Systems Inc Linksys Inc SPA941 Administration Guide October 2005 Version 0 1 DRAFT 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 1 LINKSYS A Division of Cisco Systems Inc Disclaimer Please Read This document contains implementation examples and techniques using Linksys Inc and in some instances other company s technology and products and is a recommendation only and does not constitute any legal arrangement between Linksys Inc 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 your own opinions and decision regarding your particular situation As well Linksys reserves the right to change the features and functionalities for products described in this document at any time These changes may involve changes to the described solutions over time Use of Proprietary Information and Copyright Notice This document contains proprietary information that is to be used only by Linksys customers Any unauthorized disclosure copying distribution or use of this information is prohibited Please Note Design and specifications are subject to change without notice 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Noti
47. Packet Error Number of RTP packets received that are invalid Mapped RTP Port NAT mapped RTP port Report Generation and Event Logging The SPA reports a variety of status and error reports to assist service providers to diagnose problems and evaluate the performance of their services The information can be queried by an authorized agent using 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 88 LINKSYS A Division of Cisco Systems Inc HTTP with digested authentication for instance The information may be organized as an XML page or HTML page Error and Log Reporting The SPA Error Status Code ESC is used to indicate the current operation status of the SPA unit An error state can be a relatively long transient state or a steady state The state is also represented by a special blinking pattern of the Status LED next to the RJ 11 ports The Error Status Code is a 4 digit number The first digit indicates the error class 1xxx represents normal operation states while 2xxx 9xxx represent error states that must be fixed for the unit to function properly The status code values can be read from the IVR option XXX or from the SPA web page e LED Blink Rate Definitions ON LED remains solid on OFF LED remains solid off LONG Long On 3 0s on 1s off continuously FAST 0 125s on 0 125s off continuously SLOW 0 5s on 0 5s off continuously V
48. RTP Private Key gt parameter which should be hidden from the SPA s web interface like a password Since the secure call establishment relies on exchange of information embedded in message bodies of SIP INFO requests responses the service provider must maker sure that their infrastructure will allow the SIP INFO messages to pass through with the message body unmodified Linksys provides a configuration tool called gen_mc for the generation of MC and private keys with the following syntax gen_mc lt ca key gt lt user name gt lt user id gt lt expire date gt Where ca key is a text file with the base64 encoded 1024 bit CA private public key pairs for signing verifying the MC such as 9CCY9aY U1 X5 JUU EBZmi3AmcqE9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87 PiJINKW8XQj3B9Qqe3V gYxWCQNa335Y CnDsenASeBxuMIEaBCYd11I1fVEodJZOGwXwfAdeOMhcbDOkj7LVizcsTyk2TZYTccnZ7 5TuTjj13qvYs 5nEtOrkCa84 mEwl3D9tSvVLyliwQ u Hd C8u5SNk7hsAUZaA9TqH8lw0J lqSrsf6scsmundY5j7Z5mMK5J 9uBxSB8t8vamFGDOpF4zhNtbrVvIXkKI9kmp4vph1 C5jzO9gDfs38MF zjy YrVUFdM pxXtDBxmM fGUfrpAu Xb7 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 Maximum length is 16 characters expire date is the expiration date of the MC such as 00 00 00 1 1 34 34 2034 Internally the date is
49. SLO Very Slow 1 0s on 1 0s off continuously HB Heart Beat 0 1s on 0 1s off 0 1s on 1s off continuously Note The Link LED will blink on transmit and receive TX RX of packets The LED will display solid off if no link is available The LED will display solid on if link is up but no TX RX activity is present Syslog and Debug Server Records The SPA supports detailed logging of all activities for further debugging The debug information may be sent to a configured Syslog server Via the configuration parameters the SPA allows some settings to select which type of activity events should be logged for instance a debug level setting e Troubleshooting Phone does not turn on or boot up If the phone does not turn on or boot up please execute the following steps Step 1 Check all the connections into the phone 5 Volt Power supply RJ 45 Ethernet connection and Handset cord Step 2 Check the Wall socket and confirm that AC power is available Try plugging another electrical device into the socket to confirm power 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 89 LINKSYS A Division of Cisco Systems Inc Step 3 Contact Network Administrator or Service Provider Tell them that the phone does not boot up and that all the connections including the AC have been checked Please have the Model SPA941 Serial Number and telephone number ready for the su
50. SPA941 obtains the current time information from one of the following ways e NTP Server One or two NTP servers can be configured to the phone When it first boots up the phone will try to contact the NTP server to get the current time Then the phone periodically synchronizes the current time with the NTP server The synchronization period is fixed at 1 hour In between the update the phone tracks the time with its own internal clock e SIP Messages Each SIP message request or response sent to the SPA may contain a Date header with the current time information If the header is present the SPA will use it to update its current time e Manual Setup SPA941 also allows the user to manually enter the current time and date from the phone GUI or from the web page However this value will be overridden by the NTP time or 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 29 LINKSYS A Division of Cisco Systems Inc SIP Message Date whenever they are presented to the phone Manual setup requires the user to enter the time in 24 hour format only Note that the time served by the NTP Server and the SIP Date Header are GMT time The local time is obtained by offsetting the GMT according to the time zone of the region lt Time Zone gt can be configured from the web page or provisioning This time can be further offset by lt Time Offset HH mm gt parameter which must be entered in 24
51. The Dial_Plan parameters contain the actual dial plan scripts for each extension Dial Plan Digit Sequences The plans contain a series of digit sequences separated by the character The collection of sequences is enclosed in parentheses and When a user dials a series of digits each sequence in the dial plan is tested as a possible match The matching sequences form a set of candidate digit sequences As more digits are entered by the user the set of candidates diminishes until only one or none are valid Any one of a set of terminating events triggers the SPA to either accept the user dialed sequence and transmit it to initiate a call or else reject it as invalid The terminating events are e Nocandidate sequences remain the number is rejected e 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 rejected e A timeout occurs the digit sequence is accepted and transmitted as dialed if incomplete or transformed as per the dial plan if complete e An explicit send user presses the key the digit sequence is accepted and transmitted as dialed if incomplete or transformed as per the dial plan if complete The timeout duration depends on the matching state If no candidate sequences
52. Therefore it is important to disable the use of G 729a in order to guarantee the support of 2 simultaneous G 723 G 726 codec Dial Plan Parameter Name Description Type Default DIAL PLAN Per line dial plan script DialPlanScript See below ENABLE IP DIALING Enable le Dialing Bool See the previous section for explanation of Dial Plan Script syntax 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 73 LINKSYS A Division of Cisco Systems Inc Default Dial Plan script for each line Xx 3469 1 1 0 00 2 9 xxxxxx 1 xxx 2 9 XXxXXXX XXXXXXXXXXXX Explanation of Default Dial Plan Dial Plan Entry Functionality Tax Allow arbitrary 2 digit star code 3469 11 Allow x11 sequences 0 Operator 00 Int l Operator 2 9 xxxxxx US local number 1xxx 2 9 Xxxxxx US 1 10 digit long distance number XXXXXXXXXXXX Everything else Int l long distance FWD Note If IP dialing is enabled one can dial user id a b c d port where and 7 are dialed by entering user id must be numeric like a phone number and a b c d must be between 0 and 255 and port must be larger than 255 If port is not given 5060 is used Port and User ld are optional If the user id portion matches a pattern in the dial plan then it is interpreted as a regular phone number according to the dial plan The INVITE messag
53. This equipment has been tested and found to comply with the limits for a Class B digital device in accordance with the specifications in part 15 of the FCC rules This product bears the CE Marking indicating compliance with the 89 336 EEC directive Standards to which conformity is Declared EN 61000 4 2 1995 EN 61000 4 3 1997 EN 61000 4 4 1995 EN 61000 4 5 1995 EN 61000 4 6 1996 EN 61000 4 8 1994 EN 61000 4 11 1994 EN 61000 3 2 2001 EN 61000 3 3 1995 amp EN 55022 1998 Class B Modifications to this product not authorized by Linksys could void FCC approval thereby terminating end user authority to use this product For indoor use only Read installation instructions before connecting to a power source The electric plug and socket must be accessible at all times as this is the main method to disconnect power from the device Shock Hazard Do not operate near water or similar fluid Do not work with this device during periods of lightning activity Do not touch wires at the end of cables or within sockets One Year Limited Hardware Warranty Linksys provides a one 1 year limited hardware warranty Linksys warrants to customer that this product will conform to its published specifications and will be free from defects in material and workmanship at the time of delivery and for a period of one year thereafter Without limiting the 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 8 LINKSYS
54. a a i aia 91 19 CARE FOR THE SPA941 PHONE 92 Do not expose the phone to heat sun cold water 92 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 6 LINKSYS A Division of Cisco Systems Inc Cleaning the ORO Me EE 20 ACRONYMS 21 GLOSSARY 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 LINKSYS A Division of Cisco Systems Inc e How to use this document This Administration Guide provides instructions on managing and operating the SPA941 IP Telephone The Guide covers the following topics General Operating Information Call Voice Signaling Features and Hardware Functionality Descriptions Configuration Performance and Troubleshooting Guidelines Provisioning The SPA941 User Guide provides more details on how to use the phone s features via the phone s physical interfaces including the softkeys display menus the key pad and dedicated buttons The Administrative Guide is helpful for network administrators that need detailed information on how to configure the phone from the phone s Web Server or need access to the phone s troubleshooting and performance functions Finally the document also covers provisioning of the SPA941 e Important Operating Information Please insure the safe operation of the SPA941 by following the operating instructions in this section Compliance and Safety Information
55. al dialing When a complete number is entered the SPA sends a blind REFER to 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 62 LINKSYS A Division of Cisco Systems Inc the holding party with the Refer To target equals to 98 lt target_number gt This feature allows the SPA to hand off a call to an application server to perform further processing such as call park Notes The codes should not conflict with any of the other vertical service codes internally processed by the SPA You can empty the corresponding code that you do not want to SPA to process Feature Dial Services Codes One or more code can be configured into this parameter such as 72 or 72 74 67 82 etc Max total length is 79 chars This parameter applies when the user has a dial tone ist or 2nd dial tone Enter code and the following target number according to current dial plan entered at the dial tone triggers the SPA to call the target number prepended by the code For example after user dials 72 the SPA plays a prompt tone awaiting the user to enter a valid target number When a complete number is entered the SPA sends a INVITE to 72 lt target_number gt as in a normal call This feature allows the proxy to process features like call forward 72 or Block Caller ID 67 Notes The codes should not conflict with any of the other vertical service codes intern
56. aller ID on all outbound calls Apply to all extensions Unblock Caller ID Per Call 82 Unblock caller ID on the next outbound call on the current call appearance only Secure All Calls 16 Default to prefer to use encrypted media for all outbound calls Apply to all extensions Secure No Calls 17 Default to prefer to use unencrypted media for all outbound calls Apply to all extensions Secure Next Call 18 Prefer to use encrypted media for the next outbound call on this call appearance only Do Not Secure Next Call 19 Prefer to use unencrypted media for the next outbound call on this call appearance only Do Not Disturb 78 Apply to all extensions Cancel Do Not Disturb 79 Apply to all extensions Block Anonymous Calls Apply to all extensions Cancel Block Anonymous Calls Apply to all extensions Blind Transfer 98 Prefer G711u 017110 Set preferred codec to G711u for next outbound call Prefer G711a 017111 Set preferred codec to G711a for next outbound call Prefer G723 01723 Set preferred codec to G723 for next outbound call Prefer G726r16 0172616 Set preferred codec to G726r16 for next outbound call Prefer G726r24 0172624 Set preferred codec to G726r24 for next outbound call Prefer G726r32 0172632 Set preferred codec to G726r32 for next outbound call Prefer G726r40 0172640 Set preferred codec to G726r40 for next outbound call
57. alls A Missed Call is logged for each incoming call that causes the phone to ring but not answered From the Phone GUI the user can View a call log 1 Press SETUP 2 Select Call History 3 Select the log you want to view Delete a log 1 High light the entry 2 Press SK del Edit a log 1 High light the entry 2 Press SK edit 3 Press SK save to save the changes or SK cancel to abort the changes Call a log entry 1 High light the entry 2 Press SK dial or Off Hook or Turn on Speaker or Turn on Headset or Press a Line Key Save a log entry into Personal Directory if the directory is not already full 1 High light the entry 2 Press SK save 3 Modify the entry as needed 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 25 LINKSYS A Division of Cisco Systems Inc 4 Press SK save to save the entry in the personal directory The SPA941 keeps up to 60 entries per log in reverse chronological order The logs are saved in the phone s non volatile memory All call logs can be viewed on the SPA941 web page also using the link http lt ip address calllog htm Each log is shown on the web page with the syntax ext id name phone time stamp e _ext id is the extension on which the call is made from or received at E1 E2 E3 or E4 If En is not present it implies E1 e name is the call peer s name If this field is missing the peer s na
58. ally processed by the SPA You can empty the corresponding code that you do not want to SPA to process You can add a paramter to each code in Features Dial Services Codes to indicate what tone to play after the code is entered such as 72 c 67 p Below are a list of allowed tone parameters note the use of back quotes surrounding the parmeter w o spaces c lt Cfwd Dial Tone gt d lt Dial Tone gt m lt MWI Dial Tone gt o lt Outside Dial Tone gt p lt Prompt Dial Tone gt s lt Second Dial Tone gt X No tones are place x is any digit not used above If no tone parameter is specified the SPA plays Prompt tone by default Str79 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 63 LINKSYS A Division of Cisco Systems Inc If the code is not to be followed by a phone number such as 73 to cancel call forwarding do not include it in this parameter In that case simple add that code in the dial plan and the SPA will send INVITE 73 as usual when user dials 73 Notes 1 These codes automatically appended to the dial plan It is therefore not necessary to explicitly include them in dial plan 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 64 LINKSYS A Division of Cisco Systems Inc Outbound Call Codec Selection Codes
59. apital investment or maintenance charges The LEC charges a monthly fee to the customer who must agree to sign a term agreement 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 The standard for this equipment is set by the FCC and the equipment is supplied by an interconnect company Dedicated Access Customers have direct access to the long distance provider via a special circuit T1 or private lines The circuit is hardwired from the customer site to the POP and does not pass through the LEC switch The dial tone is provided from the long distance carrier Dedicated Access Line DAL Provided by the local exchange carrier An access line from the customer s telephone equipment directly to the long distance company s switch or POP Demarcation Point This is where the LEC s ownership and responsibility wiring equipment ends and the customer s responsibilities begin Direct Inward Dialing DID Allows an incoming call to bypass the attendant and ring directly to an extension Available on most PBX systems and a feature of Centrex service Dual Tone Multifrequency DTMF Better known as the push button keypad DTMF rep
60. arance on hold e Remote Undefined The share call state is not known this station is waiting for a notification from the application server e Registration Failed This station has failed to register with the proxy server for the corresponding extension e Registering The station is attempting registration with the proxy server for the corresponding extension e Disabled This Line Key on this station is disabled e Call Back A call back repeat dialing operation is currently active on this call appearance The Line Key LED supports 4 colors Red r Green g Amber a and Off 0 The SPA941 has 4 built in blinking patterns defined e No Blink nb Steady on or off e Slow Blink sb 500ms on 500ms off e Fast Blink fb 100ms on 100ms off e Double Blink db 100ms on 100ms off 100ms on 700ms off The administrator can also define arbitrary blinking pattern by following the syntax of a LedScript The default LedScript for each call appearance state is described in section 3 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 21 LINKSYS A Division of Cisco Systems Inc Other Supplementary Services The SPA941 supports the following supplementary Services Block Caller ID If enabled the SPA941 will attempt to hide the caller ID on all outbound calls by default The user can also enable or disable this feature on a per call basis by pre dialing a code before maki
61. are as yet complete as dialed the Interdigit_Long_Timeout applies If a candidate sequence is complete but there exists one or more incomplete candidates then the Interdigit_Short_Timeout applies 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 78 LINKSYS A Division of Cisco Systems Inc White space is ignored and may be used for readability Digit Sequence Syntax Each digit sequence within the dial plan consists of a series of elements which are individually matched to the keys pressed by the user Elements can be one of the following Individual keys 0 1 2 9 The letter x matches any one numeric digit 0 9 A subset of keys within brackets allows ranges set T e g 889 means 3 or 8 or 9 Numeric ranges are allowed within the brackets digit digit e g 2 9 means 2 or 3 or or 9 Ranges can be combined with other keys e g 235 8 means 2 or 3 or 5 or 6 or 7 or 8 or Element repetition Any element can be repeated zero or more times by appending a period character to the element Hence 01 matches 0 01 011 0111 etc Subsequence Substitution A subsequence of keys possibly empty can be automatically replaced with a different subsequence using an angle bracket notation
62. at the same time Pressing Submit All Changes will apply all the modifications Important Note switching between page tabs won t apply the changes to SPA The only way to apply the changes is to press the Submit All Changes button If the Undo All Changes button is clicked any modifications to profile parameters on any and all pages will be reset back to their original values before modification NOTE Pressing the Undo All Changes has no effect on the SPA it will only reset the values on the web page Web Interface Administration Privileges The SPA supports two levels of administration privileges Administrator and User both privileges can be password protected Important note by factory default there are no passwords assigned for both Administrator and User The Administrator has the privilege to modify all the web profile parameters and can also modify the passwords of both Administrator and User A User only has the privilege to access part of the web profile parameters the parameter group that User can access is specified by the Administrator which can only be done through provisioning To access the Administrator level privilege use URL http IP_Address Of SPA admin If the password has been set for Administrator the browser will prompt for authentication The username for Administrator is admin and cannot be changed To access the User level privilege use URL http IP_Address Of SPA If th
63. atch is found and the name field is present in the matched entry it will replace the current caller ID name and will be shown on the call screen as the calling peers name The same name will also go into the incoming call log If a match is not found or the name field is not present in the matched entry the current caller ID name will be used if it exists Dialing Assistance Dialing assistance is an option the user can enable or disable under the Preferences menu If the option is enabled the phone will show up to ten potential matches from the Redial List most recent first and from the Personal Directory as the user dials the target number This feature applies to both off hook with dial tone and on hook without dial tone dialing When the suggestion list appears the user can scroll down the list using the Up Down key to select the desired target number from the list When looking for matches from the Redial List and the Directory the SPA941 will skip any leading codes so that the match is purely based on the target number The code that will be skipped are if configured in the phone e Al call forward activation and deactivation codes Secure call activation and deactivation codes Block CID activation and deactivation codes Block anonymous call activation and deactivation codes DND activation and deactivation codes All the Prefer codec and Force codec codes Referral services codes and Feature Dial Services codes Time Date The
64. ates the default value of c a Registering LED Pattern when the corresponding extension is trying to LedScript register with the proxy server Leaving this entry blank indicates the default value of c r p s Call Back Active Call Back operation is currently active on this call appearance LedScript Supplementary Services Enable or disable the corresponding supplementary services on the phone Parameter Name Description Type Default Conference Enable disable Three way conference service Bool Yes Attn Transfer Enable disable attended call transfer service Bool Yes Blind Transfer Enable disable blind call transfer service Bool Yes Do Not Disturb Enable disable do not disturb service Bool Yes Block Anonymous Enable disable block anonymous call service Bool Yes Call Call Back Enable disable call back a k a repeating dialing service Bool Yes Block Caller ID Enable disable blocking outbound Caller ID service Bool Yes Secure Call Enable disable secure call service Bool Yes CFWD All Enable disable call forward all service Bool Yes CFWD On Busy Enable disable call forward on busy service Bool Yes CFWD On No Enable disable call forward on no answer service Bool Yes Answer Ring Tone Each entry defines a ring tone to be used on the phone with an ID between 1 and 10 The ID can be used in a DirEntry to indicate which ring tone to use when the corresponding caller calls Parameter
65. by a call back request Yes or No Peer Name Name of the peer Peer Phone Phone number of the peer Duration Duration of the call in hr min sec format Packets Sent Number of RTP packets sent Packets Recv Number of RTP packets received Bytes Sent Number of RTP bytes sent Bytes Recv Number of RTP bytes received Decode Latency Decoder latency in milliseconds 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 76 LINKSYS A Division of Cisco Systems Inc Jitter Receiver jitter in milliseconds Round Trip Delay Network round trip delay ms available if the peer supports RTCP Packets Lost Total number of packets lost Packet Error Number of RTP packets received that are invalid Mapped RTP Port NAT mapped RTP port Downloaded Ring Tone Status Indicates whether the phone is downloading a ring tone and from where or idle Ring Tone 1 Information about the user downloaded ring tone 1 name size and time stamp of the tone Ring Tone 2 Information about the user downloaded ring tone 2 name size and time stamp of the tone e Dial Plan The SPA allows each extension to be configured with a distinct dial plan The dial plan specifies how to interpret digit sequences dialed by the user and how to convert those sequences into an outbound dial string The SPA syntax for the dial plan closely resembles th
66. ce on Page 2 2 LINKSYS A Division of Cisco Systems Inc Table of Contents 1 HOW TO USE THIS DOCUMENT sssssssssssnssnsnesistnttsstninstastattnstnnttnttnttnttnttnstantnntns tnn snn nannan nnn nenene 8 2 IMPORTANT OPERATING INFORMATION 8 Compliance and Safety Information sccceecceecceeeeeeesneeeneeeeeeeeee sae saneeeeeseeeseaeeesaesaaeeeeseeesescaesaseeeenseeeneas 8 Network and Service Configuration SettingS ccssccssseeeseeseeseeseeeeeeeeeeeseeseseeeenseeeeeseeesnaeseneeeeeseeneas 10 3 SA OVERVIEW ee EE asta hasta ovat ee 10 SPA941 Hardware Featureser greus eEeEeeE EE pd scdecdetenctecdecadesuceneubecgscnveshoceecceceasceepecadevcucuss 11 SPA941 FUNCtIOMalities icra fees fk aoe ee Ee Ee 12 Navigating the SPA941 GUI ccceccesseeeeseeeeteeeseeeeneeeeseaeseesneeeneeeesaaesesaaesqseeeeeaeeesaaesaseeeeneeeeeseaesesanenseeeees 13 Definitions for the Four Call Appearances ccccsccccseeeseeeeeeeeeseeeseseeeeneeeeeseaeseseeeaseeeeseeeeseaesaseeenseeeneas 14 A SCALE MR 15 Selecting Audio UO Device and Litne cccssccssseeceseeeeeseeeseeeenseeeeesaeseseeeeeseeeeeeneeseaesaseeeenseeeseeeseseaeeeeseees 15 Making fl Or UE 16 Elle Ke UE 17 Ending Gallls EE 17 FIOM ANd Reeume Eege eege eege SEENEN 17 Call Walting EE 17 Three Way COmference ccssecceseeeeeseeesseeeseeeenseeeseaesesneeensneeeneeeesaaeseseeesaseeeessaesasueesaanaeeseeseseassasneeenseaeeeaees 18 Attended Call Transfer iinei enn
67. ching Power Adapter LED Test Function SPA941 Features and Functions e Upto Four Call Appearances with Independent Configuration and Registration The SPA941 ships with two line appearances enabled A two line upgrade is available via a software license key installed locally using the SPA941 web interface or installed remotely via a secure profile update Pixel Based Display 128x64 Monochrome Graphical Liquid Crystal Display LCD Line Status Active Line Indication Name and Number Menu Driven User Interface Digits Dialed with Number Auto Completion Shared Line Appearance Full Duplex Speakerphone Call Hold Music on Hold Call Waiting Caller ID Name and Number Outbound Caller ID Blocking Call Transfer Attended and Blind Call Conferencing Automatic Redial On hook Dialing Call Pick Up Selective and Group Call Park and UnPark Call Swap Call Back on Busy Call Blocking Anonymous and Selective Call Forwarding Unconditional No Answer On Busy Hot Line and Warm Line Automatic Calling 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 LINKSYS A Division of Cisco Systems Inc Call Logs 60 entries each Made Answered and Missed Calls Redial from Call Logs Personal Directory with Auto dial 100 entries Called Number with Directory Name Matching Call Number using Name Directory Matching Log Matching or via Caller ID Subsequent Incoming
68. ciscdecesc ceded sabes cetera a aaae ara aaa aea a aaa aa Aa aaa a Anaa EE 36 Voice AIQOritTimS i seerrsresinren diene eee ccecdce deca teete cccticeeteaet TOKO ENAR ENAREN aN KELTAN AAO KAREENA ENEE seacacestead stesdedacereagtecensau 37 G 711 Ataw and My Taw ET 37 EE 37 GT QA EE 37 EN EE 37 Codec ele Hentgen 37 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 4 LINKSYS A Division of Cisco Systems Inc dTM 37 Adjustable Audio Frames Per Packet s ecccsseeeeeesseeeeeenseeeeeenseneeeeenseeeeenseneeeegseeeeeeaseneensaseeeeeenseseeeees 37 DTMF In band A Out of Band RFC 2833 SIP INFO cccsssssssessseeeeseeeeeeeeeseeseseeeenseeeesnaesenenenseeeneas 38 Call Progress Tone Generation eege ee Ee Ee geed eege 38 Call Progress Tone Pass Through ccccsecssseeeeseeeeeeeeesneeeeseeeeneeeeesaesesneeeeeeeeseaesasaaesaseaeeneeeeseaeseseeeeeneeees 38 Jitter Buffer Dynamic Adaptive ccccceeeeeeeseeeeeeeseeeeeenseeeeeensneeeenenseeeeenseeeeeeeseeeeeeaseneeeseseeeenseseeeeenens 38 Voice Activity Detection with Silence Suppression amp Comfort Noise Generation sseee 38 Configurable Dial Plan with Interdigit Timers csssccsseeeeseesseeeeseeeeneeeeeseaeseseeeenseeeseesaseaesnseeeeesnaess 38 Network Address Translation NAT Traversal cceseeesseseeeseseeeeneeseseeeeeseeeeeeseseeeeeseseseeeseseeneeeeseenenes 40 VOIP NAT Intenworkin
69. controller may have be configured improperly e The transport network has performance issues Please try to gather the following information Step 1 Identify when and where the issue occurred Please look at the Redial List calls that the user made by pressing menu 3 1 and write down the calls that were affected Then look at the Answered Calls by pressing menu 3 2 and write down the calls that were affected Step 2 Identify the type of voice quality issue i e part of conversations dropped out or were lost excessive echo unclear or garbled speech one way conversation low volume Step 3 Get the Current IP Address of the phone To get the Current IP Address please press menu 9 2 The phone will display the Current IP Address and please write it down To exit from this menu option press the menu button Step 4 Contact the network administrator or service provider Be prepared to provide the call log information and the voice quality issue type Additionally please have the Model SPA941 Serial Number telephone number Current IP Address and Registration Status Registered Not Registered ready for the support personnel The Serial number is printed on the bottom of the phone e Customer Support Methods For support on the SPA941 IP Telephone please use the following methods First contact the VAR dealer from whom you purchased your phone s Authorized service providers and VAR dealers
70. e Enable Web Admin Enable disable Admin pages of web server of SPA Bool Yes Access Admin Password The password for administrator Str63 User Password The password for User Str63 Network Configuration Parameter Name Description Type Default DHCP Enable Disable DHCP Bool Yes Host Name Host Name of SPA Str31 Domain The network domain of SPA Str127 Static IP Static IP address of SPA which will take effect if DHCP IP 0 0 0 0 is disabled NetMask The NetMask used by SPA when DHCP is disabled IP 255 255 255 0 Gateway The default gateway used by SPA when DHCP is IP 0 0 0 0 disabled Primary DNS DNS server used by SPA in addition to those supplied IP 0 0 0 0 by DHCP server if DHCP is enabled See DNS Server Order for more information when DHCP is disabled this will be the primary DNS server Secondary DNS DNS server used by SPA in addition to those DHCP IP 0 0 0 0 server supplied if DHCP is enabled See DNS Server Order for more information when DHCP is disabled this will be the secondary DNS server DNS Server Order Controls how to organize the DNS servers supplied by Choice Manual DHCP server and the ones statically configured it is used only when DHCP is enabled Manual Use statically configured DNS servers if there is any otherwise use the ones supplied by DHCP server 2003 2005 Linksys a Divisi
71. e however is still sent to the outbound proxy if it is enabled User Parameters Call Forward Parameter Name Description Type Default Cfwd All Dest Forward number for Call Forward All Service Phone Cfwd Busy Dest Forward number for Call Forward Busy Service Phone Cfwd No Ans Dest Forward number for Call Forward No Answer Service Phone Cfwd No Ans Delay Delay in sec before Call Forward No Answer triggers Uns8 20 Speed Dial Parameter Name Description Type Default 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 74 LINKSYS A Division of Cisco Systems Inc Speed Dial 2 Target phone number or URL assigned to speed dial 2 Phone Speed Dial 3 Target phone number or URL assigned to speed dial 3 Phone Speed Dial 4 Target phone number or URL assigned to speed dial 4 Phone Speed Dial 5 Target phone number or URL assigned to speed dial 5 Phone Speed Dial 6 Target phone number or URL assigned to speed dial 6 Phone Speed Dial 7 Target phone number or URL assigned to speed dial 7 Phone Speed Dial 8 Target phone number or URL assigned to speed dial 8 Phone Speed Dial 9 Target phone number or URL assigned to speed dial 9 Phone Suppl
72. e 2 20 LINKSYS A Division of Cisco Systems Inc Line Key LED Behavior The following outlines the Line Key LED Behavior e ALine Key corresponds to a call appearance Call appearances for the same extension are numbered in ascending order of their Line Key position starting at 1 The Line Key LED color and blinking pattern are fully programmable The administrator can specify a different color and pattern for each well defined states of the call appearance These states are described below e idle This call appearance is not in use The user can use it to make outbound call on this station e Local Seized This call appearance has been seized by this station to prepare for an outbound Call e Local Progressing This station is making an outbound call that is progressing e Local Active This station is engaged in a connected call on this call appearance e Local Ringing This station is ringing for an incoming call on this call appearance e Local Held This station has placed this call appearance on hold e Remote Seized This call appearance has been seized by another station to prepare for an outbound call e Remote Progressing Another station is making a call on this call appearance and is progressing e Remote Active Another station is engaged in a connected call on this call appearance e Remote Ringing Another station is ringing for an incoming call to this call appearance e Remote Held Another station has placed this call appe
73. e 2 37 LINKSYS A Division of Cisco Systems Inc DTMF In band amp Out of Band RFC 2833 SIP INFO The SPA 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 Call Progress Tone Generation The SPA has configurable call progress tones Parameters for each type of tone may include number of frequency components frequency and amplitude of each component and cadence information Call Progress Tone Pass Through This feature allows the user to hear the call progress tones such as ringing that are generated from the far end network Jitter Buffer Dynamic Adaptive The SPA can buffer incoming voice packets to minimize out of order packet arrival This process is known as jitter buffering The Jitter Buffer size will proactively adjust or adapt in size depending on changing network conditions The SPA has a Network Jitter Level control setting for each line of service The jitter level decides how aggressively the SPA will try to shrink the jitter buffer over time to achieve a lower overall delay If the jitter level is higher it shrinks more gradually If jitter level is lower it shrinks more quickly Voice Activity Detection with Silence Suppression amp Comfort Noise Generation Voice Activity Detection VAD and Silence Suppression is a means of i
74. e Call Appearance gt for that call appearance If the extension is shared then whether or not a call appearance assigned to this extension is shared follows the setting of lt Share Call Appearance gt for that call appearance 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 69 LINKSYS A Division of Cisco Systems Inc NAT Settings Parameter Name Description Type Default NAT Mapping Enable If set to yes the SPA will substitute private address port Bool No with external address port where appropriate Otherwise the SPA will use the private address port only NAT Keep Alive Enable If set to yes the configured lt NAT Keep Alive Msg gt is Bool No sent periodically every lt NAT Keep Alive Intvl gt seconds NAT Keep Alive Msg Contents of the keep alive message to be sent to a given Str31 NOTIFY destination periodically to maintain the current NAT mapping It could be an empty string If value is NOTIFY a NOTIFY message is sent as keep alive If value is REGISTER a REGISTER message w o Contact is sent If value is PING a PING message is sent For all other values the string is sent as specified NAT Keep Alive Dest Destination to send NAT keep alive messages to If value FQDN PROXY is PROXY it will be sent to the current proxy or outbound proxy Network Settings
75. e Copyright Notice on Page 2 39 LINKSYS A Division of Cisco Systems Inc Network Address Translation NAT Traversal The SPA941 supports NAT traversal for VoIP signaling and media packets This section overviews the basics around NAT The SPA941 NAT features are reviewed in the configuration section Why NAT A NAT allows multiple devices to share the same external IP address to access the resources on the external network The NAT device is usually available as one of the functions performed by a router that routes packets between an external network and an internal or private one A typical application of a NAT is to allow all the devices in a subscriber s home network to access the Internet through a router with a single public IP address assigned by the ISP The IP header of the packets sent from the private network to the public network can be substituted by the NAT with the public IP address and a port selected by the router according to some algorithm In other words recipient of the packets on the public network will perceive the packets as coming from the external address instead of the private address of the device where the packets are originated In most Internet protocols the source address of a packet is also used by the recipient as the destination to send back a response If the source address of the packets sent from the private network to the public network is not modified by the router the recipient may not be able
76. e SPA can be instructed to contact a SIP proxy server in a domain named in SIP messages The SPA shall consult the DNS server to get a list of hosts in the given domain that provides SIP services If an entry exists the DNS server will return a SRV record which contains a list of SIP proxy servers for the domain with their host names priority listening ports etc The SPA shall try to contact the list of hosts in the order of their stated priority Re registration with Primary SIP Proxy Server If the SPA is currently using a lower priority proxy server it should periodically probe the higher priority proxy to see if it is back on line and attempt to switch back to the higher priority proxy whenever possible It is very important that switching proxy server should not affect calls that are already in progress Codec Name Assignment Negotiation of the optimal voice codec is sometimes dependent on the SPA device s ability to match a codec name with the far end device gateway codec name The SPA allows the network administrator to 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 36 LINKSYS A Division of Cisco Systems Inc individually name the various codecs that are supported such that the correct codec successfully negotiates with the far end the equipment Voice Algorithms The SPA941 supports the following voice algorithms G 711 A law and mu law This very low complexit
77. e corresponding syntax specified by MGCP and MEGACO Some extensions are added that are useful in an end point The dial plan functionality is regulated by the following configurable parameters e Interdigit_Long_Timer e Interdigit_Short_Timer e Dal Plan unique per extension 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 Default 10 The Interdigit_Long_Timer specifies the default maximum time in seconds allowed between dialed digits when no candidate digit sequence is as yet complete see discussion of Dial_Plan parameter for an explanation of candidate digit sequences 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 77 LINKSYS A Division of Cisco Systems Inc Interdigit Short Timer ParName Interdigit_Short_Timer Default 3 The Interdigit_Short_Timer specifies the default maximum time in seconds allowed between dialed digits when at least one candidate digit sequence is complete as dialed see discussion of Dial_Plan parameter for an explanation of candidate digit sequences Dial Plan independent for each extension ParName Dial_Plan 1 Dial_Plan 2 Dial_Plan 3 Dial_Plan 4 Default xx 8469 11 O 00 2 9 xxxxxx 1xxx 2 9 XxxxxxSO XXXXXXXXXXXX
78. e password has been set for User the browser will prompt for User authentication The username for User is user and cannot be changed 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 43 LINKSYS A Division of Cisco Systems Inc When browsing Administrator pages one can switch to User privileges by click the link User Login Note if User password was set the browser will prompt for User authentication when you click User Login link On the other side from the User pages the user can switch to Administrator privilege by clicking the link Admin Login Authentication is needed if Administrator password has been set Warning Switching between the User and Administrator will discard the uncommitted changes that have already been made on the web pages Web Interface Basic and Advanced Views The web configuration interface provides a Basic and an Advanced view from which the various configuration parameters can be accessed The SPA Provisioning tab is only visible from the Advanced Administrator view of the web interface Warning Switching between the basic and advanced view will discard the uncommitted changes that have been made on the web pages e SPA941 Configuration Parameters This section tabulates all the SPA941 parameters The parameters can be configured in 2 ways a phone configuration web page and b remote provisioning In addition a small subs
79. e pattern 100ms on 100ms off 100ms on 100ms off 100ms on 900ms off 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 49 LINKSYS A Division of Cisco Systems Inc DirEntry A script that describes a directory entry Syntax n name p phone r ring tone id name is the name of the entry phone is the phone number or URL of the entry ring tone id is the ID of the ring tone 1 10 that should be played when the entry is called Example 1 n Joe Smith p 14089991 234 ring tone default Example 2 p 123456 192 168 2 12 5061 r 5 unknown name ProvisioningRuleSyntax Scripting syntax used to define configuration resync and firmware upgrade rules Refer to the provisioning discussion for an explanation of the syntax DialPlanScript Scripting syntax used to specify line 1 and line 2 dial plans Refer to the dial plan section of this document for an explanation 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 50 LINKSYS A Division of Cisco Systems Inc System Parameters System Configuration Parameter Name Description Type Default Restricted Access This feature is used when implementing software Str127 Domains customization Enable Web Server Enable disable web server of SPA Bool Yes Web Server Port TCP port through which the SPA web server will Uns8 80 communicat
80. ed Last Caller Number The number of the last caller Mapped SIP Port NAT Mapped SIP Port Call 1 2 Status State State of the call Idle Dialing Calling Proceeding Ringing Answering Connected Hold Holding Resuming or Reorder Tone Tone playing for this call Dial 2 Dial Outside Dial Ring Back Ring Busy Reorder SIT1 4 Call Waiting Call Forward Conference 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 FAX 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 Indicate whether the call is triggered by a call back request Yes or No Peer Name Name of the peer Peer Phone Phone number of the peer Duration Duration of the call in hr min sec format Packets Sent Number of RTP packets sent Packets Recv Number of RTP packets received Bytes Sent Number of RTP bytes sent Bytes Recv Number of RTP bytes received Decode Latency Decoder latency in milliseconds Jitter Receiver jitter in milliseconds Round Trip Delay Network round trip delay ms available if the peer supports RTCP Packets Lost Total number of packets lost
81. eeeeeesaeeeenseaeeeenseeeeeenseeeeeenneeneeenas 44 10 SPA941 CONFIGURATION DARAMETERG eeeeeeeeieeiesrissrissrissrssrrissrissriesrinstnnstnnttnnstnnnnntt 44 NOtatlONMS ee eegene Ee 45 ACHT 45 System Parameters dees dree dE a eea e aaae aa EE EH EES dE Ee 51 System Configuration ioeie aiiin iiaii idii iida i deet 51 Network Configuration assinei a aaia ai i iia ed a iai aiad ead 51 Provisioning SEET CT CET 52 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 5 LINKSYS A Division of Cisco Systems Inc SIP Parameters saan sees aonana ea pea aaa aaaea Neona aa panana da Papia iea Nanana sutteaeadecucuesuteeysieesuaestecced 54 Regional Parametere ccccssceeceesseeeeeessneeeeenseeeeeenseaeeeenseeeeeenseeeeeseseeeeeseaseaeeeseseaeeeseseneeeseseeneeseseeseeseseasen 59 Phone Parameters aico Lecce cei tescecis Jeccecestencces dectechiVecuecisdecccessdectechiVesacesdeuuecsseseccer desuechhtenccshvd 66 Extension 1 4 Parameters i dese 69 User Parameters o ciccciscccccciecccecescecvectecue dee EENEG 74 Info Parameters Read Opbvh 2 2ee0gN gege dE teneco ati estetetessciueteecleatiestacues 75 E Wee ET WEE 77 gt PROVISIONING OVERVIEW ei eee tiaces toes a aa a raa aa e Eege Edge 81 PFOVISIONIING EE 81 Provisioning CapabillitieS ccseccsseeeeseesseeeeseeeeneeeesesaeeeseeeeeeeeessaesesaeeenseeeeseesesaesaseeeseneeeeseaesaseenenseaeees 81 Configuration Profile ca Ee ee
82. efault value of c r p d LedScript Remote Ringing LED pattern during the Remote Ringing state where the shared call appearance is in ringing on another station Not applicable if the call appearance is not shared Leaving this entry blank indicates the default value of c r p d LedScript Remote Active LED pattern during the Remote Active state where another station is engaged in an active call on this shared call appearance Not applicable is this call appearance is not shared Leaving this entry blank indicates the default value of c r p d LedScript Remote Held LED pattern during the Remote Held state where another station has placed this call appearance on hold Not applicable if the call LedScript 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 67 LINKSYS A Division of Cisco Systems Inc appearance is not shared Leaving this entry blank indicates the default value of c r p s Disabled LED pattern when the Call Appearance is disabled not available LedScript for any incoming or outgoing call Leaving this entry blank indicates the default value of c 0 Register Failed LED pattern when the corresponding extension has failed to LedScript register with the proxy server Leaving this entry blank indic
83. eing used Input Mode Soft Key Cursor appending Cursor inserting Numeric num Blinking underscore _ Blinking vertical line Alphanumeric alpha Blinking full height block i Blinking open rectangle IP Address ue Blinking half height block 8 Blinking half rectangle L When entering digits during numeric mode just enter the corresponding digits from the key pad When entering letters or symbols during alphanumeric mode the phone will show a template of 2 or more choices of symbols as the user presses each digit key the user can scroll through the choices by pressing the same key multiple times To accept a symbol as input the user can a stop pressing the digit key for 1 5s b pressing another digit key c navigate to another input field if applicable by pressing the UP DOWN key or d save the value by pressing the proper soft key such SK save or SK ok When entering IP address pressing the digit will insert a dot or a colon if there are 3 dots entered already When entering numeric alphanumeric or IP address values the following additional soft keys will be available for easier editing e SK lt lt Move cursor to the left by 1 character e SK gt gt Move cursor to the right by 1 character e SK erase lt Erase the character to the left of the cursor 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 27 LINKSYS A Divis
84. ementary Service Settings Parameter Name Description Type Default Block CID Block Caller ID on off for all calls Bool No Block ANC Block Anonymous Calls on or off Bool No DND DND on or off Bool No Secure Call Setting If set to yes all outbound calls are secure calls by default Bool No Dial Assistance Enable disable dialing assistance If enabled the phone Bool No will show up to 10 potential matches from the Redial List most recent first and from the Personal Directory as the user is dialing the target number This feature applies to both off hook with dial tone and on hook without dial tone dialing The user can scroll down the list using the Up Down key to select the desired target number from the list Audio Volume Parameter Name Description Type Default Ringer Higher Louder Range is 1 16 Uns8 8 Speaker Higher Louder Range is 1 16 Uns8 8 Handset Higher Louder Range is 1 16 Uns8 8 Headset Higher Louder Range is 1 16 Uns8 Info Parameters Read Only System Information DHCP Enabled Disabled Current IP Current IP address of 941 Host Name Hostname of the 941 Domain Domain name used by 941 Current Netmask Current Network Mask Current Gateway Current Gateway Primary DNS Primary DNS server Secondary DNS Secondary DNS servers Product Information Product Name Production Name SPA941 2003 2005 Link
85. en the user presses the conf soft key or the xfer soft key the SPA941 will attempt to pick an idle call appearance that is on the same Extension as the last active call If that call appearance is not available the SPA941 attempts to pick one whose Extension has the same Proxy server as the last active call Finally if that fails the first idle call appearance is selected in the order L1 L2 L3 L4 Making Calls There are two steps in making a call 1 Select an audio device and 2 Dial the number The SPA941 allows the user to perform these two steps in either order In other words the user can select an audio device either before or after dialing The SPA941 supports two types of dialing explicit or implicit Explicit dialing refers to one of the following cases enter the target number digits one by one enter the speed dial assignment of the target number select an entry from a directory and press the dial soft key press a dedicated key such as the Voice Mail key Implicit dialing refers to dialing a highlighted entry from one of the device s directories or logs A number can be implicitly dialing by selecting an audio device while the number is highlighted on the display 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 16 LINKSYS A Division of Cisco Systems Inc If an audio device is selected prior to dialing then only explicit dialing is supported Both exp
86. enen se aeos e nanasan eona ecuetevcceststuirenuechsntey cases aeaaaee aa aaa andanan nha aneas 18 Blind Call Trans ten inii aaa aaa aaa aa EE EE Eed SEA aiei adatan 19 o IR Ette cnn a ak E E Ee ee Ee Ee E 19 Message Waiting Indication MWI ssusnsesuunnnnunnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnn nunne nnnnnnnnnnnn nnmnnn nnmnnn nnna 20 Accessing Voice Mail s ecccceseeeceeeseeeeeeeseeeeeeesneeeeensneeeeeesneeseensneeseeesneeeseesaneeseesanaesessaneeseesaneeeeeesneesesens 20 Ode Re UE 20 Shared Call ApDeatanCge eege SEENEN EES 20 Line Key LED BehavViol cccsseeccceseeeseseeeeeseeeeeeeseseeneeseneeeeeseseeneesesenneeseseseeeeseseseeeseseeneesasesneeseseeeeeseseenanes 21 Other Supplementary Services ccccscccsecceseeeeeeeeeeeneseseeeeeeneeeeeeeescaeseseeeeeeeeeseaesasaesaanaeeeeeeessnaeseseeeeneeees 22 Block Gallen ID incsccakvicecccccce r eaa a ear re Tear r a pe a arae eae ara aa ar a ae re aa aaa ea Anae Sa Eain Eoaea 22 Block Anonymous Ca a aa Taa ar aeaa Eesen EE ee 22 Do Not Disturb DN D EE 22 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 3 LINKSYS A Division of Cisco Systems Inc Sec re e EE 22 Secure Call Implementation ccsccsseecesceeeeeeseseeeenseeeeeeeeseneeeenseeeseaesasaeeeaseeeeeeeeesaesaseaeenseeeessaesaaeaeeneeees 22 USER IME ACES tistics gesiess eege Eegenen AAR 23 Service Provider Requirements A 24 5 MEMORY FEATURES
87. entire text string Definitions for the Four Call Appearances Some definitions e Station A SPA941 phone or a similar device in the Linksys SPA family with one or more Extensions and Call appearances provisioned e Ext An extension in the SPA941 is a VoIP account in a voice service provider VSP s network or an IP PBX system An extension can be uniquely identified with a User ID like a phone number that is unique within the VSP Up to 4 extensions can be configured in the SPA941 which are refered to as Ext 1 Ext 2 Ext 3 and Ext 4 or simply E1 E2 E3 and E4 respectively Note that the same extension can be configured on more than one station These extensions are called Shared Extensions Extension 1 is referred to as the Primary Extension Some features can only be activated on the Primary Extension such as Call Forwarding and Voice Mail Waiting Indicator VMWI e Call Appearance Physically a call appearance corresponds to a Line Key on a station There are four Line Keys on the SPA941 which are referred to as Line Keys 1 2 3 and 4 or L1 L2 L3 and L4 respectively Functionally a call appearance is an instance of an extension If an extension is assigned to Line Keys on multiple stations it is a Shared Line Appearance One 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 14 LINKSYS A Division of Cisco Systems Inc extension can be assigned to multiple Line Key
88. et of user options can be set via the phone s GUI under various setup menus On the configuration web page the parameters are organized into several groups with each group hidden under a tab on the top of the web page To see the group of parameters the user click on the corresponding tab on the web page The following groups are defined at present e Info Status Information read only This is the default tab when the web page is first loaded System System level parameters including network and debug parameters SIP Parameters to adjust the SIP stack behavior Provisioning Parameters that control remote provisioning Regional Parameters that depends on country or region such as call progress tones and codes Phone Parameters that apply to all the extensions configured for the phone Ext 1 Parameters that apply to Extension 1 Ext 2 Parameters that apply to Extension 2 Ext 3 Parameters that apply to Extension 3 Ext 4 Parameters that apply to Extension 4 User User level parameters other parameters are presumably administrator level Each parameter has a name and corresponding value A parameter name is of the form seg _seg n where seg is text segment composed of ascii characters only excluding space and underscore 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 44 LINKSYS A Division of Cisco Systems Inc indica
89. g ceceecceeeeeee ee eeeeee seen neee eee nnee see neee see neee see sees see neee sees gees seen sees nnmnnn nnmnnn nnmnnn nnmnnn nnana 40 7 DATA NETWORKING FEATURES SUPPOR ED eee eeeeaeeeeeeeeseaeeesaeeeeeeeeeeee 41 MAC Address IEEE E RE 42 IPv4 Internet Protocol Version 4 RFC 791 upgradeable to v6 RFC 188 42 ARP Address Resolution Protocol annern ennn nn 42 DNS A Record RFC 1706 SRV Record RFC 202 42 DiffServ RFC 2475 and ToS Type of Service RFC 791 1349 0 eeceeecceceeeeeeeeeeeeeeeceeeeeteaeeeeneeeeeaees 42 DHCP Client Dynamic Host Configuration Protocol RFC 2121 42 ICMP Internet Control Message Protocol HEC OO 42 TCP Transmission Control Protocol DEC Oo 42 UDP User Datagram Protocol DEC 681 42 RTP Real Time Protocol RFC 1889 RFC 1890 ceecceceeeeceeeeeeeeeeeeeeeceaeeeeaaeeseeeeeseaeeesaeeneeeseaees 42 RTCP Real Time Control Protocol RFC 19901 42 SRTP Secure Real Time Control Protocol RFC XNNXl 42 8 CONFIGURING AND PROVISIONING OVERVIEW seeseesssssseessesssrsssrrssrnsrrsstnnstnnnrnssennstnnssnnseennt 42 9 WEB INTERFACE ADMINISTRATION AND GECUDITN 42 Web Interface CONVENTIONS 2c s ccc Eeer eer ee SEELEN ee Seege 43 Web Interface Administration Privileges cccscccseccsseeeeeeeesseeseseeeeneeneeseaeseseeesaseeeeeeeeeseaeseneeeenseeeneaees 43 Web Interface Basic and Advanced VieWS ccssecccceeeeeceeeeeeeeeeeneeeeeennee
90. g the Hold key or by e pressing another Line Key to answer an incoming all resume a held call or start a new call When a call is on hold the corresponding Line Key LED slowly blinks red To resume a call that is on hold press the corresponding Line Key When the phone is hosting a three way conference two call appearances are in the connected state Pressing the Hold key places both calls on hold The calls can be individually brought back to the connected state by pressing the corresponding Line Keys Call Waiting The Call Waiting function is activated when a device has a call in the active state and another call is incoming The phones in the SPA series do not support multiple calls on the same Line Key Incoming calls are assigned to an unused Line Key causing the Line Key to quickly blink red Note that the Voice Mail Waiting Indicator also blinks red whenever there is an incoming call The phone will not ring However to alert the user the call waiting tone is played into the active audio device 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 17 LINKSYS A Division of Cisco Systems Inc The SPA941 allow multiple lines to simultaneously cause Call Waiting All four call appearances can ring at the same time after one of the Extension Keys is answered the other three call appearances will go into the Call Waiting state Three Way Conferencing There are two ways to init
91. he macro variable MA expands to the MAC address of each specific SPA941 e g MA cfg would expand to O0O0e08dabcde cfg for a unit with that MAC address Provisioning Proper Once the SPA941 is pre provisioned it will resync periodically to the specified profile URL maintaining its internal configuration synchronized with the provisioning server Typically a separate configuration file is maintained for each SPA941 in the network The periodic profile would contain individual information about each phone including account identifiers and credentials 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 84 LINKSYS A Division of Cisco Systems Inc Please refer to the Linksys SPA Provisioning Guide for a more exhaustive description of the available provisioning features in the SPA941 e Firmware Upgrade The SPA is firmware upgradeable via TFTP or HTTP Firmware loads are released as single binary files which contain all the modules pertaining to any one release version By convention the firmware loads are named with the extension bin e g spa941 bin Firmware upgrades are attempted only when the SPA is idle since they trigger a software reboot During a firmware upgrade the phone is out of service and cannot be used to initiate or receive calls The LEDs flash during the course of an upgrade providing progress indication To initiate a remotely provisioned upgrade program
92. hour format only This parameter can be configured from the Phone GUI also The lt Time Zone gt and lt Time Offset HH mm gt offset values are not applied to the manual time date setup Daylight Saving Time The SPA941 supports auto adjustment for daylight saving time To enable this feature the administrator must configure the lt Daylight Saving Time Rule gt regional parameter This parameter is a rule with three 3 fields Each field is separated by semicolon as shown below start lt start time gt end lt end time gt save lt save time gt where lt start time gt and lt end time gt are of the form lt month gt lt day gt lt weekday gt HH mm ss lt save time gt is of form HH mm ss lt month gt 1 2 3 12 for Jan Feb Dec lt day gt 1 2 3 31 lt weedday gt 1 2 3 7 for Mon Tue Sun or O which has special meaning HH hour 0 23 mm minute 0 59 ss second 0 59 lt start time gt and lt end time gt specify the start and end time date of daylight saving time and lt save time gt is the amount of hour min sec to add to the current time during daylight saving period The lt save time gt value can be preceded by a negative sign if subtraction is desired instead of addition If lt weekday gt is 0 it means the date to start or end daylight saving is at exactly the given date In that case the lt day gt value mu
93. iate a three way conference on the SPA941 e During an active call if the phone has one or more idle call appearances press the conf soft key This places the active call on hold and selects the next available call appearance for dialing the dial tone is heard in the active audio device Dial the third party the party to be added to the conference call After the third party s line rings or is answered presses the conf soft key again to join the two calls into a conference e If the phone has a call is on hold while another call is either ringing or is connected the confLx soft key appears If the phone has only one call on hold pressing the confLx soft key joins the call on hold with the active call If the phone has more than one call on hold pressing the confLx soft key places the active call on hold and prompts the user to select the other Line Key to join with the conference call Once the conference starts the SPA941 plays a special brief tone to all three parties to indicate that a conference call is in progress The initiator of the conference call can terminate it at any time by hanging up The initiator leave the conference without ending the call by pressing join soft key this allows the other two parties to continue the call This is implemented as a call transfer which may not work if the other two parties are hosted by different service providers Note If the extension is configured with an external Conference Bridge URL
94. ic Payload G729b dynamic payload type Uns8 99 Notes 1 Valid range is 96 127 2 The configured dynamic payloads are used for outbound calls only where the SPA presents the SDP offer For inbound calls with a SDP offer SPA will follow the caller s dynamic payload type assignments SDP Audio Codec Names Parameter Name Description Type Default NSE CODEC NAME NSE Codec name used in SDP Str31 NSE 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 56 LINKSYS A Division of Cisco Systems Inc AVT Codec Name AVT Codec name used in SDP Str31 telephone event G711a Codec Name G711a Codec name used in SDP Str31 PCMA G711u Codec Name G711u Codec name used in SDP Str31 PCMU G726r16 Codec Name G726 16 Codec name used in SDP Str31 G726 16 G726r24 Codec Name G726 24 Codec name used in SDP Str31 G726 24 G726r32 Codec Name G726 32 Codec name used in SDP Str31 G726 32 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 1 SPA uses the configured codec names in its outbound SDP 2 SPA ignores the codec names in incoming SDP for standard payload types 0 95 3 For dynamic payload types SPA identif
95. ich Line Key to transfer the call to Notes e At e completion of call transfer operation the holding call peer will be the transferee and the connected or proceeding call peer will be the transfer target e The case where a call transfer is completed when the transfer target is ringing is called a semi attended call transfer This is different from a blind transfer which is described in the next subsection Blind Call Transfer A blind transfer allows the user to transfer a call without speaking with the receiver of the transferred call For the user the call ends as soon as the transfer target s phone number is dialed The underlying mechanism is the user s phone sends the contact information for the transfer target to the phone of the other party on the call The information is carried in a signaling message When the message is sent the user s phone is dropped from the call and the other party s phone automatically dials the transfer target To perform a blind transfer press the bxfer soft key and then dial the target number The bxfer soft key will automatically become active during a call but it may not be visible on the display If the bxfer key is not visible press the left right direction arrows on the directional rocker know until it scrolls onto the display Some older SPA phones do not have bxfer soft keys To perform a blind transfer on these older devices during the call press the xfer soft key or the Line Key of an
96. ies the codec by the configured codec names Comparison is case insensitive NAT Support Parameters Parameter Name Handle VIA received Description If set to yes the SPA will process the received parameter in the VIA header inserted by the server ina response to any one of its request Else the parameter is ignored Type Default Bool No Handle VIA rport Insert VIA received If set to yes the SPA will process the rport parameter in the VIA header inserted by the UAS in a response Else the parameter is ignored If set to yes the SPA will insert received parameter in VIA header in SIP responses if received from IP and VIA sent by IP differ Bool Bool No No Insert VIA rport If set to yes the SPA will insert rport parameter in VIA header in SIP responses if received from port and VIA sent by port differ Bool No Substitute VIA addr If set to yes the SPA will use nat mapped IP port values in VIA header Bool No Send Resp To Src Port If set to yes SPA will send a response to the source port where the corresponding request is received from Else the SPA will send the response as indicated in the VIA header by the UAC Bool No STUN Server STUN server to contact for NAT mapping discovery FQDN STUN Enable Enable the use of STUN to discover NAT mapping Bool No STUN Test Enable
97. illisecond MWI Message Waiting Indication OSI Open Switching Interval PCB Printed Circuit Board PS Provisioning Server PSQM Perceptual Speech Quality Measurement 1 5 the lower the better PSTN Public Switched Telephone Network NAT Network Address Translation OOB Out of band REQT SIP Request Message RESP SIP Response Message RSC SIP Response Status Code such as 404 302 600 RTP Real Time Protocol RTT Round Trip Time SDP Session Description Protocol SDRAM Synchronous DRAM sec seconds SIP Session Initiation Protocol SP Service Provider SPA Linksys IP Telephone SSL Secure Socket Layer TFTP Trivial File Transfer Protocol TCP Transmission Control Protocol UA User Agent uC Micro controller UDP User Datagram Protocol URL Uniform Resource Locator VM Voice Mail VMWI Visual Message Waiting Indication Indicator VQ Voice Quality WAN Wide Area Network XML Extensible Markup Language e Glossary e ACD Automatic Call Distribution A switching system designed to allocate incoming calls to certain positions or agents in the order received and to hold calls not ready to be handled often with a recorded announcement e Area Code A Three digit code used in North America to identify a specific geographic telephone location The first digit can be any number between 2 and 9 The second and third digits can be any number 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 94 LINKSYS A Div
98. ing inside calling The dial tone that is heard when the customer picks up the phone is an internal dial tone e SS7 System Signaling Number 7 Technology used by large carriers to increase the reliability and speed of transmission between switches e Switch Switching Equipment that connects and routes calls and provides other interim functions such as least cost routing IVR and voicemail It performs the traffic cop function of telecommunications via automated management decisions e Touchtone DTMF The tone recognized by a push button touchtone telephone e Unified Messaging Platform that lets users send receive and manage all email voice and fax messages from any telephone PC or information device e Voice Mail A system that allows storage and retrieval of voice messages through voicemail boxes 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 96
99. ing MOH to work That is the SIP 2xx response from the remote party in reply to the re INVITE from the SPA to put the call on hold must have the SDP indicate a sendrecv or recvonly attribute and the remote destination address and port must not be 0 Audio Settings Parameter Name Description Type Default Preferred Codec Select a preferred codec for all calls However the actual codec used in a call still depends on the outcome of the codec negotiation protocol G711u G711a G726 16 G726 Choice G711u 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 72 LINKSYS A Division of Cisco Systems Inc 24 G726 32 G726 40 G729a G723 Use Pref Codec Only Only use the preferred codec for all calls The call will fail if Bool No the far end does not support this codec Silence Supp Enable Enable silence suppression so that silent audio frames are Bool No not transmitted G729a Enable Enable the use of G729a codec at 8 kbps Bool Yes G723 Enable Enable the use of G723 codec at 6 3 kbps Bool Yes G726 16 Enable Enable the use of G726 codec at 16 kbps Bool Yes G726 24 Enable Enable the use of G726 codec at 24 kbps Bool Yes G726 32 Enable Enable the use of G726 codec at 32 kbps Bool Yes G726 40 Enable Enable the use of G726 codec at 40 kbps Bool Yes DTMF Tx Method Method to transm
100. int or question mark immediately following the parameter name indicates the parameter should be user read write or read only 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 Typical usage would be as in the following example Part 123 Par2 Here Par is managed by the administrator through remote provisioning and is only visible in the admin login webUI http spa ip addr admin advanced On the other hand Par2 is managed by the end user is visible in the user login webUI http spa ip addr advanced and is unaffected by periodic resyncs to the provisioning server In this way a service provider is given full control over which parameters become end user inaccessible read only or read write following provisioning of the SPA In its most basic configuration the SPA941 resyncs to the provisioning server periodically downloading the configuration profile specified as a URL in the Profile_Rule parameter Eample Profile_Rule http my local server net profile path spa941 cfg Please refer to the Linksys SPA Provisioning Guide for a more exhaustive description of the available provisioning features in the SPA941 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 83 LINKSYS A Division of Cisco Systems Inc e Prov
101. ion of Cisco Systems Inc e SK clear Clear the entire entry Note that SK lt lt SK gt gt SK erase lt and the digit key presses will automatically repeat by keep pressing the corresponding key down without releasing it If input value is wider than what the LCD display can show the SPA941 automatically scroll the input to the left so that the user can continue to enter input On the other hand the SPA941 will stop accepting further input for a give field if the maximum allowed input length has been reached for that input field For Boolean input the SPA941 will show SK y n pressing which will toggle the current input between yes and no For Option input the SPA941 will show SK option pressing which will either a change the input field to the next available choice in round robin fashion or b bring up a menu of choices that the user can select from For most settings the input value is saved once the user press SK save or SK ok before exiting the edit screen for that setting However there are a few exceptions where the user must press SK save again at the upper level when all the required changes have been made to the settings so that they are saved into the configuration at the same time These exceptions are Preferences e Call Forward e Network Speed Dialing The SPA941 has eight programmable speed dial numbers which can be set either through the GUI or through the User tab of the phone s web page To use
102. ions are encrypted The phone screen will be updated with the CID information extracted from the Mini Certificate received from the other end The call state label on the phone screen will also be prepended with a symbol such as Connected instead of Connected The second stage in setting up a secure all can be further divided into two steps Step 1 the caller sends a Caller Hello message base64 encoded and embedded in the message body of a SIP INFO request to the called party with the following information Message ID 4B S Version and flags 4B 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 22 LINKSYS A Division of Cisco Systems Inc SSRC of the encrypted stream 4B Mini Certificate 252B Upon receiving the Caller Hello the callee responds with a Callee Hello message base64 encoded and embedded in the message body of a SIP response to the caller s INFO request with similar information if the Caller Hello message is valid The caller then examines the Callee Hello and proceeds to step 2 if the message is valid In step 2 the caller sends the Caller Final message to the callee with the following information 2 Message ID 4B Encrypted Master Key 16B or 128b Encrypted Master Salt 16B or 128b With the master key and master salt encrypted with the public key from the callee s mini certificate The master key and master salt are used by bo
103. ision of Cisco Systems Inc 2008 Billing Increment The division by which the call is rounded In the field it is common to see full minute billing on the local invoice while 6 second rounding is the choice of most long distance providers that bill their customers directly Blocked Calls Caused by an insufficient network facility that does not have enough lines to allow calls to reach a given destination May also pertain to a call from an originating number that is blocked by the receiving telephone number Bundled Service Offering various services as a complete package Call Completion The point at which a dialed number is answered Call Termination The point at which a call is disconnected CDR Call Detail Records A software program attached to a VoIP telephone system that records information about the telephone number s activity Carrier s Carrier Companies that build fiber optic and microwave networks primarily selling to resellers and carriers Their main focus is on the wholesale and not the retail market Casual Access Casual Access is when customers choose not to use their primary carriers to process the long distance call being made The customer dials the carriers 101XXXX number CO Central Office Switching center for the local exchange carrier Centrex This service is offered by the LEC to the end user The feature rich Centrex line offers the same features and benefits as a PBX to a customer without the c
104. isioning Flow Provisioning a pool of SPA941 phones consists of two main steps e Configuring all devices to their common provisioning environment and e Configuring each device to their unique network attributes The first step also known as pre provisioning includes programming initial parameters such as a resync URL a resync period and an administrator password The second step provisioning proper manages the SPA941 under normal provisioned conditions configuring all pertinent parameters for normal operation unique to each device Pre Provisioning New units need to be told where to find their configuration within the customer s network This step can be automated within a local network By default each SPA941 will resync out of the box to the configuration profile spa941 cfg located in the virtual root directory of the TFTP boot server specified by the Local Area Network s DHCP parameters DHCP option 66 This shared configuration profile can redirect the SPA941 to its eventual home URL This pre provisioning step is typically performed in a controlled environment For sufficiently large volume orders Linksys can customize the pre provisioning parameters into each unit at the time of manufacture For instance pre provisioning can be as simple as configuring only the Profile Rule parameter with the final URL of the phone s profile For example Profile_Rule http prov voip one com spa941 profile MA cfg Here t
105. istance In either case an outside line tone is played after the initial 8 or 9 and neither prefix is transmitted when initiating the call lt 9 gt 1 XXX XXXXXXX lt 8 1212 gt XXXXXXX The following allows only placing international calls 011 call with an arbitrary number of digits past a required 5 digit minimum and also allows calling an international call operator 00 In addition it lengthens the default short interdigit timeout to 4 seconds S 4 00 011 xxxxx x The following allows only US style 1 area code local number but disallows area codes and local numbers starting with 0 or 1 It also allows 411 911 and operator calls 0 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 80 LINKSYS A Division of Cisco Systems Inc 0 49 11 1 2 9 xx 2 9 xxxxxx The following allows US style long distance but blocks 9xx area codes 1 2 8 xx 2 9 xxxxxx The following allows arbitrary long distance dialing but explicitly blocks the 947 area code 1 947 xxxxxxx 1 XXX XXXXXXX The following implements a Hot Line phone which automatically calls 1 212 5551234 SO lt 12125551234 gt The following provides a Warm Line to a local office operator 1000 after 5 seconds unless a 4 digit extension is dialed by the user P5 lt 1000 gt xxxx
106. it DTMF signals to the far end Inband InBand Auto Send DTMF using the audio path INFO Use the SIP AVT INFO method AVT Send DTMF as AVT events Auto INFO Use Inband or AVT or INFO based on outcome of codec Auto negotiation DTMF Process AVT If set to yes SPA will include AVT payload type in its SDP Bool Yes and will process AVT DTMF packets sent by the peer Else it will not include AVT payload type in the SDP Release Unused Codec If set to yes SPA will relinquish all the low bit rate codecs Bool Yes included in the original SDP that are not supported by the called party hence other concurrent calls can reuse these low bit rate codecs such as g729a Otherwise the SPA will not relinquish these resources until the end of the call even if the called party did not indicate support for these codecs Notes 1 A codec resource is considered as allocated if it has been included in the SDP codec list of an active call even though it eventually may not be the one chosen for the connection So if the G 729a codec is enabled and included in the codec list that resource is tied up until the end of the call whether or not the call actually uses G 729a If the G729a resource is already allocated and since only one G 729a resource is allowed per SPA no other low bit rate codec may be allocated for subsequent calls the only choices are G711a and G711u On the other hand two G 723 1 G 726 resources are available per SPA
107. l be updated with the Name and Number extracted from the Mini Certificate sent by the other party The callee should check the name and number again to ensure the identity of the caller The caller should also double check the name and number of the callee to make sure this is what he she expects Note that the SPA will not switch to secure mode if the callee s CID Number from its Mini Certificate does not agree with the user id used in making the outbound call the callers SPA will perform this check after receiving the callee s Mini Certificate 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 23 LINKSYS A Division of Cisco Systems Inc Service Provider Requirements The SPA Mini Certificate MC has a 512 bit public key used for establishing secure calls The administrator must provision each subscriber of the secure call service with an MC and the corresponding 512 bit private key The MC is signed with a 1024 bit private key of the service provider who acts as the CA of the MC The 1024 bit public key of the CA signing the MC must also be provisioned to each subscriber The CA public key is used by the SPA to verify the MC received from the other end If the MC is invalid the SPA will not switch to secure mode The MC and the 1024 bit CA public key are concatenated and base64 encoded into the single parameter lt Mini Certificate gt The 512 bit private key is base64 encoded into the lt S
108. laces dial pulses with electronically produced tones for network signaling Enhanced Service Services that are provided in addition to basic long distance and accessed by way of a touchtone phone through a series of menus Exchange Code NXX The first three digits of a phone number 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 95 LINKSYS A Division of Cisco Systems Inc e Flat rate Pricing The customer is charged one rate sometimes two rates one for peak and one for off peak rather than a mileage sensitive program rate e IC Interexchange Carrier A long distance provider that maintains its own switching equipment e IVR Interactive Voice Response Provides mechanism for information to be stored and retrieved using voice and a touchtone telephone e Local Loop The local telephone company provides the transmission facility from the customer to the telephone company s office which is engineered to carry voice and or data e North American Numbering Plan NANP How we identify telephone numbers in North America We can identify the telephone number based on their three separate components NPA NXX XXXxX e PIN Personal Identification Code A customer calling billing code for prepaid and pay as you go calling cards e Private Branch Exchange Advanced phone system commonly used by the medium to larger customer It allows the customer to perform a variety of in house rout
109. le parameter Parameter Name Description Type Default Provision Enable Master enable for configuration profile resync Bool yes operations Resync On Reset Resyncs configuration profile from configuration Bool yes server whenever the SPA resets Resync Random Delay Spread interval for resync requests Time 2 Resync Periodic Resyncs configuration profile periodically after Timed 3600 reset Resync Error Retry Delay Retry interval following resync failure Timed 3600 Forced Resync Delay Maximum time SPA will wait before initiating a TimeO 14400 resync operation if it is not idle Resync From SIP Enables resync of configuration profile from a Bool Yes SIP command Resync After Upgrade Attempt Whether or not to resync following an upgrade Bool Yes 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 52 LINKSYS A Division of Cisco Systems Inc attempt Resync Trigger 1 Scriptable resync condition Script empty Resync Trigger 2 Scriptable resync condition Script empty Resync Fails On FNF Whether _file not found response from Bool Yes provisioning server should be considered a failed resync Profile Rule Primary configuration profile URL script ProfileScript spa941 cfg Profile Rule B Secondary configuration profile URL script ProfileScri
110. licit and implicit dialing are allowed if the audio device is selected after dialing Explicit dialing before the audio device is selected is called on hook dialing When an audio device is selected before dialing a call appearance for making the call is also indirectly selected The SPA941 will pick the first available call appearance in descending order starting with Line Key One If no call appearances are available no audio devices are selected A call appearance can be explicitly selected by pressing the corresponding line key An audio device is selected automatically based on the device s programmed preferences Set by pressing menu gt Preferences gt Preferred Audio Device or by accessing the User tab of the web based interface Answering Calls A line key LED rapidly blinks red when a call is incoming The user can answer the call by e Selecting an audio device The SPA941 will answer the ringing call If there are simultaneous incoming calls the call ringing on the lowest numbered line key is answered e Pressing the corresponding Line Key The default audio device will be automatically selected Ending Calls When a call is in the active state not on hold or parked the call can be ended by turning off the currently selected audio device A call in a standby state cannot be ended it must be brought back to the active state first Hold and Resume An active call can be placed on hold by e explicitly pressin
111. lt Extension Select an extension to be used for this Line key Choices are Choice 1 1 2 3 4 for the 4 Line version and 1 2 for the 2 Line version Short Name A short label to be shown on the LCD display for the Line Key n Str15 Share Call Yes indicates Line Key nis a shared call appearance Otherwise Bool no Appearance this call appearance is not shared a k a private 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 66 LINKSYS A Division of Cisco Systems Inc Line Key LED Pattern Each parameter in this section represents the call appearance state where the specified LED pattern should apply Parameter Name Description Type Default Idle LED pattern during the Idle state where the call appearance is not is in use and is available to make a new call Leaving this entry blank indicates the default value of c g LedScript Local Seized LED pattern during the Local Seized state where this station has seized the call appearance to prepare for a new outbound call Leaving this entry blank indicates the default value of c r LedScript Local Progressing LED pattern during the Local Progressing state where this station is attempting on this call appearance an outgoing call that is in proceeding i e the called number is ringing Leaving this entry blank indicates the default value of c
112. luding retransmissions SIP Bytes Sent Total number of bytes of SIP messages sent including retransmissions SIP Bytes Received Total number of bytes of SIP messages received including retransmissions External IP External IP address used for NAT mapping Ext 1 4 Status One Sectio n Per Extension Registration State Registration state of the line Not Registered Registered or Failed Last Registration At Local time of the last successful registration Next Registration In Number of seconds before the next registration renewal Message Waiting Indicate whether new voice mails available Yes or No Mapped SIP Port NAT Mapped SIP Port Call 1 4 Status One Section Per Call each Call corresponds to a Line Key State State of the call Idle Dialing Calling Proceeding Ringing Answering Connected Hold Holding Resuming or Reorder Tone Tone playing for this call Dial 2 Dial Outside Dial Ring Back Ring Busy Reorder SITi 4 Call Waiting Call Forward Conference 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 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 Indicate whether the call is triggered
113. ly to change the short timer override use S delay value lt space gt 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 79 LINKSYS A Division of Cisco Systems Inc Pause A sequence may require an explicit pause of some duration before continuing to dial digits in order for the sequence to match The syntax for this is similar to the timer override syntax P delay value lt space gt The delay value is measured in seconds This syntax allows for the implementation of Hot Line and Warm Line services To achieve this one sequence in the plan must start with a pause with a 0 delay for a Hot Line and a non zero delay for a Warm Line Implicit sequences The SPA implicitly appends the vertical code sequences entered in the Regional parameter settings to the end of the dial plan for both line 1 and line 2 Likewise if Enable_IP_Dialing is enabled then ip dialing is also accepted on the associated line Examples The following dial plan accepts only US style 1 area code local number with no restrictions on the area code and number 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 lt 1212 gt 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 d
114. may access the Linksys voice product technical support web site under the Support tab at http www linksys com This site contains FAQs and other technical documentation and soft tools for supporting the Linksys IP Telephone and other Linksys VoIP products Please see the SPA941 User Guide for information about the configuration and settings that a typical end user will encounter The SPA941 User Guide can be found on the Linksys web site See the Support tab at http www linksys com 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 91 LINKSYS A Division of Cisco Systems Inc Service Provider Support Service providers may contact Linksys and receive direct support by sending e mail to e Care for the SPA941 Phone Do not expose the phone to heat sun cold water The SPA941 phone is an electronic device and should not be exposed to heat sun cold or water A rule of thumb is that if its too hot or cold for you to be comfortable its not a good place for the phone either Do not expose the phone to water as this will create a shock hazard Cleaning the phone To clean the phone please use a lightly moistened paper towel Do not spray or pour cleaning solution directly onto the phone 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 92 LINKSYS A Division of Cisco Systems Inc e Acronyms A D Analog To Digital
115. me is not available e phone is the call peer s phone number user id If this field is missing the peer s phone number is not available e tis the time stamp at which the call log is entered It has the format mm dd hh mm lt a p gt where a am and p pm Example 1 E2 Joe Smith 14089991234 t 10 12 11 12a Example 2 14089991234 10 12 01 12p On E1 name not available Personal Directory A directory entry consists of the following information e Name e Phone number or SIP URL e Ring Tone Up to 100 entries can be stored in the SPA941 An entry can be added or edited from the phone GUI or on the web page or through remote provisioning To do this from the web page or through remote provisioning you specify the value as a DirEntry script To view or edit the personal directory via the phone web interface use the link http lt ip address gt pdir htm The user can invoke the Directory menu on the phone in 2 ways a Press the menu button and select Directory or b Press SK dir when you re not already in a SETUP screen The user can add a new entry to the Directory by selecting the New Entry option the first item on the Directory menu and press SK add To modify or delete an existing item highlight the corresponding entry and press SK edit or SK del When trying to delete an entry the phone will prompt the user for confirmation before proceeding 2003 2005 Linksys a Division of Cisco Syste
116. ms Proprietary See Copyright Notice on Page 2 26 LINKSYS A Division of Cisco Systems Inc The user can copy an existing directory entry to create a new entry To do this high light the entry you want to copy and press SK copy Then high light the New Entry option and press SK paste The user can edit the new entry and then press SK save to save the entry The user can also save a call log entry into the directory To do this select a call log entry and press SK save Then edit the entry if needed and then press SK save to save the new entry in the directory Entering and Saving Settings The SPA941 GUI allows the user to enter alphanumeric and numeric inputs will ease The following data types may be entered into the phone e Numeric String such as a phone number when dialing Alphanumeric String such a name when adding a Directory entry IP Address such as a DNS server IP address Boolean y n such as enabling or disabling DHCP Option such as selecting a preferred audio device or a ring tone When any two or more of the numeric string alphanumeric string and IP address types can apply to an entry the phone will show a soft key for switching input modes among the applicable types The soft keys will change when pressed to allow the user to scroll through the available modes such as SK num gt SK alpha gt SK IP The cursor will also change according to the following table to remind the user which input mode is b
117. n 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 RingScript A script that describes a ring tone Syntax n ring tone name w waveform id or path c cadence id b break time t total time ring tone name is a name to identify this ring tone specification This name will appear on the Ring Tone menu of the phone The same name can be used in a SIP Alert Info header in an inbound INVITE request to tell the phone to play the corresponding ring tone specification Because of this the name should contain characters allowed in a URL only Waveform id is the index of the desired waveform to use in this ring tone specification There are 4 built in waveforms at present 1 A classic phone with mechanical bell 2 Typical phone ring 3 A classic ring tone 4 A wide band frequency sweep signal This field can also be a network path url to download a ring tone data file from a server on the fly In this case the syntax of the field is w tftp hostname port path cadence id is the index of the desired cadence to play the given waveform 8 cadences 1 8 as defined in lt Cadence 1 gt through lt Cadence 8 gt Cadence id can be 0 If w 3 4 or an url Setting c 0 implies the on time is the natural length of the ring tone file break time specifies the number of seconds to break between two bursts of ring tone such as b
118. n of the volume level Each volume has 16 steps in nonlinear logarithmic scale Each setting is a 16 bit multiplier in the range 41 to 32764 in Q3 12 format or 0 01 to 7 999 or 40 to 18 dB Each step is equal to 58 16 1 or 3 87 dB Additional volume settings can be added and accessed via the phone setup screens These volumes are e DTMF Echo Volume use lt DTMF Playback Level gt e Call Progress Tone Volume setting the ToneScript for each tone Ring Tone The SPA941 offers 10 ring tones The characteristics of each ring tone is configurable using a RingScript see Section 3 In a RingTone script the user can assign a name for the ring tone and specifies 2 components for it 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 32 LINKSYS A Division of Cisco Systems Inc e Waveform 1 2 3 or 4 e Cadence 1 2 3 or8 The user can assign a default ring tone for each extension The user can assign a ring tone for each personal directory entry which overrides the default ring tone Ring tone can be selected from the GUI In addition the SPA941 allows 2 user downloadable ring tones Therefore a total of 12 ring tone choices are available to the user to be used as default ring for an extension or assigned to individual callers in the personal directory The user downloaded ring tone as labeled as User 1 and User 2 in the choices for the lt Default Ring gt On the
119. ncreasing the number of calls supported by the network by reducing the required bi directional bandwidth for a single call VAD uses a very sophisticated algorithm to distinguish between speech and non speech signals Based upon the current and past statistics the VAD algorithm decides whether or not speech is present If the VAD algorithm decides speech is not present the silence suppression and comfort noise generation is activated This is accomplished by removing and not transmitting the natural silence that occurs in normal 2 way connection the IP bandwidth is used only when someone is speaking During the silent periods of a telephone call additional bandwidth is available for other voice calls or data traffic since the silence packets are not being transmitted across the network Configurable Dial Plan with Interdigit Timers The SPA has three configurable interdigit timers e Initial timeout T handset off hook no digit pressed yet e Long timeout L one or more digits pressed more digits needed to reach a valid number as per the dial plan 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 38 LINKSYS A Division of Cisco Systems Inc e Short timeout S current dialed number is valid but more digits would also lead to a valid number Please see Section 10 for complete Dial Plan implementation details 2003 2005 Linksys a Division of Cisco Systems Proprietary Se
120. ng the call Block Anonymous Call Reject all inbound calls with no caller ID with a 406 response Do Not Disturb DND Reject all inbound calls as if the phone is busy That is the phone either responds a 302 if call forward on busy if enabled or 486 otherwise Secure Call If enabled the SPA941 will try to negotiate on all outbound calls to use encrypted media SRTP by default The user can also enable or disable this feature on a per call basis by pre dialing a code before making the call Note that in order to use Secure Call on an extension lt Mini Certificate gt and lt SRTP Private Key gt must be configured for that extension The enabling and disabling of these services applies to calls on all configured extensions on the SPA941 Secure Call Implementation A secure call is established in two stages The first stage is no different form a normal call setup Right after the call is established in the normal way with both sides ready to stream RTP packets the second stage starts where the two parties exchange information to determine if the current call can switch over to the secure mode The information is transported by base64 encoding and embedding in the message body of SIP INFO requests and responses with a proprietary format If the second stage is successful the SPA will play a special Secure Call Indication Tone for short while to indicate to both parties that the call is secured and that RTP traffic in both direct
121. of a mapping from any external address port to the corresponding private source address port These characteristics of a NAT can be exploited by an SPA to let external entities send SIP messages and RTP packets to it when it is installed on a private network VoIP NAT Interworking 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 40 LINKSYS A Division of Cisco Systems Inc In the case of SIP the addresses where messages data should be sent to an SPA are embedded in the SIP messages sent by the device If the SPA is sitting behind a NAT the private IP address assigned to it is not usable for communications with the SIP entities outside the private network The SPA must substitute the private IP address information with the proper external IP address port in the mapping chosen by the underlying NAT to communicate with a particular public peer address port For this the SPA needs to perform the following tasks Discover the NAT mappings used to communicate with the peer This could be done with the help of some external device For example a server could be deployed on the external network such that the server will respond to a special NAT Mapping Discovery request by sending back a message to the source IP address port of the request where the message will contain the source IP address port of the original request The SPA can send such a request when it first attempts to communicate with a SIP
122. on of Cisco Systems Proprietary See Copyright Notice on Page 2 51 LINKSYS A Division of Cisco Systems Inc Manual DHCP DHCP Manual Both DNS servers supplied by DHCP server and the ones statically configured but in different order DNS Query Mode Do parallel or sequential DNS Query Choice Parallel Syslog Server Specify the Syslog server name and port This feature FQDN specifies the server for logging SPA system information and critical events Debug Server The debug server name and port This feature FQDN specifies the server for logging SPA debug information The level of detailed output depends on the debug level parameter setting Debug Level The higher the debug level the more debug Choice information will be generated Zero 0 means no debug information will be generated Primary NTP IP address or name of primary NTP server Str127 Server or IP Secondary NTP IP address or name of secondary NTP server Str127 Server or IP Notes 1 Parallel DNS query mode SPA will send the same request to all the DNS servers at the same time when doing a DNS look up the first incoming reply will be accepted by SPA 2 Tolog SIP messages Debug Level must be set to at least 2 3 If both Debug Server and Syslog Server are specified Syslog messages are also logged to the Debug Server Provisioning Parameters Provisioning operations are gated by the Provision_Enab
123. on profile see Section 21 on a complete list Note that if service is enabled but the corresponding code is emptied the service can still be enabled disabled by the end user from the phone GUI or web page If a service is disabled any soft key associated with that service will be hidden from the GUI These are SK dnd SK dnd SK cfwd and SK cfwd Any menu item associated with a disabled service will be preceded with an exclamation mart and no soft keys will be visible to select that entry e Voice and Signaling Features SIPv2 Session Initiation Protocol Version 2 RFC 3261 3265 SIP Proxy Redundancy Static or Dynamic via DNS SRV In typical commercial IP Telephony deployments all calls are established through a SIP proxy server An typical SIP proxy server may serve tens of thousands subscribers It is important that a backup server is available so that an active server can be temporarily switched out for maintenance The SPA supports the use of backup SIP proxy servers so that service disruption should be next to non existent SIP Proxy Dynamic Redundancy The dynamic nature of SIP message routing makes the use of a static list of proxy servers inadequate in some scenarios In deployments where user agents are served by different domains for instance it would not be feasible to configure one static list of proxy servers per covered domain into an SPA One solution to this situation is through the use DNS SRV records Th
124. oprietary See Copyright Notice on Page 2 53 LINKSYS A Division of Cisco Systems Inc SIP Parameters SIP Parameters Parameter Name Description Type Default Max Forward SIP Max Forward value Range 1 255 Uns8 70 Max Redirection Number of times to allow an INVITE to be Uns8 5 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 2 challenged 0 255 SIP User Agent Name User Agent Header to be used by the unit in Str63 Linksys outbound requests If empty the header is not version included SIP Server Name Server Header to used by the unit in responses to Str63 Linksys inbound responses If empty the header is not version included SIP Accept Language Accept Language Header to be used by the unit Str31 If empty the header is not included Remove Last Reg Remove last registration before registering a new Bool no one if value is different one DTMF Relay MIME This is the MIME Type to be used in a SIP INFO Str31 application dtmf relay Type message used to signal DTMF event Use Compact Header If set to yes the SPA will use compact SIP Bool no headers in outbound SIP messages If set to no the SPA will use normal SIP headers SIP Timer Values sec
125. pport personnel The Serial number is printed on the bottom of the phone Phone will not make or receive calls If the phone will not make or receive calls please execute the following steps Step 1 Please press menu button then 11 then 2 This will provide the registration information for Extension 1 of the phone The screen should display E1 Registered and other information in the format provided below There may be a problem and the screen will display Not Registered LINKSYS o Ao o V Step 2 Write down if the phone is Registered or Not Registered To exit from this menu option press the menu button Step 3 Get the Current IP Address of the phone To get the Current IP Address please press menu 9 2 The phone will display the Current IP Address and please write it down To exit from this menu option press the menu button 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 90 LINKSYS A Division of Cisco Systems Inc Step 4 Contact the network administrator or service provider Please have the Model SPA941 Serial Number telephone number Current IP Address and Registration Status Registered Not Registered ready for the support personnel The Serial number is printed on the bottom of the phone Calls with Poor Voice Quality The source of this issue often is one of two sources e The phone or call
126. pt empty Profile Rule C Secondary configuration profile URL script ProfileScript empty Profile Rule D Secondary configuration profile URL script ProfileScript empty Log Resync Request Msg Syslog message generated when attempting a ProfileMsg See resync provisioning discussion section Log Resync Success Msg Syslog message generated after a successful ProfileMsg See resync provisioning discussion section Log Resync Failure Msg Syslog message generated after a failed resync ProfileMsg See provisioning discussion section Upgrade Parameters Parameter Name Description Type Default Upgrade Enable Master enable for firmware upgrade operations Bool Yes Upgrade Error Retry Delay Retry interval following upgrade failure Timed 3600 Upgrade Rev Limit Restricts downgrades to versions greater than a RevNum empty specified minimum version Upgrade Rule Upgrade script UpgradeScript empty Log Upgrade Request Msg Syslog message generated when attempting an UpgradeMsg See upgrade provisioning discussion section Log Upgrade Success Msg Syslog message generated after a successful UpgradeMsg See upgrade provisioning discussion section Log Upgrade Failure Msg Syslog message generated after a failed upgrade UpgradeMsg See provisioning discussion section License Keys Premium features license keys The key need only String empty be entered once write only parameter 2003 2005 Linksys a Division of Cisco Systems Pr
127. s on a SPA941 In fact all four call appearances can be instances of the same extension This extension is not a Shared Line Appearance unless it is also assigned to a Line Key on another station Any of the four Line Keys can be disabled Each call appearance supports one call at a time either active or on hold VoIP Interface VI An Extension and its associated control parameters configured on a particular station The SPA941 includes a rich set of configuration parameters to control the operation when calling via the account Configuring an extension on the SPA941 includes configuring the core account information and the set of VI control parameters While the account information is usually the same for a shared extension on different stations the rest of the VI parameters can be different For example the dial plan or the preferred codec to use when making a call on this extension could be different for two different stations sharing the extension Call Appearance State CAST The state of a call appearance which can be one of the following o Disabled The Line Key is disabled Idle Ready The Call Appearance is ready for use Dialing Collecting digits from the user to be dialed out from this Line Calling Waiting for the called party to respond Proceeding a k a Progressing Called party s station is ringing Ringing a k a Alerting Incoming call station is ringing Connected Connected with remote party Held Remote party is on hold
128. something less this value then the minimum value is used Reg Max Expires Maximum registration expiration time allowed from Timed 7200 the proxy in the Min Expires header If value is larger than this then the maximum value is used Reg Retry Intvl Interval to wait before the SPA retries registration Time 30 again after encountering a failure condition during last registration Reg Retry Long When Registration fails with a SIP response code Time 1200 Interval that does no match lt Retry Reg RSC gt the SPA will wait for the delay specified in this parameter before retrying If this parameter is 0 the SPA will stop retrying This value should be much larger than lt Reg Retry Intvl gt which should not be 0 Response Status Code Handling Parameter Name Description Type Default SIT1 RSC SIP response status code to INVITE on which to RscTmplt play the SIT1 Tone SIT2 RSC SIP response status code to INVITE on which to RscTmplt play the SIT2 Tone SIT3 RSC SIP response status code to INVITE on which to RscTmplt play the SIT3 Tone SIT4 RSC SIP response status code to INVITE on which to RscTmplt play the SIT4 Tone Try Backup RSC SIP response status code on which to retry a RscTmplt backup server for the current request Retry Reg RSC Interval to wait before the SPA retries registration Timed 30 again after encountering a failure condition during last registration RTP Parameters Parameter Name Description Type Default RTP Port Min Minimum
129. st not be negative If lt weekday gt is not zero then the daylight saving starts or ends on the lt weekday gt on or after the given date if lt day gt is positive or on or before the given date if lt day gt is negative If lt day gt is 1 it means the lt weekday gt on or before the end of the month in other words the last occurrence of lt weekday gt in that month 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 30 LINKSYS A Division of Cisco Systems Inc Optional values inside are assumed to be 0 if they are not specified Midnight means 0 0 0 of the given date Example 1 Starts at midnight on 17 Sunday of April ends at midnight on the last Sunday of October add 1 hour USA N America Below are all equivalent rules Start 4 1 7 0020 end 10 31 7 00 0 save start 4 1 7 end 10 1 7 save 1 start 4 1 7 0 end 10 1 7 0 save 1 Example 2 Starts at midnight on last Sunday of April ends at midnight on the last Sunday of September add 1 hour Egypt start 4 1 5 end 9 1 4 save 1 Egypt Example 3 Starts at midnight on first Sunday of October ends at midnight on the 3rd Sunday of March add 1 hour New Zealand start 10 1 7 3 22 7 save 1 New Zealand Checking Phone Status Besides the Info group on the web page the Phone GUI has a Status menu where one can check the current status of the phone Reboot and Restart Reboot
130. stems Inc Premium Features Premium functionality in the SPA941 is enabled via license keys that can be obtained from Linksys Specifically the 4 independently configurable extension capability can be enabled with such a key This is an upgrade from the 2 independently configurable extension capability that comes with the default SPA941 The keys consist of a sequence of letters and digits organized in groups of 4 separated by the character Each key is unique to a specific phone To enable the premium feature in a phone program the key supplied for that phone into the License_Keys parameter either via the phone s webuUl or via remote provisioning This operation need only be performed once Thereafter the premium feature is enabled on that phone The License_Keys parameter is write only and will revert to the empty state after every submit or resync operation If the key is programmed via a remote profile resync it is advisable that the License_Keys parameter be emptied in the profile stored on the server once the key has been installed in the phone To obtain premium feature license keys for the SPA941 please contact Linksys sales linksys com e Functional URLs for Upgrades Reboot and Resync The web interface of the SPA supports several functions through special URLs Upgrade Reboot and Resync Administrator privilege is needed for these functions Upgrade URL Through upgrade URL the user can upgrade the SPA
131. sys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 75 LINKSYS A Division of Cisco Systems Inc Serial Number Serial number of this device Software Version Firmware version running on 941 Hardware Version Hardware version of this device MAC Address MAC address Client Certificate Installed Not Installed Phone Status Current Time Current time and date E g 10 3 2003 16 43 00 Elapsed Time Total time elapsed since last reboot E g 25 days and 18 12 36 Broadcast Pkts Sent Total number of broadcast packets sent Broadcast Pkts Recv Total number of broadcast packets received Broadcast Bytes Sent Total number of broadcast bytes sent Broadcast Bytes Recv Total number of broadcast bytes received and processed Broadcast Packets Dropped Total number of broadcast packets received but not processed Broadcast Bytes Dropped Total number of broadcast bytes received but not processed RTP Packets Sent Total number of RTP packets sent including redundant packets RTP Packets Recv RTP Bytes Sent Total number of RTP packets received including redundant packets Total number of RTP bytes sent RTP Bytes Received Total number of RTP bytes received SIP Messages Sent Total number of SIP messages sent including retransmissions SIP Messages Recv Total number of SIP messages received inc
132. t_Timer is used after any one digit if at least one matching sequence is complete as dialed but more dialed digits would match other as yet incomplete sequences 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 61 LINKSYS A Division of Cisco Systems Inc Vertical Service Activation Codes Parameter Name Description Type Default Call Return Code Call the last caller ActCode 69 Blind Transfer Code Blind transfer current call to the target specified ActCode 98 after the activation code Cfwd All Act Code Forward all calls to the target specified after the ActCode 72 activation code Cfwd All Deact Code Cancel call forward all ActCode 73 Cfwd Busy Act Code Forward busy calls to the target specified after ActCode 90 the activation code Cfwd Busy Deact Code Cancel call forward busy ActCode 91 Cfwd No Ans Act Code Forward no answer calls to the target specified ActCode 92 after the activation code Cfwd No Ans Deact Code Cancel call forward no answer ActCode 93 Call Back Act Code Callback when the last outbound call is not busy ActCode 66 Call Back Deact Code Cancel callback ActCode 86 Block CID Act Code Block CID on all outbound calls ActCode 67 Block CID Deact Code Unblock CID on all outbound calls ActCode 66 Block CID Per Call Act Code Block CID on the next ou
133. tbound call ActCode 81 Block CID Per Call Deact Code Unblock CID on the next inbound call ActCode 82 Block ANC Act Code Block all anonymous calls ActCode wed Block ANC Deact Code Unblock all anonymous calls ActCode 87 DND Act Code Enable Do Not Disturb ActCode 78 DND Deact Code Disable Do Not Disturb ActCode 79 Secure All Call Act Code Make all outbound calls secure ActCode 16 Secure No Call Act Code Make all outbound calls not secure ActCode 17 Secure One Call Act Code Make the next outbound call secure This ActCode 18 operation is redundant if all outbound calls are secure by default Secure One Call Deact Code Make the next outbound call not secure This ActCode 19 operation is redundant if all outbound calls are not secure by default Referral Services Codes One or more code can be configured into this Str79 parameter such as 98 or 97 98 123 etc Max total length is 79 chars This parameter applies when the user places the current call on hold by Hook Flash and is listening to 2nd dial tone Each code and the following valid target number according to current dial plan entered on the 2nd dial tone triggers the SPA to perform a blind transfer to a target number that is prepended by the service code For example after the user dials 98 the SPA plays a special dial tone called the Prompt Tone while waiting for the user the enter a target number which is checked according to dial plan as in norm
134. tes optional components and n 1 2 3 4 Parameter names are case sensitive Parameter values may be case sensitive or case insensitive depending on the parameter Examples are Proxy 1 RTP_Packet_Size DND_Act_Code When a parameter name is displayed on the web page the underscore is replaced by a single space and the n component if exists will be hidden Examples are Proxy RTP Packet Size DND Act Code Notations e var in italics represents a variable or value to be substituted by the actual value it represents var can appear standalone or anywhere within a longer string e alb c represents choices The value can be one of the given choices e optional value represents an optional value e lt Par Name gt represents a configuration parameter name such as lt Proxy gt In a profile the corresponding tag is formed by replacing the space with an underscore _ such as lt RTP Packet Size gt is replaced with RTP_Packet_Size Some parameters carry an implicit index value if they apply to multiple extensions such as lt Proxy gt which exists for Extensions 1 2 3 and 4 Ina profile a n must be appended to the parameter to make it unambiguous In this example n 1 2 3 4 for Proxy 1 Proxy 2 Proxy 3 and Proxy 4 respectively e Anempty default value field implies an empty string Notes The SPA shall continue to use the last configured values for tags that are not present in a given profile
135. th ends for the derivation of session keys for encrypting subsequent RTP packets The callee then responds with a Callee Final message which is an empty message A Mini Certificate contains the following information User Name 32B S User ID or Phone Number 16B s Expiration Date 12B Public Key 512b or 64B Signature 1024b or 512B The signing agent is implicit and must be the same for all SPA s that intended to communicate securely with each other The public key of the signing agent is pre configured into the SPA s by the administrator and will be used by the SPA to verify the Mini Certificate of its peer The Mini Certificate is valid if a it has not expired and b its signature checks out User Interface The SPA can be set up such that all outbound calls are secure calls by default or not secure by default If outbound calls are secure by default user has the option to disable security when making the next call by dialing 19 before dialing the target number If outbound calls are not secure by default user has the option to make the next outbound call secure by dialing 18 before dialing the target number On the other hand user cannot force inbound calls to be secure or not secure it is at the mercy of the caller whether he she enables security or not for that call If the call successfully switches to the secure mode both parties will hear the Secure Call Indication Tone for a short while and the CID wil
136. the GUI to configure the speed dial select the Speed Dial entry after pressing menu key The speed dial numbers are assigned to the digits two through nine If a slot is not configured it will show lt Not Assigned gt on the entry Highlight the speed dial entry you want to add or modify press SK edit to make the changes and press SK ok to store the value A speed dial entry can be a phone number or URL The user can also enter a name that matches one of the directory entries As you enter the value the SPA941 will attempt to match the entry to the directory entries and brings up a list of potential matches on the screen User can scroll down to select one of the suggested entries To dial a speed dial dial the corresponding single digit like a 1 digit phone number The user can enter after the speed dial number but it is not necessary 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 28 LINKSYS A Division of Cisco Systems Inc Caller and Called Name Matching When making an outgoing call SPA941 will try to find the dialed number in the personal directory first and then from the missed call log and finally from the answered call log If a match is found and the name field is present in the matched entry it will be shown on the call screen as the called peer s name For incoming calls SPA941 will try to find the caller s phone number in the personal directory If a m
137. then conference is not limited to three way it can be as many parties as the conference bridge supports There are two restrictions that apply when using an external conference bridge 1 The conference can only be initiated by pressing the conf soft key when the third party has answered instead of when the phone is either ringing or answered and 2 Ringing call cannot be brought into a conference by pressing the confLx soft key only calls that are on hold can be brought into the conference with the confLx soft key Attended Call Transfer An attended call transfer allows a user to optionally transfer a call to a third party after having a discussion with that third party There are two ways to perform an attended call transfer e During an active call press the xfer soft key to place the current call on hold and to activate an idle Line Key Dialed the transfer target number on the newly activated line When the target is either ringing or answered press the xfer soft key again to complete the transfer e While one or more call appearances is on hold initiate a call on an idle Line Key While the call is either ringing or is connected press the xferLx line key If only one other call is on hold the active 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 18 LINKSYS A Division of Cisco Systems Inc call is connected with the call on hold Otherwise the user is prompted to select wh
138. to a firmware specified by the URL Note If the value of upgrade enable parameter in Provisioning tab is no the user cannot upgrade the SPA even if the web page tells you that the upgrade will be done when it is not in use See to get more information on firmware upgrade The syntax of Upgrade URL is http lt spa ip addr gt upgrade protocol server name port firmware pathname or 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 The firmware pathname is typically the file name of the SPA binary located in the root directory of the TFTP server If no firmware pathname is specified spa bin is assumed e For example http 192 168 2 217 upgrade tftp 192 168 2 251 spa bin Resync URL Through Resync URL the user can force the SPA to do a resync to a profile specified in the URL 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 86 LINKSYS A Division of Cisco Systems Inc Note The SPA will resync only when it is idle The syntax of Resync URL is http lt spa ip addr gt resync protocol server name port profile pathname e If no parameter follows resync the profile rule setting in provisioning is used e f no protocol
139. ues of the configuration file Provisioning refers to preparing the device with the correct configuration and firmware files for the network implementation e Web Interface Administration and Security The SPA provides a built in web server Configuration and administration can be performed through this convenient web interface 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 42 LINKSYS A Division of Cisco Systems Inc Web Interface Conventions The SPA uses the following conventions with the web administration capabilities e The SPA web administration supports two privilege levels Administrator and User To use the User privilege simply point a web browser at the IP address of the SPA to use the administrator privilege use URL http lt IP_ Address Of SPA gt admin e Version 1 0 of the SPA supports Internet Explorer 5 5 and above and Netscape 7 0 and above e The web configuration pages can be password protected See 0 for more information about password protect The user name of web Administrator is admin The user name of web User is user Note The user names for both administrator and User are fixed and cannot be changed After making changes to SPA configuration parameters pressing Submit All Changes button will apply all the changes and if necessary automatically reboot the device Multiple changes may be made on multiple page tabs of the web interface
140. und call MOH is disabled if this parameter is not specified empty Message Waiting Default Ring Conference Bridge URL SIP Settings Parameter Name Description Type Default SIP Port SIP message listening port and transmission port Port 5060 Ext SIP Port External port to substitute for the actual SIP port of the Port 0 unit in all outgoing SIP messages If 0 is specified no SIP port substitution is performed SIP 100REL Enable Enable the support or the 100rel SIP extension for Bool No reliable transmission of provisional responses 18x and the use of PRACK requests SIP Proxy Require Auth Resync Reboot SIP Remote Party ID SIP Debug Option None 1 line full exclude OPTIONS exclude Choice REGISTER exclude NOTIFY Proxy and Registration Parameter Name Description Type Default Proxy SIP Proxy Server for all outbound requests FQDN Use Outbound Proxy Enable the use of lt Outbound Proxy gt If set to no Bool No lt Outbound Proxy gt and lt Use OB Proxy in Dialog is ignored Outbound Proxy SIP Outbound Proxy Server where all outbound requests FQDN No are sent as the first hop Use OB Proxy In Dialog Whether to forcer SIP requests to be sent to the Bool Yes outbound proxy within a dialog Ignored if lt Use Outbound Proxy gt is no or lt Outbound Proxy gt is empty
141. will make this codec the only codec that ActCode 0272632 can be used for the associated call Prefer G726r40 Code Dialing code will make this codec the preferred ActCode 0172640 codec for the associated call Force G726r40 Code Dialing code will make this codec the only codec that ActCode 0272640 can be used for the associated call Prefer G729a Code Dialing code will make this codec the preferred ActCode 01729 codec for the associated call Force G729a Code Dialing code will make this codec the only codec that ActCode 02729 can be used for the associated call Notes 1 These codes automatically appended to the dial plan It is therefore not necessary to explicitly include them in dial plan Miscellaneous Parameters Default Description Type Setting the local date year is optional andcanbe Str10 Parameter Name Set Local Date mm dd yyyy 2 digit or 4 digit Set Local Time Setting the local time second is optional Str8 HH mm ss 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 65 LINKSYS A Division of Cisco Systems Inc Time Zone Number of hours to add to GMT to form local time Choice GMT 07 00 for caller id generation Choices GMT 12 00 GMT 11 00 GMT GMT 01 00 GMT 02 00 GMT 13 00 Time Offset HH mm ss Additional time
142. xel based display and a graphical user interface that provides access to the phone s functions and features Figure 1 1 shows an illustration of the phone 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 10 LINKSYS A Division of Cisco Systems Inc Figure 1 1 SPA941 Linksys IP Phone SPA941 Hardware Features e Pixel Based Display 128x64 Monochrome Graphical Liquid Crystal Display LCD e Four Illuminated Call Appearance Line Buttons with Tricolor LEDs o LED Indicates Line State Active Idle On Hold Unregistered o Line LED Configurable to 13 Different States On Off Color Flash e Dedicated Illuminated Buttons for o Audio Mute On Off o Headset On Off o Speakerphone On Off e Four Soft Key Buttons e Four Way Rocking Direction Knob for Menu Navigation 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 LINKSYS A Division of Cisco Systems Inc Voice Mail Message Waiting Indicator Light Voice Mail Message Retrieval Button Dedicated Call Hold Button Menu Settings Button for Access to Feature Set up and Configuration Menus Volume Control Up Down Rocking Knob Controls Handset Headset Speaker Ringer Standard 12 Button Dialing Pad High Quality Handset and Cradle Built In High Quality Microphone and Speaker Headset Jack 2 5 millimeter Ethernet LAN 10BaseT RJ 45 Five volt DC Universal 100 240 Volt Swit
143. y codec supports uncompressed 64 kbps digitized voice transmission at one through ten 5 ms voice frames per packet This codec provides the highest voice quality and uses the most bandwidth of any of the available codecs G 726 This low complexity codec supports compressed 16 24 32 and 40 kbps digitized voice transmission at one through ten 10 ms voice frames per packet This codec provides the high voice quality G 729A The ITU G 729 voice coding algorithm is used to compress digitized speech Linksys supports G 729 G 729A is a reduced complexity version of G 729 It requires about half the processing power to code G 729 The G 729 and G 729A bit streams are compatible and interoperable but not identical G 723 1 The SPA supports the use of ITU G 723 1 audio codec at 6 4 kbps Codec Selection The administrator can select which low bit rate codec to be used for each line G711a and G711u are always enabled Dynamic Payload When no static payload value is assigned per RFC 1890 the SPA can support dynamic payloads for G 726 Adjustable Audio Frames Per Packet This feature allows the user to set the number of audio frames contained in one RTP packet Packets can be adjusted to contain from 1 10 audio frames Increasing the number of packets decreases the bandwidth utilized but it also increases delay and may affect voice quality 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Pag
144. y components given in the FreqScript shall be used in that segment if more than one frequency component is used in a segment the components are summed together Example 1 Dial Tone 350 19 440 19 10 0 1 2 Number of Frequencies 2 Frequency 1 350 Hz at 19 dBm Frequency 2 440 Hz at 19 dBm Number of Cadence Sections 1 Cadence Section 1 Section Length 10s Number of Segments 1 Segment 1 On forever with Frequencies 1 and 2 Total Tone Length 10s Example 2 Stutter Tone 350 19 440 19 2 1 1 1 2 10 0 1 2 Number of Frequencies 2 Frequency 1 350 Hz at 19 dBm Frequency 2 440 Hz at 19 dBm Number of Cadence Sections 2 Cadence Section 1 Section Length 2s Number of Segments 1 Segment 1 On 0 1s Off 0 1s with Frequencies 1 and 2 Cadence Section 2 Section Length 10s Number of Segments 1 Segment 1 On forever with Frequencies 1 and 2 Total Tone Length 12s 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 47 LINKSYS A Division of Cisco Systems Inc Example 3 SIT Tone 985 16 1428 16 1777 16 20 380 0 1 380 0 2 380 0 3 0 4 0 Number of Frequencies 3 Frequency 1 985 Hz at 16 dBm Frequency 2 1428 Hz at 16 dBm Frequency 3 1777 Hz at 16 dBm Number of Cadence Sections 1 Cadence Section 1 Section Length 20s Number of Segments 4 Segment 1 On 0 38s Off Os with Frequency 1 Segment 2 O
145. ys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 59 LINKSYS A Division of Cisco Systems Inc activated 19 2 2 2 1 2 10 0 1 2 Holding Tone Indicate to the local user that the far end has ToneScript 600 placed the call on hold 16 1 1 1 1 1 1 1 9 5 1 Conference Tone Plays to all parties when a Three way ToneScript 350 conference is in progress 16 30 1 1 1 1 9 7 1 Secure Call Indication This tone is played when a call is successfully ToneScript 397 19 507 Tone switched to secure mode It should be played 19 15 0 2 0 2 1 1 1 only for a short while lt 30s and at a reduced 2 1 2 level lt 19 dBm so that it will not interfere with the conversation Notes 1 Reorder Tone is played automatically when lt Dial Tone gt or any of its alternatives times out Distinctive Ring Patterns 8 cadence scripts can be specified to be used as building blocking to construct a ring tone in a RingScript Parameter Name Description Type Default Cadence 1 Cadence 1 script CadScript 60 2 4 Cadence 2 Cadence 2 script CadScript 60 3 2 1 2 3 4 Cadence 3 Cadence 3 script CadScript 60 8 4 8 4 Cadence 4 Cadence 4 script CadScript 60 4 2 3 2 8 4 Cadence 5 Cadence 5 script CadScript 60 4 2 3 2 8 4 Cadence 6 Cadence 6 script CadScript 60 4 2 3 2 8 4 Cadence 7 Cadence 7 script CadScript 60 4 2 3 2 8
146. ytes Dropped Total number of broadcast bytes received but not processed RTP Packets Sent Total number of RTP packets sent including redundant packets RTP Packets Received Total number of RTP packets received including redundant packets RTP Bytes Sent Total number of RTP bytes sent RTP Bytes Received Total number of RTP bytes received 2003 2005 Linksys a Division of Cisco Systems Proprietary See Copyright Notice on Page 2 87 LINKSYS A Division of Cisco Systems Inc SIP Messages Sent Total number of SIP messages sent including retransmissions SIP Messages Received Total number of SIP messages received including retransmissions SIP Bytes Sent Total number of bytes of SIP messages sent including retransmissions SIP Bytes Received Total number of bytes of SIP messages received including retransmissions External IP External IP address used for NAT mapping Line 1 2 Status Hook State State of the hook switch On or Off Registration State Registration state of the line Not Registered Registered or Failed Last Registration At Local time of the last successful registration Next Registration In Number of seconds before the next registration renewal Message Waiting Indicate whether new voice mails available Yes or No Call Back Active Indicate whether a call back request is in progress Yes or No Last Called Number The last number call

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