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Cisco Systems BTS 10200 Switch User Manual

1. Expected Call and Network Behavior This service gives the user the convenience of recalling the last incoming call to their number automatically User Action Required to Deactivate or End No user action required 5 16 Automatic Call Back Service Description User Action Required to Activate or Use This feature allows the user to place a call to the last number they tried to reach whether the call was answered unanswered or busy by dialing an activation code Pick up the receiver Listen for dial tone Drees Expected Call and Network Behavior If the number called is idle the call will ring through and complete normally If the called number is busy the user will hear a special announcement and the feature will monitor the called number for up to 30 minutes When both 2004 Linksys Proprietary See Copyright Notice on Page 2 lines are idle the user hears a special ring During the monitoring process the user can continue to originate and receive calls without affecting the Call Return on Busy request Call Return on Busy requests can be canceled by dialing the deactivation code User Action Required to Deactivate or End 5 17 Call FWD Unconditional Lift the receiver Listen for dial tone Dress Service Description All calls are immediately forwarded to the designated forwarding number The PHONE ADAPTER will not ring or provide call
2. 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 GO 2 sc 2ne 2s on oBl cB ly 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 Total Ring Length 60s FreqScript A mini script that specifics the frequency and level parameters of a tone Up to 127 characters Syntax F L F2 _2 F3 Ls F L F s Ls Fe Lell where Fi F are frequency in Hz unsigned integers only and L L are corresponding levels in dBm with up to 1 decimal places White spaces before and after the comma are allowed but not recommended Example 1 Call Waiting Tone 440 10 Number of Frequencies 1 Frequency 2 440 Hz at 10 dBm Example 2 Dial Tone 350 19 440 19 2004 Linksys Proprietary See Copyright Notice on Page 2 33 Number of Frequencies 2 lsheeciieiney il S50 ik aie ILS Cleo Frequency 2 440 Hz at 19 dBm e ToneScript A mini script that specifies the frequency level and cadence parameters
3. 2004 Linksys Proprietary See Copyright Notice on Page 2 90 ringing and call waiting tone patterns to be played when incoming calls arrive The choice of alerting pattern to use is carried in the incoming SIP INVITE message inserted by the SIP Proxy Server or other intermediate application server in the Service Providers domain User Action Required to Activate or Use Pick up the receiver Listen for dial tone Press Expected Call and Network Behavior With this service incoming calls from up to telephone numbers can be automatically identified by distinctive ringing A distinctive ringing pattern i e short long short accompanies incoming calls from the designated telephone numbers If the user is engaged in conversation and a call from one of the designated numbers arrives a distinctive call waiting tone i e short long short accompanies the incoming call Calls from other telephone numbers ring normally User Action Required to Deactivate or End 5 22 Speed Calling Up to Eight 8 Numbers or IP Addresses Service Description The PHONE ADAPTER supports user programming of up to 8 long distance local international or emergency numbers and or IP addresses for fast and easy access User Action Required to Activate or Use Pick up the receiver Listen for dial tone Drees Dial the single digit code under which the number is to be stored 2 9 Dial the co
4. 30 A 1200 Response Status Code Handling SIT1_RSC SIT2_ RSC SIT3_RSC SIT4 RSC Try_Backup_RSC Retry Reg RSC RTP Parameters RTP Port Min RTP Port Max RTP_ Packet Size Max RTP ICMP Err RTCP_Tx Interval SDP Payload Types NSE_Dynamic_Payload AVT_Dynamic_Payload G726r16_ Dynamic_Payload G726r24 Dynamic _Payload G726r40_ Dynamic _Payload wu wu wu wu wu wu 16384 16482 0 030 wou wou 700 101 og 97 96 2004 Linksys Proprietary See Copyright Notice on Page 2 104 G729b Dynamic Payload EEN NSE_Codec_Name NSE AVT_Codec_Name telephone event G711u_Codec_Name PCMU G711la_Codec_Name PCMA y G726r16 Codec Name G726 16 G726r24 Codec Name G726 24 G726r32 Codec Name G726 32 G726r40_Codec_Name G726 40 G729a_Codec_ Name G729a G729b Codec_Name G729ab G723_Codec_Name G723 NAT Support Parameters Handle VIA received No Handle VIA_rport No Insert _VIA_ received No Insert_VIA_rport INON Substitute_VIA Addr No Send Resp To Src Port No STUN_Enable No STUN_Test_Enable No STUN_Server IP EXT_IP un EXT RTP Port Min un NAT Keep Alive Intvl TILST Sp KKK Line_Enable 1 nyest Streaming Audio Server SAS SAS Enable 1 No SAS DLG Refresh _Intvl 1 30 SAS _ Inbound RTP Sink 1 vn NAT Settings NAT Mapping Enable 1 No MAT Keep Alive Enable 1
5. Undo All Changes has no effect on the PHONE ADAPTER it will only reset the values on the web page 3 3 2 Administration Privileges The PHONE ADAPTER 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 the URL for your model number as described in the previous section 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 hitp IP_Address Of PHONE ADAPTER If the password has been set for User the browser will prompt for User authentication The username for User is user and cannot be changed 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 f
6. 30 aeta I ER E EE Y 763 951 3 00 33 1 e y dey E Ae a O 3 2004 Linksys Proprietary See Copyright Notice on Page 2 111 CWT8_Cadence Distinctive Ring CWT Pattern Names Ringl Name Ring2 Name Ring3 Name Ring4 Name Ring5 Name Ring Name Ring7 Name Rings Name 2 3 SEN Bellcore ri Bellcore r2 Bellcore r3 Bell Bell Bell Bell Bell Ring and Call Waiting Tone Spec Ring Waveform Ring Frequency Ring Voltage CWT_Frequency Control Timer Values sec Hook_Flash_Timer Min Hook Flash Timer Max Callee_On_Hook_Delay Reorder Delay Call_Back_Expires Call Back Retry Intvl Call Back Delay VMWI_Refresh_ Intvl Interdigit_Long Timer Interdigit Short _Timer CPC_Delay CPC Duration Vertical Service Activation Call _Return_Code Blind Transfer Code Call Back _Act_Code Call Back Deact Code Cfwd_All Aer Code Cfwd All Deact_Code Cfwd_Busy Act Code Cfwd_Busy Deact Code Cfwd_No Ans Act Code Cfwd_No Ans Deact Code Cfwd_Last_Act_Code Cfwd_Last_Deact_Code Block Last_Act_Code Block Last Deacrt Code Accent Last Act Code Accept_Last_Deact_Code CW_Act Code CW_Deact_Code CN Der Call Act Code CN Der Call Deact_Code Block CD Aer Gode Block CD Deact_Code Block CID Per Call Aer Code Block CD Per Call Deact_Code Block ANC Act Code Block ANC Deact_Code DND_Act_Code DND_Deact_Code CID Act Code CID_Deact_Code CWCID Act Code CWCID Deact_Code lcore r4
7. 95 3 For dynamic payload types PHONE ADAPTER identifies the codec by the configured codec names Comparison is case insensitive 4 6 4 Secure Media 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 PHONE ADAPTER 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 directions are encrypted If the user has a CIDCW capable phone and CIDCW service is enabled then the CID will be updated with the information extracted from the Mini Certificate received from the other end The Name field of this CID will be prepended with a symbol 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 Version and flags 4B SSRC of the encrypted str
8. Enable Disable DHCP 101 Enter 1 to enable Requires Password Enter 0 to disable Check IP Address 110 None IVR will announce the current IP address of PHONE ADAPTER Set Static IP Address 111 Enter IP address DHCP must be Disabled using numbers on otherwise you will hear the telephone key Invalid Option if you try pad Use the to set this value star key when entering a decimal point Requires Password Check Network Mask 120 None IVR will announce the current network mask of 2004 Linksys Proprietary See Copyright Notice on Page 2 30 PHONE ADAPTER Set Network Mask 121 Enter value using DHCP must be Disabled numbers on the otherwise you will hear telephone key pad Invalid Option if you try Use the star key to set this value when entering a e decimal point Requires Password Check Static Gateway IP 130 None IVR will announce the Address current gateway IP address of PHONE ADAPTER Set Static Gateway IP 131 Enter IP address DHCP must be Disabled Address using numbers on otherwise you will hear the telephone key Invalid Option if you try pad Use the to set this value star key when e entering a decimal Requires Password point Check MAC Address 140 None IVR will announce the MAC address of PHONE ADAPTER in hex string format Check Firmware Version 150 None IVR will announce the version of the fir
9. Substitute VIA addr Use nat mapped IP port values in VIA header Bool No Send Resp To Src Port Send response to the request source port instead of the VIA sent by port 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 If enabled with lt STUN Enable gt yes and a valid lt STUN Servers the PHONE ADAPTER 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 Warning 399 Phone Adapter 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 PHONE ADAPTER 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 Bool No Ext IP External IP address to substitute for the actual IP address of the unit in all outgoing SIP messages If 0 0 0 0 is specified no IP address substitution is performed P 0 0 0 0 2004 Links
10. The SPC syntax also controls the parameter s user level access when using the built in web interface to the PHONE ADAPTER PAP2 only An optional exclamation point or question mark immediately following the parameter name indicates the parameter should be user read write or read only 2004 Linksys Proprietary See Copyright Notice on Page 2 21 respectively If neither mark is present the parameter is made inaccessible to the user from the web interface Note that this syntax has no effect on the admin level access to the parameters When using the SPC a service provider is given full control over which parameters become inaccessible read only or read write following provisioning of the PHONE ADAPTER If the parameter specification is missing entirely from the plain text file the user level access to the parameter will remain unchanged in the PHONE ADAPTER If the plain text file contains multiple occurrences of the same parameter value specification the last such occurrence overrides any earlier ones Parameter names in the plain text file must match the corresponding names appearing in the PHONE ADAPTER web interface with the following modifications e _Inter word spaces are replaced by underscores _ e g Multi_Word_Parameter e For the PHONE ADAPTER line and user specific parameters use bracketed index syntax to identify which line or user they refer to e g Line_Enable 1 and Line_Enable 2 Comments are delim
11. 51 The administrator can select a method for conveying DTMF and hookflash on a per line basis In addition the administrator can also configure the MIME type Content Type header used when conveying DTMF or hookflash in SIP INFO messages The MIME type is set once for both lines DTMF Tx Method Method to transmit DTMF signals to the far end Choice Auto Inband Send DTMF using the audio path INFO InBand Use the SIP INFO method AVT Send DTMF as AVT AVT events Auto Use Inband or AVT based on INFO outcome of codec negotiation Auto Hook Flash Tx Method Select the method to signal Hook Flash events Choice None e None do not signal hook flash events None e AVT use RFC2833 AVT event 16 AVT e INFO use SIP INFO method with the single line INFO signal hf in the message body The MIME type for this message body is taken from the lt Hook Flash MIME Type gt paramter DTMF Relay MIME This is the MIME Type to be used in a SIP Str31 application dtmf relay Type INFO message used to signal DTMF event Hook Flash MIME This is the MIME Type to be used in a SIP Str31 application hook flash Type INFO message used to signal hook flash event 4 6 2 Codec and Audio Settings The following parameters are used to enable or disable access to specific codecs echo cancellation and FAX support Parameter Name D
12. G726 24 dynamic payload type Uns8 97 G726r40 Dynamic Payload G726 40 dynamic payload type Uns8 96 G729b Dynamic 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 PHONE ADAPTER presents the SDP offer For inbound calls with a SDP offer PHONE ADAPTER will follow the caller s dynamic payload type assignments Parameter Name Description Type Default NSE Codec Name NSE Codec name used in SDP Str31 NSE AVT Codec Name AVT Codec name used in SDP Str31 telephone event G711a Codec Name G711a Codec name used in SDP Sen 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 2004 Linksys Proprietary See Copyright Notice on Page 2 53 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 PHONE ADAPTER uses the configured codec names in its outbound SDP 2 PHONE ADAPTER ignores the codec names in incoming SDP for standard payload types 0
13. Interactive Voice Response Provides mechanism for information to be stored and retrieved using voice and a touchtone telephone 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 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 XXXX PIN Personal Identification Code A customer calling billing code for prepaid and pay as you go calling cards 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 routing inside calling The dial tone that is heard when the customer picks up the phone is an internal dial tone SS7 System Signaling Number 7 Technology used by large carriers to increase the reliability and speed of transmission between switches 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 Touchtone DTMF The tone recognized by a push button touchtone telephone Unified Messaging Platform that lets users send receive and manage all email voice and fax messages from any telephone
14. PC or information device Voice Mail A system that allows storage and retrieval of voice messages through voicemail boxes 2004 Linksys Proprietary See Copyright Notice on Page 2 116 2004 Linksys Proprietary See Copyright Notice on Page 2 117
15. RFC 1321 102 7 3 3 HTTPS with Client Certificate AAA 102 7 4 Administration and Maintenance Features 102 7 4 1 Web Browser Administration and Configuration via Integral Web Gernver 102 7 4 2 Telephone Key Pad Configuration with Interactive Voice Prompts 102 7 4 3 Automated Provisioning amp Upgrade via TFTP HTTP and HTTPS eeeseeeeeeeseeeeeeesenereereeen 102 7 4 4 Periodic Notification of Upgrade Availability via NOTIFY or HTTBR 102 7 4 5 Non Intrusive In Service Upgrades A 102 7 4 6 Report Generation and Event Logging eccceecceseseeeeeeseeeeeeeeseeeeeeeeseeeesseeeseeesaeeeeeeeeeseeeseeeeeaes 102 7 4 7 Syslog and Debug Server Hecorde AA 102 8 List of all Configuration parameters cccccceeceeeeceteeeeeeeeeeaaeeeeeeee cae eeseaeeeeeeeseeeesaeeesaeeneeeeeed 102 Qe e te 113 10 GOSS EE 115 2004 Linksys Proprietary See Copyright Notice on Page 2 5 1 Introduction This guide describes basic administration and use of the Linksys Technology PHONE ADAPTER phone adapter an intelligent low density Voice over IP VoIP gateway The PHONE ADAPTER enables carrier class residential and business IP Telephony services delivered over broadband or high speed Internet connections By intelligent we mean the PHONE ADAPTER maintains the states of all the calls it terminates It is capable of making proper decisions in reaction to user input events such as on off hook or hook flash with little or no involvement by
16. Ring6 Cadence Ring Cadence Ring8_ Cadence 1 options 1 2 3 4 5 6 7 8 1 options 1 2 3 4 5 6 7 8 8 HR options 1 2 3 4 5 6 7 8 none 1 7 options 1 2 3 4 5 6 7 8 wou H wou p 5 A New VM Available options New VM VM Arrives UI No UI A 350 19 440 19 10 0 1 2 420 19 520 19 10 0 1 2 420 16 10 0 1 520 19 620 19 10 0 1 2 480 19 620 19 10 5 5 1 2 480 19 620 19 10 25 25 1 2 480 10 620 0 10 125 125 1 2 440 19 480 19 2 4 1 2 600 16 1 25 25 1 985 16 1428 16 1777 16 380 0 3 0 4 0 914 16 1371 16 1777 16 380 0 3 0 4 0 914 16 1371 16 1777 16 380 0 3 0 4 0 985 16 1371 16 1777 16 380 0 3 0 4 0 350 19 440 19 2 1 1 14 2 10 350 19 440 19 2 2 2 14 2 10 1 1 0 142 0 14 2 600 19 1 1 1 1 1 1 1 9 5 1 350 19 20 1 1 1 1 9 7 1 397 19 507 19 15 0 2 0 2 1 1 1 2 1 2 Distinctive Call Waiting Tone Patterns CWT1_ Cadence CWT2_ Cadence CWT3_ Cadence CWT4_ Cadence CWT5_ Cadence CWT6_ Cadence CWT7_Cadence 60 2 4 60 3 2 1 2 3 4 60 8 4 8 4 60 4 2 3 2 8 4 It 327 5272 7229 E E 60 2 24 227 lt 4 lt 274 3 60 4 2 4 2 4 4 60 0 25 9 75 30 3 9 7 get ee bh EECH 30 1 1 3 1 1 9 3 VE e RE DE De sd ad 95 3 30 3 1 1 1 3 9 1
17. SO lt 12125591234 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 2004 Linksys Proprietary See Copyright Notice on Page 2 65 PS lt 71000 gt zess Explanation of Default Dial Plan The Default Dial Plan script for each line is xx 3469 1 1 0 00 2 9 xxxxxx 1 xxx 2 9 XxxXXX XXXXXXXXXXXX Dial Plan Entry Functionality XX 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 IP Dialing If IP dialing is enabled one can dial user id a b c d port where and 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 message however is still sent to the outbound proxy if it is enabled 4 8 1 Speed Dialing Settings If assigned Speed Dials enable a user to dial a single digit from 2 through 9 and then the character to dial th
18. key key string key string alias opt password quoted pass phras ett val hex string ee Gou V APOST vall wak weeimnoc Wo II esoe Ye pome Wa ea IA S mmer bel Wesco Vinci Vbeteeeil server ip4quad fqdn Attribute can contain the name of any configuration parameter If the server and scheme are unspecified the TFTP server name provided by the LAN s DHCP server is used instead Also an FQDN with multiple DNS entries is multiply resolved by the PHONE ADAPTER The variables available for macro substitution with example values are as follows PN PAP2 Product Name PSN PAP2 Product Series Number MA 000f66aaa010 MAC Address MAU OO0OF66AAA010 MAC Address upper case MAC 00 0f 66 aa a0 10 MAC Addr with Colon separators SN CH500D600862 Serial Number SWVER 1 052 Firmware Version Number HWVER Oca Hardware Version Number UPGCOND IPOS st 8 Upgrade Condition SCHEME ECS Access Scheme SERV http example com Server Name SERVIP LOA 2s 3200 Server IP Address PORT 69 TCP IP Request Port PATH guest pap2 cfg File path Ie LOZ LOS I LOZ IP address of the PHONE ADAPTER EXTIP 45 73 21 44 Configured or discovered external IP address for example using STUN PRVST 0 Error Code of Last Profile Rule EES ECK 0 Error Code of Last Upgrade Rule ERR corrupt file Error Info message EO 12 some value ernenne
19. lcore r5 lcore r6 lcore r7 lcore r8 Sinusoid 25 70 H 440 10 In 9 wou wow 1800 30 W 25 30 10 gn 2 wou Codes 69 9Q8 66 R86 k72 k73 90 91 kor k93 63 83 6O 80 x64 KEE 456 k57 71 70 NEKTAR A 68 g7 82 k77 Q7 x78 KAREN e65 g5 k25 k45 options 2004 Linksys Proprietary See Copyright Notice on Page 2 Sinusoid Trapezoid Dier Ring Act Code t26 Dist_Ring_Deact_Code MPA Gl 2 Speed Dial Act Code TETJANA Secure All Call Act Code NETET z Secure_No Call_Act_Code DEETAN Se Secure One Call Act Code WHET Qe G Secure One Call Deact Code eI19 Referral Services Codes ZP Feature Dial Services Codes nn Outbound Call Codec Selection Codes Prefer _G711u_Code 017110 Force _G711lu_Code 027110 Prefer G71la_Code Iert Force _G71la_Code V O27LILM Prefer G723 Code 01723 Force_G723 Code 02723 Prefer G726r16_ Code 0172616 Force _G726r16 Code 0272616 Prefer G726r24 Code 0172624 Force _G726r24 Code 0272624 Prefer G726r32_ Code 0172632 Force _G726r32_ Code 0272632 Prefer G726r40_ Code 0172640 Force _G726r40_ Code 0272640 Prefer G729a_Code 01729 Force _G729a_ Code 02729 Miscellaneous Get Local Date_ mm dd VR Get Local Time DH mml P Se Time Zone GMT 07 00 option
20. s line is busy because of the following Primary line already in a call primary and secondary line in a call or conference Call FWD No Answer Calls are forwarded to the designated forwarding number after a configurable time period elapses while the PHONE ADAPTER is ringing and does not answer 1 1 4 2 12 Anonymous Call Blocking By setting the corresponding configuration parameter on the PHONE ADAPTER the subscriber has the option to block incoming calls that do not reveal the caller s Caller ID 1 1 4 2 13 Distinctive Priority Ringing The PHONE ADAPTER supports a number of ringing and call waiting tone patterns to be played when incoming calls arrive The choice of alerting pattern to use is carried in the incoming SIP INVITE message inserted by the SIP Proxy Server or other intermediate application server in the Service Provider s domain 1 1 4 2 14 Speed Dialing The PHONE ADAPTER supports speed dialing of up to eight 8 phone numbers or IP addresses To enter a telephone number speed dial using a touch tone telephone the user dials a feature code 74 followed by a number 2 9 then the destination speed dialed target number When the user wishes to speed dial a target number they press the corresponding speed dial assigned number followed by the pound key Users may also enter review speed dials from User1 User2 web pages This interface or similar is required to enter IP address targets 1 1 4 3 P
21. which may include a firmware upgrade or configuration modification to the PHONE ADAPTER Application Servers Application servers are used to provide value added services such as call forwarding outgoing or incoming call blocking Voice Mail Servers Specialized servers provide voice mail services to the IP Telephony service subscribers When the subscriber is busy or the PHONE ADAPTER is out of service for maintenance or other reason incoming calls to the subscriber may be redirected to the voice mail servers where the caller can leave a voice mail The voice mail server will then notify the subscriber s PHONE ADAPTER of the availability of voice mail s in his mailbox The subscriber can then contact the voice mail server to retrieve his voice mail s The PHONE ADAPTER can indicate the message waiting status to the subscriber through a number of methods such as stuttered dial tone heard through the telephone every time the subscriber lifts up the handset until the voice mail is retrieved 1 1 2 Provisioning Overview The PHONE ADAPTER is configurable in many ways such that it can provide a wide range of customizable services and operate in many diverse environments with a variety different vendors SIP Proxy Servers VoIP Gateways Voice Mail Servers NAT applications etc Provisioning is the process by which the PHONE ADAPTER obtains a set of configuration parameters in order for it to operate in the Service Providers network The comple
22. 255 s SAS Inbound RTP Sink The purpose of this parameter is to work around devices that do not play inbound RTP if the SAS line declares itself as a sendonly device and tells the client not to stream out audio This parameter is a FQDN or IP address of a RTP sink to be used by the PHONE ADAPTER SAS line in the SDP of its 200 response to inbound INVITE from a client It will appear in the c line and the port number and if specified in the m line of the SDP If this value is not specified or equal to 0 then c 0 0 0 0 and a sendonly will be used in the SDP to tell the SAS client to not to send any RTP to this SAS line If a non zero value is specified then a sendrecv and the SAS client will stream audio to the given address Special case If the value is IP then the Str63 2004 Linksys Proprietary See Copyright Notice on Page 2 73 SAS line s own IP address is used in the c line and a sendrecv In that case the SAS client will stream RTP packets to the SAS line The default value is empty SIP Debug Option None 1 line full exclude OPTIONS exclude Choice none REGISTER exclude NOTIFY Network Jitter Level 4 settings are available very high high medium Choice High low This parameter affects how jitter buffer size is adjusted in the PHONE ADAPTER Jitter buffer size is adjusted dynamically The minimum jitter buffer size is 30 ms or 10 ms current
23. 4 Ring3 Cadence Cadence script for distinctive ring 3 CadScript 60 8 4 8 4 Ring4 Cadence Cadence script for distinctive ring 4 CadScript 60 4 2 3 2 8 4 Ring5 Cadence Cadence script for distinctive ring 5 CadScript 60 4 2 3 2 8 4 Ring6 Cadence Cadence script for distinctive ring 6 CadScript 60 4 2 3 2 8 4 Ring7 Cadence Cadence script for distinctive ring 7 CadScript 60 4 2 3 2 8 4 Ring8 Cadence Cadence script for distinctive ring 8 CadScript 60 4 2 3 2 8 4 CWT 1 Cadence Cadence script for distinctive CWT CadScript 30 3 9 7 Call Waiting Tone 1 CWT2 Cadence Cadence script for distinctive CWT 2 CadScript 30 1 1 1 9 7 CWT3 Cadence Cadence script for distinctive CWT 3 CadScript 30 1 1 1 1 1 9 5 CWT4 Cadence Cadence script for distinctive CWT 4 CadScript 30 1 1 3 1 1 9 3 CWT5 Cadence Cadence script for distinctive CWT 5 CadScript 30 3 1 1 1 3 9 1 CWT6 Cadence Cadence script for distinctive CWT 6 CadScript 30 1 1 3 1 1 9 3 CWT7 Cadence Cadence script for distinctive CWT 7 CadScript 30 1 1 3 1 1 9 3 2004 Linksys Proprietary See Copyright Notice on Page 2 68 CWT8 Cadence Cadence script for distinctive CWT 8 CadScript 2 3 3 2 Ring Waveform Waveform for the ringing signal Sinusoid Sinusoid Trapezoid Ring Frequency Frequency of the ringing signal Valid values Uns8 25 are 10 100 Hz Ring Voltage Rin
24. 4 8 Dial Plan Config ratio EEN 61 4 8 1 Speed Dialing Gettnge iernii a ae dani eed 66 2004 Linksys Proprietary See Copyright Notice on Page 2 3 4 9 Progress Tone and Ring Configuration cccccceeeeeseeeceeeeeeeeeeeseaeeeeeeeeseeeeeseaeeesaeeeeaeeee 67 4 9 1 Distinctive Ring and Other Ring Settings A 67 4 9 2 Progress Jones use hina eid ee Re Le td a 69 4 10 Less Frequently Used ParamterS c cccccccceeeeeeneeceeeeeceaeeseaeeeseaeeesaaeeseaeeseeeesaeeesaeeeenees 70 4 10 1 Advanced Protocol Parameters AAA 70 4 10 2 Additional User Account Information ec ceeeesceeeeneeeeeenneeeseeneeeeeeaeeeeeeaaeeeeenaaeeesenaeeeeseeeeeeags 73 4 10 3 Per Line Polarity Settings s c csccscotessceeasccecgcceesacs e ataei daa aep akea antenei aeaa Ti eadera aoi edadia 75 4 10 4 Additional Timer Values S C ccccesenceceseeeeeeeneeeeeeseneeeeeseeeeseegeeeeesseesesenensenseseeeeeesensenenaes 75 4 10 5 Miscellaneous Parameters cccceseceeeeseeceeeseeneeeeseeeeeeeeeeeeeeeseesesceseeeseeeesesensesenseeeeseneeeeees 76 So Expected Feature Behaviors thai nda eivetdad tiv hia dan iene eatery 79 5 1 Originatingza Phone Call vec atest et Ade ea alee ek eee eth nh een Aes 79 5 2 Receiving a Phone Call 79 5 3 Callen ID o sni a a a idnt2 vias eee nnd dees Wa Rebeca ti Reba veda ad Rebuaas teed even vented 80 5 4 Calling Line Identification Presentation CUID 80 5 5 Calling Line Identifica
25. 9 To obtain this list for another version of software run the profile compiler utility Spc KKK Linksys PHONE ADAPTER Series Configuration Parameters kkk System Configuration Restricted_Access_Domains WA Enable Web Server Yes Web Server Port BOM 3 2004 Linksys Proprietary See Copyright Notice on Page 2 102 Enable Web Admin Access Yes Admin_Passwd nn User Password foun Internet Connection Type DHCP Yes Static IP E 2 oo NetMask en Gateway elt Se Optional Network Configuration HostName oft Domain Primary DNS Secondary DNS i DNS_Server Order Manual options l k TI nun b D nun D wu Manual Manual DHCP DHCP Manual DNS Query Mode Parallel options Parallel Sequential Syslog Server d Debug Server D cf Debug Level o options 0 1 2 3 Primary NTP Server dE Secondary NTP Gerver An Configuration Profile Provision Enable Yes Resync_On_Reset Yes Resync_Random_ Delay 2 Resync_Periodic 3600 Resync_Error_ Retry Delay 3600 Forced Resync Delay 14400 Resync_From_SIP Yes Resync_ After Upgrade Attempt Yes Resync_ Trigger 1 d Resync_ Trigger 2 D i Resync_ Fails On_FNF No Profile Rule init cfg Profile Rule B mn Profile Rule C ma Profile Rule D mi g Log_Resync_Request_Msg SPN MAC Requesting resync SSCHEME SSERVIP SPORTSPATH Log Resync_ Success Msg SPN SMAC Success
26. America the PHONE ADAPTER can be configured to complete dialing to France after the country code and exactly 10 digits using 01133xxxxxxxxxxS 0 as a dial plan digit sequence When this sequence matches it overrides the short interdigit timer causing an immediate call If the S 0 had been absent the PHONE ADAPTER would wait for the short interdigit timer to expire before placing the call 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 PHONE ADAPTER 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 Maximum Length Each dial plan cannot exceed 2047 bytes after all configured vertical codes have been added to the Dial_Plan parameter Examples The following dial plan accepts only US style 1 area code local number with no restrictions on the area code and number 2004 Linksys Propr
27. Auto FAX Passthru_Method 1 NSE options None NSE ReINVITE Hook _Flash_Tx Method 1 None options None AVT INFO FAX Process NSE 1 Yes Release Unused Codec 1 Yes Dial Plan Dial Plan 1 xx 3469 11 0 00 2 9 xxxxxx 1xxx 2 9 xxxxxxS0 XXXXXXXXXXXX Enable IP Dialing 1 No FXS Port Polarity Configuration Idle Polarity 1 Forward options Forward Reverse Caller Conn _Polarity 1 Forward options Forward Reverse Callee Conn Polarity 1 Forward options Forward Reverse Call Forward Settings Cfwd_All Dest 1 d Cfwd_Busy Dest 1 Cfwd_No_ Ans Dest 1 Cfwd_No Ans Delay 1 wu wu l H 20 G Selective Call Forward Settings Cfwd_ Sell Caller 1 Cf wd_Sell1 Dest 1 wu wu Cfwd_Sel2 Caller 1 Cf wd_Sel2 Dest 1 wu wu Cf wd_Sel3 Caller 1 Cfwd_Sel3 Dest 1 wu wu Cfwd_Sel4 Caller 1 Cfwd_Sel4 Dest 1 Cfwd_Sel5 Caller 1 l Cfwd_Sel5 Dest 1 youn wu wu wu 7 wu Cfwd_Sel6 Caller 1 Cfwd_Sel6 Dest 1 Cfwd_Sel7_Caller 1 Cfwd_Sel7_ Dest 1 Cfwd_Sel8 Caller 1 Cfwd_Sel8 Dest 1 Cfwd_Last_Caller 1 Cfwd_Last_ Dest 1 Block Last _Caller 1 Accept _Last_ Caller 1 wu wu wu wu wu wu wu wu wu Speed Dial Settings Speed Dial 2 1 wo Speed Dial 3 1 Speed Dial 4 1 Speed Dial 5 1 Speed Dial 6 1
28. Copyright Notice on Page 2 49 ADAPTER Also set the Substitute_VIA_Addr and NAT_Mapping_Enable parameters Follow the instructions of the NAT software to configure static NAT mappings between the external address and ports EXT_SIP_Port EXT_RTP_Port_Min and the internal address and ports SIP_Port RTP_Port_Min Set the RTP_Port_Max parameter to a smaller number for example RTP_Port_Min plus 8 There must be mappings for the every port number between RTP_Port_Min and RTP_Port_Max when using the Manual Configuration approach Reserving 8 ports is safe since it allows both lines to have two simultaneous calls with a port for RTP and RTCP Parameter Name Description Type Default Handle_VIA_received If set to yes the PHONE ADAPTER will process the received parameter in the VIA header inserted by the server in a response to any one of its request Else the parameter is ignored Bool No Handle_VIA_rport If set to yes the PHONE ADAPTER will process the rport parameter in the VIA header inserted by the server in a response to any one of its request Else the parameter is ignored Bool No Insert VIA received Insert received parameter in VIA header in SIP responses if received from IP and VIA sent by IP differ Bool No Insert VIA rport Insert rport parameter in VIA header in SIP responses if received from port and VIA sent by port differ Bool No
29. Network via an Ethernet Connection 3 One or Two RJ 11 Phone Cable s Please observe the following steps to install the PHONE ADAPTER From the rear Side of the PHONE ADAPTER 1 Insert a standard RJ 45 Ethernet cable included into the LAN port 2 Insert the power adapter cable into the 5V power adapter cable receptacle Ensure that the power adapter jack is snugly attached to the PHONE ADAPTER 2 Insert a standard RJ 11 telephone cable into the Phone 1 port 2 Connect the other end of the cable to an analog telephone or fax machine 3 Insert a standard RJ 11 telephone cable into the Phone 2 port Optional 4 Connect the other end of the cable to an analog telephone or fax machine Note Do not connect RJ 11 telephone cable from the PHONE ADAPTER to the wall jack to prevent any chance of connection to the circuit switched Telco network You may now insert the plug end of the power adapter into a live power outlet which will power up the PHONE ADAPTER 3 Software Configuration Mechanisms The PHONE ADAPTER provides for secure remote provisioning and remote upgrade Linksys recommends that providers use a secure first time provisioning mechanism using HTTPS described in more detail in section 3 2 Subsequent provisioning is achieved through configuration profiles transferred to the device via TFTP HTTP or HTTPS These configuration profiles can be encrypted using AES 256 bit symmetric key encryption using a key configured into the de
30. No NAT Keep Alive Msg 1 SNOTIFY NAT Keep Alive Dest 1 SPROXY Network Settings SIP_TOS DiffServ_Value 1 0x68 Network Jitter Level 1 high options low medium high very high RTP_TOS DiffServ_ Value 1 0xb8 SIP Settings SIP_ Port 1 5060 SIP_100REL Enable 1 No EXT SIP Port 1 un Auth_Resync Reboot 1 Yes z SIP Debug Option 1 none options none 1 line 1 line excl OPT 1 line excl NTFY 1 line excl REG 1 line excl OPT NTFY REG full full excl OPT full excl NTFY full excl REG full excl OPT NTFY REG Call Feature Settings Blind Attn Xfer Enable 1 No MOH_Server 1 Be Xfer Wien Hanoup Conf 1 Yes Proxy and Registration 2004 Linksys Proprietary See Copyright Notice on Page 2 105 Proxy 1 Use_Outbound_ Proxy 1 Outbound Proxy 1 Use_OB Proxy In Dialog 1 Register 1 Make Call Without_Reg 1 Register Expires 1 Ans_ Call Without_Reg 1 Use_ DNS SRV 1 DNS SRV Auto Prefix 1 Proxy Fallback _Intvl 1 Voice Mail Server 1 Subscriber Information Display Name 1 User DI Password 1 Use_Auth_ID 1 Auth_ID T Mini Certificate 1 SRTP_ Private Key 1 wu No wu Yes n Ye s W W No A 3600 W No b No No z 3600 wu wu wu wu W No A wu wu wu Supplementary Service Subscription Call Waiting Serv 1 Block CID Serv 1 Block ANC Serv 1 Dist Ring Serv
31. Outbound Proxy Server where all outbound FQDN No requests are sent as the first hop Use OB Proxy In Dialog Whether to force 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 NAT Keep Alive Intvl Interval between sending NAT mapping keep alive Uns16 15 message in sec 4 6 Media and SDP Session Description Protocol Configuration 4 6 1 DTME and Hookflash By default the PHONE ADAPTER sends DTMF to the far end using RFC2833 style AVT tones This method of conveying DTMF tones sends a representation of a tone someone pressed the 7 key to the RTP peer as a separate RTP audio codec but with timing information synchronized with the speech audio codec This method of DTMF conveyance works in most topologies however in some environments the service provider may have an application server which is not in the media path or may be responsible for protocol conversion to a protocol or device which does not support AVT tones Likewise hookflash events by default are handled internally by the PHONE ADAPTER and used to trigger supplementary services which are implemented on the PHONE ADAPTER If a provider needs to convey a hookflash event to an application server to initiate a network oriented feature the PHONE ADAPTER is configurable to send these events 2004 Linksys Proprietary See Copyright Notice on Page 2
32. ParName Log_Upgrade_Success_Msg Default SPN SMAC Successful upgrade SSCHEME SSERVIP PORTSPATH SERR The Log_Upgrade_Success_Msgq is a script that defines the message sent to the configured Syslog server whenever the PHONE ADAPTER successfully completes an upgrade from the upgrade server The string supports one level of macro substitution with the same variables as for the Upgrade_Rule above An empty string does not generate a syslog message Log Upgrade Failure Message ParName Log_Upgrade_Failure_Msg Default SPN SMAC Upgrade failed SERR The Log_Upgrade_Failure_Msg is a script that defines the message sent to the configured Syslog server whenever the PHONE ADAPTER fails to complete an upgrade from the upgrade server The 2004 Linksys Proprietary See Copyright Notice on Page 2 45 string supports one level of macro substitution with the same variables as for the Upgrade_Rule above An empty string does not generate a syslog message Parameter Name Description Type Default Upgrade Enable Master enable for firmware upgrade Bool Yes operations Upgrade Error Retry interval following upgrade failure Time 3600 Retry Delay Upgrade Rule Upgrade script UpgradeScript empty Log Upgrade Syslog message generated when UpgradeMsg See Request Msg attempting an upgrade provisioning discussion section Log Upgrade Syslog message generated after a UpgradeMsg_ Se
33. The encoder and decoder pair in a compression algorithm is known as a codec The compression ratio of a codec is expressed in terms of the bit rate of the compressed speech The lower the bit rate the smaller the bandwidth required to transmit the audio packets Voice Quality is usually lower with lower bit rate however But Voice Quality is usually higher as the complexity of the codec gets higher at the same bit rate Silence Suppression The PHONE ADAPTER applies silence suppression so that silence packets are not sent to the other end in order to conserve more transmission bandwidth instead a noise level measurement can be sent periodically during silence suppressed intervals so that the other end can generate artificial comfort noise that mimics the noise at the other end using a CNG or comfort noise generator Packet Loss Audio packets are transported by UDP which does not guarantee the delivery of the packets Packets may be lost or contain errors which can lead to audio sample drop outs and distortions and lowers the perceived Voice Quality The PHONE ADAPTER applies an error concealment algorithm to alleviate the effect of packet loss Network Jitter The IP network can induce varying delay of the received packets The RTP receiver in the PHONE ADAPTER keeps a reserve of samples in order to absorb the network jitter instead of playing out all the samples as soon as they arrive This reserve is known as a jitter buffer The bigge
34. a a I eS AEN 6 1 1 The Session Initiation Protocol nnt 6 1 1 1 Components of a SIP Network eececcsessecesceeeeceeeneeneseeeeaeeneaeeeeaceneaeeseaeeesaaeseeeessaeseneeseaeseneeneaas 8 k r ProviSioning Overview eege EES EE ec 9 LI Security OVErView ege a eel cede Eae Epas uaes eine kees inat aeiee rito iaai 10 1 1 3 1 Proxy S IVers E E ae A E ees AEE E Se ed A E A ET 11 US SIP EE 11 1 1 4 1 BASIC SEIVIGES eege Eder A E 12 1 1 4 2 Enhanced Services renerien aieeaii Ne E E edie denen eee anv canes 12 1 1 4 3 PSTN Interworkingwii cet ebessen eee en chest nite ead A IS dat ee ee 14 1 2 Network Address Translation NAT Traversal 15 1 2 1 What is a NAT or NAPT Network Address Port Translator c cceccceeseeeeeeeeeeeeeeneeeeneeeeneeeeneeee 15 1 2 2 NOIP NAT Interworking ss getest a a haveltea cu deneeaee cites cedesunetectevssdecanebecteanvestec 16 1 3 Voice Quality Cvervlew cee eeeaaeseeneeeceaeeeeaaesseneeceaeeesaaesdeaeeseeeessaeeeeaaeeesaees 16 227 Hardware Meessel Eeer 17 2 1 Phone Adapter LED Giats iii eiiiieiiessiiisiiicnieiidi anii de yide dia iaa 19 2 2 Broadband Router RT31P2 LED Status c ec ccceeeeeeeeeeee eee eecaeeeeeeeeseeeesaeeeseeeeenees 19 3 Software Configuration Mechanisms c cccccceseeceeceeceeeeeceeeeeaeeseeeeesaeeseaaeseeeeeseaeeesiaeeeeeeeeaes 20 3 1 Configuration Profile Formats 0 c cccccceceeececeeeeeeseeceeeeeceaeeeeeeeeseaeeesaaeeeeaeeseeeeesaeesenaeesene
35. and sends an SSLv2 ClientHello message The configuration server then presents a server certificate signed by Linksys in a ServerHello message and requests the certificate of the client The Terminal Adapter validates the server certificate and provides its client certificate From the client certificate the provider is assured of the authenticity of the MAC address serial number and model number of the Linksys device which has connected The terminal adapter will then use an HTTP GET over this TLS secure channel to fetch its initial configuration An Apache web server can be setup to perform all the certificate verification automatically as configuration directives An example configuration is listed below lt Directory linksys secure setup gt SSLVerifyClient require SSLVerifyDepth 1 SSLRequireSSL SSLCertificateFile provider cert signed by linksys pem SSLCertificateKeyFile provider private key pem SSLCertificateChainFile linksys cert pem SSLCACertificateFile linksys cert pem SSLRequire SSL_CLIENT_VERIFY eq SUCCESS and SSL_CLIENT_I_DN_O eq Linksys and SSL_CLIENT_S_DN_O eq Linksys and SSL_CLIENT_S_DN_CN eq REQUEST_FILENAME lt Directory gt 2004 Linksys Proprietary See Copyright Notice on Page 2 25 Within this directory the Apache module mod_ssl verifies the client certificate and verifies that the MAC address in the certificate corresponds the configuration file it is requesting Either this directo
36. current call 5 6 Call Waiting Service Description The user can accept a call from a 3rd party while engaging in an active call The PHONE ADAPTER shall alert the subscriber of the 2nd incoming call by playing a call waiting tone User Action Required to Activate or Use If the you choose to answer the second call either Press and release your phone s switch hook the button you release when you take your phone off the hook or Press the flash button if your phone has one This puts your first call on hold and automatically connects you to your second call To put your second caller back on hold and return to your first caller press the switch hook or flash button again You can alternate between calls as often as you like Expected Call and Network Behavior If the user is on a call when another call comes in they will hear a series of beeps tones 2004 Linksys Proprietary See Copyright Notice on Page 2 81 alerting them to the second call The person calling will hear normal ringing User Action Required to Deactivate or End See Cancel Call Waiting 5 7 Disable or Cancel Call Waiting Service Description The PHONE ADAPTER supports disabling of call waiting permanently or on a per call basis User Action Required to Activate or Use To temporarily disable Call Waiting for the length of one call Before placing a call Lift Receiver
37. 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 e Short timeout S current dialed number is valid but more digits would also lead to a valid number 7 2 20 Message Waiting Indicator Tones MWI 7 2 21 Polarity Control The PHONE ADAPTER allows the polarity to be set when a call is connected and when a call is disconnected This feature is required to support some pay phone system and answering machines 7 2 22 Calling Party Control CPC CPC signals to the called party equipment that the calling party has hung up during a connected call by removing the voltage between the tip and ring momentarily This feature is useful for auto answer equipment which then knows when to disengage 7 2 23 International Caller ID Delivery In addition to support of the Bellcore FSK and Swedish Danish DTMF methods of Caller ID CID delivery release 2 0 adds a large subset of ETSI compliant methods to support international CID equipment The figure below shows the CID CIDCW architecture used in the PHONE ADAPTER Different flavors of CID delivery method can be obtained by mixing and matching some of the steps as shown It should be noted that the choice of CID method will affect the following features e On Hook Caller ID Associated with Ringing This type of Caller ID is used for incoming calls when the attached phone is on hook s
38. is set up in the deployment 4 7 1 Supplementary Services activated internally Once Supplementary Services on the PHONE ADAPTER are Enabled the services can be activated or deactivated dynamically by dialing specific configurable dial strings For example the default dial string to activate or deactivate most features is a character followed by a two digit code The following table lists the parameters which set these dial strings used internally by the PHONE ADAPTER If a provider wishes to offer a service which is activated or deactivated in an application server in their network instead of internally in the PHONE ADAPTER the dial pattern for that service should NOT be present in these configuration parameters Parameter Name Description Type Default Call Return Code Call the last caller ActCode 69 Blind Transfer Code Blind transfer current call to the target ActCode_ 98 specified after the activation code Cfwd All Act Code Forward all calls to the target specified ActCode 72 after the activation code Cfwd All Deact Code Cancel call forward all ActCode_ 73 Cfwd Busy Act Code Forward busy calls to the target specified ActCode 90 after 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 ActCode 92 specified after the activation code Cfwd No Ans Deact Code
39. of a call progress tone May contain up to 127 characters Syntax FreqScript Z Z2 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 Dom doff hon 2 0ffi 2 fi Lon sdoft Aha Loni ofti4 fi 4 Lonis offis fis oni s Offis fis l where fij ny noJ na n ns Nng and 1 lt nk lt 6 indicates which of the frequency 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 Eregueney M SSO Hz hae POTABm Frequency 2 440 Hz at 19 dBm Number of Cadence Sections 1 Cadence Section 1 Section Length 10 s Number of Segments 1 Segment 1 On forever with Frequencies 1 and 2 Total Tone Length 10s Example 2 Stutter Tone SEENEN REH 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 Example 3 SIT Tone SIE LA2VO 16 1777 162 1 S8O O 1 380 072 380 0 3 0 4 00 Number of Frequencies 3 Frequency 1 98
40. of passwords and pass phrases the internally generated key is 128 bits in length The following command line syntax generates a generic and unencrypted CFG file spc pap2 txt pap2 cfg A targeted CFG file with basic encryption is specified by supplying the MAC address of the target device spc target O000e08aaa010 pap2 txt pap2 cfg 2004 Linksys Proprietary See Copyright Notice on Page 2 23 An encrypted CFG file requires either a password or quoted pass phrase or a hex string The following lines illustrate command line invocations for various combinations of keys and algorithms spc rc4 ascii key apple4sale pap2 txt pap2 cfg spc aes ascii key lucky777 pap2 txt pap2 cfg spc aes ascii key my secret phrase pap2 txt pap2 cfg spc aes hex key 8d23fe7 a5c29 pap2 txt pap2 cfg A CFG file can be both targeted and key encrypted as suggested by the following example spc target 000e08aaa010 aes hex key 9a20 eb47 a txt a cfg The status messages printed by spc can be suppressed with the quiet command line option Or they can be redirected to a file with the log file_name command line option In the latter case the spc command line invocation itself is also printed in the log file preceded by a timestamp spe oaet S SPER a OORE ON Jeer 3 1 2 Encrypting and Compressing XML configuration files The Linksys PHONE ADAPTER supports en
41. status indication 1 Two 2 RJ 11 Type Analog Telephone Jack Interfaces Figure 4 above These interfaces accept standard RJ 11 telephone connectors An Analog touchtone telephone or fax machine may be connected to either interface If the service supports only one incoming line the analog telephone or fax machine should be connected to port one 1 of the RT31P2 Port one 1 is the outermost telephone port on the RT31P2 and is labeled Phone 1 2 Four 4 Ethernet 10 100 baseT three 3 for Local Network and one 1 for Internet all the 4 ports uses RJ 45 Jack Interface Figure 5 above This interface accepts a standard or crossover Ethernet cable with standard RJ 45 connector For optimum performance Linksys recommends that a Category 5 cable or greater be used in conjunction with the PHONE ADAPTER 3 LEDs 2004 Linksys Proprietary See Copyright Notice on Page 2 18 2 1 Phone Adapter LED Status LED Color s Activity Description Off Power OFF Blue On Power On Device Ready Power Blue Blue Blinking Booting System Self Test Firmware upgrade POST Power On Self Test failure not bootable Red On or Device malfunction Off No Connection on Ethernet Ethernet Blue Blue On Ethernet Connection established Blue Blinking Data Sending Receiving Off Phone is not in use not provisioned or registered Phone 1 3 SCH Phone 2 Blue Blue On Registered provisioned Blue Bli
42. take effect when the telephone is hung up and if necessary the PHONE ADAPTER will automatically reboot 2004 Linksys Proprietary See Copyright Notice on Page 2 29 3 After one minute of inactivity the unit times out The user will need to re enter the configuration menu from the beginning by pressing 4 If while entering a value like an IP address and you decide to exit without entering any changes you may do so by pressing the star key twice within a half second window of time Otherwise the entry of the star key will be treated as a dot decimal point Example To enter IP address use numbers 0 9 on the telephone key pad and use the star key to enter a decimal point To enter the following IP address value 192 168 2 215 A Use the touchtone key pad to enter 192 168 2 215 B When prompted enter 1 to save setting to configuration C Hang up the phone to cause setting to take effect or D Enter the value of the next setting category to modify 5 Hang up the phone to cause all settings to take effect PHONE ADAPTER Interactive Voice Response IVR Menu IVR Action IVR Menu Choice Parameter s Notes Enter IVR Menu None Ignore SIT or other tones E KAk until you hear Linksys configuration menu Please enter option followed by the pound key or hang up to exit Exit IVR Menu 3948 None Check DHCP 100 None IVR will announce if DHCP in enabled or disabled
43. 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 replaces 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 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 IXC Interexchange Carrier A long distance provider that maintains its own switching equipment IVR
44. to contact other provisioning URLs Each profile rule is executed only if the previous profile rule was executed successfully 2004 Linksys Proprietary See Copyright Notice on Page 2 37 These strings each supports one level of macro expansion using a small set of variables Following macro substitution the rule is evaluated to obtain the URL of the CFG file to be requested from the provisioning server The URL can be partially specified in which case default values are assumed for the unspecified terms The filepath portion of the URL must always be specified The Profile_Rule supports additional syntax that allows the URL to include conditions for example based on a function of the firmware release currently running in the PHONE ADAPTER This mechanism can aid the service provider s firmware upgrade sequence by allowing them to define different configuration profiles for different stages of an upgrade sequence The conditional syntax consists of a sequence of condition url pairs separated by the character The condition component tests the current firmware version number against a specified value If the last url in the sequence does not have an associated condition it will be attempted unconditionally The sequence of conditions is evaluated until one is satisfied The URL associated with that condition is then used to resync the PHONE ADAPTER No additional URLs in the rule are considered A profile rule which att
45. waiting when Call FWD Unconditional is activated User Action Required to Activate or Use Lift the receiver Listen for dial tone Drees Listen for dial tone and enter the telephone number you are forwarding your call to Activation will be confirmed with three short bursts of tone and your forwarding will be activated Alternatively the user can activate this feature from a web browser interface Expected Call and Network Behavior This feature allows a user the option to divert forward all calls to their telephone number to any number using the touchtone keypad of their telephone or web browser interface This service is activated or deactivated from the phone being forwarded or the web browser interface User Action Required to Deactivate or End Lift the receiver Listen for dial tone Drees You will hear a confirmation tone signaling your change has been accepted Alternatively the user can deactivate this feature from a web browser interface 2004 Linksys Proprietary See Copyright Notice on Page 2 88 5 18 Call FWD Busy Service Description Calls are forwarded to the designated forwarding number if the subscriber s line is busy because of the following Primary line already in a call primary and secondary line in a call or conference User Action Required to Activate or Use Lift the receiver Listen for dial tone Drees Listen for dial tone and en
46. 1 Cfwd_All Serv 1 Cfwd_Busy Serv 1 Cfwd_No Ans Serv 1 Cfwd_ Sel Serv 1 Cfwd_Last_Serv 1 Block Last _Serv 1 Accept Last Serv 1 DND_Serv 1 CID Serv 1 CWCID Serv 1 Call Return _Serv 1 Call Back Serv 1 Three Way Call Serv 1 Three Way Conf Serv 1 Attn Transfer Serv 1 Unattn_ Transfer Serv 1 MWI_ Serv 1 VMWI_ Serv 1 Speed Dial Serv 1 Secure Call Serv 1 Referral Serv 1 Feature Dial Serv 1 Audio Configuration Preferred Codec 1 ut Yes Yes Yes W Yes W W Yes Yes W Yes W Yes W Yes Yes Yes W W Yes H Yes W Yes W Yes W W Yes W Yes W Yes Yes W ut Yes Yes Yes Yes Yes W Yes Yes G711u G726 24 G6726 32 6726 40 G6729a G723 Silence Supp Enable 1 Use_Pref Codec Only 1 Echo Canc_Enable 1 G729a_Enable 1 Echo _Canc_Adapt_Enable 1 G723 Enable 1 Echo Supp Enable 1 G726 16 Enable 1 FAX CED Detect_Enable 1 G726 24 Enable 1 FAX CNG Detect_Enable 1 G726 32 Enable 1 FAX Passthru_Codec 1 No No W Ye s H Ye s n H W Ye s H Ye s Ye s W A n Ye s H W Ye s W S Ye s d Ye s S W Ye s F G711u 1 1 1 bh options G711u G711la G726 16 options G711u G71la 2004 Linksys Proprietary See Copyright Notice on Page 2 106 G726 40 Enable 1 Yes FAX Codec_Symmetric 1 Yes DTMF_ Tx Method 1 Auto options InBand AVT INFO
47. 1 4 2 8 Call Return The PHONE ADAPTER supports a service that allows the PHONE ADAPTER to automatically dials the last caller s number 1 1 4 2 9 Call Return on Busy If the last called number is busy the subscriber can order this service to monitor the called party and to receive a notification from the PHONE ADAPTER such as special phone ring when that party becomes available 1 1 4 2 10 Automatic Call Back This feature allows the user to place a call to the last number they tried to reach whether the call was answered unanswered or busy by dialing an activation code 1 1 4 2 11 Call Forwarding These services forward all the incoming calls to a static or dynamically configured destination number based on three different settings These services may be offered by the PHONE ADAPTER or by the SIP proxy server They can be activated by entering certain or code followed by entering a 2004 Linksys Proprietary See Copyright Notice on Page 2 13 telephone number to forward calls to The PHONE ADAPTER provides audio instructions to prompt the user for a forwarding number and confirms that the requested service has been activated Call FWD Unconditional All calls are immediately forwarded to the designated forwarding number The PHONE ADAPTER will not ring or provide call waiting when Call FWD Unconditional is activated Call FWD Busy Calls are forwarded to the designated forwarding number if the subscriber
48. 2 70 from the proxy in the Min Expires header If 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 User Agent Header to be used by the unit in Str63 Linksys Name outbound requests If empty the header is not version included SIP Server Name Server Header to used by the unit in Str63 Linksys responses to inbound responses If empty version the header is not included SIP Accept Accept Language Header to be used by the Str31 Language unit If empty the header is not included Remove Last Reg Remove last registration before registering a Bool no new one if value is different one Use Compact If set to yes the PHONE ADAPTER will use Bool no Header compact SIP headers in outbound SIP messages If set to no the PHONE ADAPTER will use normal SIP headers SIP Timer Values sec SIP T1 RFC 3261 T1 value RTT Estimate Range 0 Time3 5 64 sec SIP T2 RFC 3261 T2 value Maximum retransmit Time3 4 interval for non INVITE requests and INVITE responses Range 0 64 sec SIP T4 RFC 3261 T4 value Maximum duration a Time3 5 message will remain in the network Range 0 64 sec SIP Timer
49. 5 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 0s with Frequency 1 2004 Linksys Proprietary See Copyright Notice on Page 2 34 Segment 2 On 0 38s Off 0s with Frequency 2 Segment 3 On 0 38s Off 0s with Frequency 3 Segment 4 On 0s Off 4s with no frequency components Total Tone Length 20s e ProvisioningRuleSyntax Scripting syntax used to define configuration resync and firmware upgrade rules Refer to the provisioning discussion in the next section for a detailed explanation of the syntax e DialPlanScript Scripting syntax used to specify line 1 and line 2 dial plans Refer to the dial plan section of this document for a detailed explanation of the syntax 4 1 1 Conventions e lt Par Name gt represents a configuration parameter name In a profile the corresponding tag is formed by replacing the space with an underscore _ such as Par_Name e An empty default value field implies an empty string lt gt e The PHONE ADAPTER shall continue to use the last configured values for tags that are not present in a given profile e Templates are compared in the order given The first not the closest match is selected The parameter name must match exactly e If more than one definition for a parameter is given in a configura
50. 7 7 2 4 2 EE 97 7 2 4 3 E EE 97 7 2 4 4 ED EE 97 LECHEN ee le EE 97 F286 Dynamic Payload WEE EE 97 7 2 7 Adjustable Audio Frames Per Packet AAA 97 7 2 8 Fax Tone Detection Pass Throughb 97 7 2 9 DTMF In band amp Out of Band RFC 2833 SIP INFO 97 7 2 10 Call Progress Tone Generation 98 7 2 11 Call Progress Tone Pass Through 98 7 2 12 Jitter Buffer Dynamic Adaptive AAA 98 7 2 13 FullsDuplex AUG EE 98 7 2 14 Echo Cancellation Up to 8 ms Echo Tal 98 7 2 15 Voice Activity Detection with Silence Suppression A Comfort Noise Generaton 98 7 2 16 Attenuation Gain Adluetment 98 7 2 17 Signaling Hook Flash Event iesisnun grire eisene tei iia ted dese E cee EEE 98 7 2 18 Configurable Flash Switch Hook Timer cecceeeceeseeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeseaeeseeeeseaeeeeatess 99 7 2 19 Configurable Dial Plan with Interdigit Timers sseesseeseeeeeeeieeneseneetteentettnsenssirntrnstnnetnneennenne 99 7 2 20 Message Waiting Indicator Tones MT 99 7 2 21 Polarity Control cscs leac ch havi eek EE A ee and oe ceed 99 7 2 22 Calling Party Control CPC ue 99 7 2 23 International Caller ID Delivery AAA 99 7 2 24 Streaming Audio Server GAS 100 7 2 25 Music Oni Hold MOM aa ee aeaa ecard ae r are ia ee a Aa A O A oa ECER a oii ESAs 100 7 3 Security EE 102 7 3 1 Multiple Administration Layers Levels and Permissions AAA 102 7 3 2 HTTP Digest Encrypted Authentication via MD5
51. 711u G711a FAX Codec Symmetric Force unit to use symmetric codec during FAX Bool Yes passthru FAX Passthru Method Choices None NSE RelNVITE Choice NSE FAX Process NSE Bool Yes Release Unused Codec Bool Yes 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 PHONE ADAPTER 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 PHONE ADAPTER 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 4 6 3 Dynamic Payload Types and SDP Codec Names Note You should only need to change the payload type mappings if you are interworking with a non standard implementation Parameter Name Description Type Default NSE Dynamic Payload NSE dynamic payload type Uns8 100 AVT Dynamic Payload AVT dynamic payload type Uns8 101 G726r16 Dynamic Payload G726 16 dynamic payload type Uns8 98 G726r24 Dynamic Payload
52. A DHCP server for auto assignment of private IP addresses subnet mask and default router assignment to devices attached to the private network i e computers IP Telephony 2004 Linksys Proprietary See Copyright Notice on Page 2 8 gateways etc The default router in this case is the IP address of the LAN interface of the router itself Performs NAT on packets sent from the private network to the public network This is an important feature such that recipients of the private packets will perceive them as originated from a public IP address the router s WAN interface and will therefore return messages to the proper public IP address and port Different routers may use different rules for allocating port numbers at the WAN interface to forward packets from a private IP address port to a public IP address port The allocated port number is also used for routing packets from external IP addresses to a private address Most routers will accept a number of static port mapping rules for forwarding packets received on a specific port at the WAN interface to a specific IP address port in the private network PSTN VoIP Gateways These devices are required if user agents are expected to make calls to or receive calls from the PSTN Many gateways may be deployed in order to service a wide area Gateways also behave like SIP user agents The proxy server can be configured with cost saving rules based call routing information so that it may decide whi
53. ADAPTER detects the type of equipment in use on the basis of its answer tone When it detects the equipment answering the call the PHONE ADAPTER performs a switchover from the current audio codec to G 711 codec 7 2 9 DTMF In band amp Out of Band RFC 2833 SIP INFO 2004 Linksys Proprietary See Copyright Notice on Page 2 97 The PHONE ADAPTER may relay DTMF digits as out of band events to preserve the fidelity of the digits This can enhance the reliability of DTMF transmission required by many IVR applications such as dial up banking and airline information 7 2 10 Call Progress Tone Generation The PHONE ADAPTER has configurable call progress tones Parameters for each type of tone may include number of frequency components frequency and amplitude of each component and cadence information 7 2 11 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 7 2 12 Jitter Buffer Dynamic Adaptive The PHONE ADAPTER 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 PHONE ADAPTER has a Network Jitter Level control setting for each line of service The jitter level decides how aggressively the PHONE ADAPTER will try to shrink the jitter buffer over ti
54. B INVITE time out value Range 0 64 sec Time3 32 SIP Timer F Non INVITE time out value Range 0 64 Time3 32 sec SIP Timer H INVITE final response time out value Range Time3 32 0 64 sec SIP Timer D ACK hang around time Range 0 64 sec Time3 32 SIP Timer J Non INVITE response hang around time Time3 32 Range 0 64 sec INVITE Expires INVITE request Expires header value in sec TimeO 180 0 do not include Expires header in INVITE Range 0 2 1 RelNVITE Expires RelNVITE request Expires header value in TimeO 30 sec 0 do not include Expires header in the request Range 0 2 1 Reg Min Expires Minimum registration expiration time allowed TimeO 1 from the proxy in the Expires header or as a Contact header parameter If proxy returns something less this value then the minimum value is used Reg Max Expires Maximum registration expiration time allowed Timed 7200 2004 Linksys Proprietary See Copyright Notice on Page 2 71 value is larger than this then the maximum value is used Reg Retry Intvl Interval to wait before the PHONE ADAPTER retries registration again after encountering a failure condition during last registration Time 30 Reg Retry Long Interval When Registration fails with a SIP response code that does no match lt Retry Reg RSC gt the PHONE ADAPTER will wait for the delay specified in this parameter before retrying If this parameter i
55. Caller ID information for a call is desired i e trying to reach a party that does not accept Caller ID blocked calls User Action Required to Activate or Use Lift the receiver Listen for dial tone Drees Listen for dial tone Dial the telephone number you are calling Expected Call and Network Behavior User Action Required to Deactivate or End Caller ID will be sent to the distant party for this call only Users must repeat this process at the start of each call No action required This service is only in 2004 Linksys Proprietary See Copyright Notice on Page 2 80 effect for the duration of the current call 5 5 Calling Line Identification Restriction CLIR Caller ID Blocking Service Description This feature allows the user to block the delivery of their Caller ID to the number they are calling This feature must be activated prior to dialing each call and is only in effect for the duration of each call User Action Required to Activate or Use Lift the receiver Listen for dial tone Drees Listen for dial tone Dial the telephone number you are calling You must repeat this process at the start of each call Expected Call and Network Behavior The user activates this service to hide his Caller ID when making an outgoing call User Action Required to Deactivate or End No action required This service is only in effect for the duration of the
56. Cancel call forward no answer ActCode_ 93 Cfwd Last Act Code Forward the last inbound or outbound calls ActCode 63 to the target specified after the activation code 2004 Linksys Proprietary See Copyright Notice on Page 2 58 Cfwd Last Deact Code Cancel call forward last ActCode_ 83 Block Last Act Code Block the last inbound call ActCode_ 60 Block Last Deact Code Cancel blocking of the last inbound call ActCode_ 80 Accept Last Act Code Accept the last outbound call Let it ring ActCode 64 through when DND or Call Forward All is in effect Accept Last Deact Code Cancel Accept Last ActCode_ 84 Call Back Act Code Callback when the last outbound call is not ActCode 66 busy Call Back Deact Code Cancel callback ActCode_ 86 CW_Act_Code Enable Call Waiting on all calls ActCode_ 56 CW_Deact_Code Disable Call Waiting on all calls ActCode_ 57 CW Per Cal Act Code Enable Call Waiting for the next call ActCode_ 71 CW Per Call Deact_Code Disable Call Waiting for the next call ActCode_ 70 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 outbound call ActCode_ 81 Blcok_CID Per Call Deact Code Unblock CID on the next inbound call ActCode_ 82 Block ANC Act Code Block all anonym
57. Cisco SYSTEMS LINKSYS A Division of Cisco Systems Inc 17401 Armstrong Ave Irvine CA 92614 Linksys PAP2 and RT31P2 PHONE ADAPTER Administration Guide August 2004 2004 Linksys Proprietary See Copyright Notice on Page 2 1 Disclaimer Please Read This document contains implementation examples and techniques using Linksys and in some instances other company s technology and products and is a recommendation only and does not constitute any legal arrangement between Linksys and the reader either written or implied The conclusions reached and recommendations and statements made are based on generic network service and application requirements and should be regarded as a guide to assist you in forming your own opinions and decision regarding your particular situation As well Linksys Technology 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 Major portions of this document are the sole property of Sipura Technology Inc and are provided to its licensee Linksys LLC and protected by United States and international copyright laws c 2003 2004 Sipura technology Inc All rights reserved 2004 Linksys Proprietary See Copyright Notice on Page 2 2 Table of Contents Te Ol ttogiOeHgtt ege fide te hit dil eee ea ie aie
58. Drees Listen for dial tone then dial the number you want to call Call Waiting is now disabled for the duration of this call only To deactivate Call Waiting while on a call Press the switch hook or flash button briefly This puts the first call on hold Listen for three short tones and then a dial tone Drees Listen for dial tone then return to your call by pressing the switch hook or flash button Call Waiting is now disabled for the duration of this call To deactivate Call Waiting while on a permanent basis until cancelled Lift the receiver Listen for dial tone Drees You will hear a confirmation tone signaling your request to cancel Call Waiting has been accepted Expected Call and Network Behavior Callers who dial your number will receive a busy signal or if available the caller will be forwarded to voice mail or another predetermined forwarding number User Action Required to Deactivate or End If you have cancelled Call Waiting temporarily 2004 Linksys Proprietary See Copyright Notice on Page 2 82 no user action is required If you deactivated call waiting and wish to reinstate the service do the following Lift the receiver Listen for dial tone Drees You will hear a confirmation tone signaling your request to cancel Call Waiting has been accepted 5 8 Call Waiting with Caller ID Service Description When the user is on the p
59. ER will not attempt to fall back after a fail over Subscriber Information 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 Bool No are used for SIP authentication Else the pair lt User ID gt and lt Password gt are used 4 5 Configuration for NAT Traversal In general there are 3 general approaches to enable NAT traversal available on the PHONE ADAPTER STUN Simple Traversal of UDP through NAT Using an outbound rewriting proxy and manual configuration If the PHONE ADAPTER is not behind a NAT the default settings should be used Note The Linksys model RT31P2 includes NAT Network Address Translator functionality As long as the IP address of the WAN Port is a public IP address the RT31P2 can be configured with all NAT Traversal features NAT Traversal off since the PHONE ADAPTER portion shares the same IP address as the WAN Port If the address obtained on the WAN Port is already a private address then the RT31P2 still needs to be configured for NAT traversal The Outbound Proxy approach works through more than 99 of NATs but it requires the service provider to relay RTP media packets for every call To use this approach set the foll
60. Hardware Overview The PHONE ADAPTER has one of the smallest form factors on the market It can be installed in minutes as a table top or wall mount CPE device Figures Figure 2 and Figure 3 show the front and rear of the PHONE ADAPTER respectively Figures 4 and 5 show the front and rear of the RT31P2 Broadband Router respectively 2004 Linksys Proprietary See Copyright Notice on Page 2 17 Figure 3 PAP2 Back Figure 2 PAP2 Front Broadband Router Figure 3 RT31P2 Front Figure 4 RT31P2 Back The PAP2 PHONE ADAPTER has the following interfaces for networking power and visual status indication 1 Two 2 RJ 11 Type Analog Telephone Jack Interfaces Figure 3 above These interfaces accept standard RJ 11 telephone connectors An Analog touchtone telephone or fax machine may be connected to either interface If the service supports only one incoming line the analog telephone or fax machine should be connected to port one 1 of the PHONE ADAPTER Port one 1 is the outermost telephone port on the PHONE ADAPTER and is labeled Phone 1 2 One Ethernet 10baseT RJ 45 Jack Interface Figure 3 above This interface accepts a standard or crossover Ethernet cable with standard RJ 45 connector For optimum performance Linksys recommends that a Category 5 cable or greater be used in conjunction with the PHONE ADAPTER The Broadband Router RT31P2 has the following interfaces for networking power and visual
61. 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 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
62. In wen wai wou wou 5 A He H HE HE options 1 2 3 4 5 6 7 8 options 1 2 3 4 5 6 7 8 options 1 2 3 4 5 6 7 8 none options 1 2 3 4 5 6 7 8 New VM Available Available New VM Becomes Available New VM Arrives W No n Yes No A 30 7 wu ut No z No b SNOTIFY 1 SPROXY 0x68 high Oxb8 5061 W No wu Ye s W none 1 b H options options New VM low medium high very high options none 1 line 1l line excl OPT full excl NTFY full excl REG full excl OPT NTFY REG Call Feature Settings Blind Attn Xfer Enable 2 MOH Server 2 Xfer When _Hangup_ Conf 2 Proxy and Registration Proxy 2 Use_Outbound_ Proxy 2 No wu H ut Yes wu W No S b 2004 Linksys Proprietary See Copyright Notice on Page 2 108 Outbound Proxy 2 Use_OB Proxy In Dialog 2 Register 2 Make Call Without_Reg 2 Register Expires 2 Ans_ Call Without_Reg 2 Use DNS SRV 2 DNS SRV Auto Prefix 2 Proxy Fallback _Intvl 2 Voice Mail Server 2 Subscriber Information Display Name 2 User _ ID 2 Password 2 Use_Auth_ID 2 Auth_ID 2 Mini Certificate 2 SRTP_ Private Key 2 wu Yes Ye s W No S 3600 No A No No 7 3600 wu wu wu wu No R wu wu wu Supplementary Service Subscription Call Waiting Serv 2 Block CID Serv 2 Block ANC Serv 2 Dis
63. InBand AVT INFO Auto FAX Passthru_Method 2 NSE options None NSE ReINVITE Hook _ Flash _ Tx Method 2 None options None AVT INFO FAX Process _ NSE 2 Yes Release_Unused_Codec 2 Yes Dial Plan Dial Plan 2 xx 3469 11 0 00 2 9 xxxxxx 1xxx 2 9 xxxxxxS0 XXXXXXXXXXXX Enable IP Dialing 2 No FXS Port Polarity Configuration Idle Polarity 2 Forward options Forward Reverse Caller Conn _Polarity 2 Forward options Forward Reverse Callee Conn Polarity 2 Forward options Forward Reverse Call Forward Settings Cfwd_All Dest 2 AH Cfwd_Busy Dest 2 Cfwd_No_ Ans Dest 2 Cfwd_No_ Ans Delay 2 wu wu I H 20 G Selective Call Forward Settings Cfwd_ Sell Caller 2 Cf wd_Sell Dest 2 Cfwd_Sel2 Caller 2 Cfwd_Sel2 Dest 2 Cf wd_Sel3 Caller 2 Cfwd_Sel3 Dest 2 Cf wd_Sel4 Caller 2 Cfwd_Sel4 Dest 2 Cfwd_Sel5 Caller 2 Cf wd_Sel5 Dest 2 Cfwd_Sel6 Caller 2 Cfwd_Sel6 Dest 2 Cfwd_Sel7_ Caller 2 Cf wd_Sel7 Dest 2 Cfwd_Sel8 Caller 2 Cfwd_Sel8 Dest 2 Cfwd_Last_ Caller 2 Cfwd_Last_ Dest 2 Block Last _Caller 2 Accept Last Caller 2 wu wu wu wu wu wu wu wu wu wu wu wu wu wu wu wu wu wu wu I l l l I l l I l l l I l l l I foun Speed Dial Settings Speed Dial 2 2 wu Speed Dial 3 2 Speed Dial 4 2 S
64. L 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 e 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 e ActCode Activation code for a supplementary service such as 69 It can contain up to 7 characters e PhTmplt A phone number template Each template may contain 1 or more patterns separated by a White Phone Adapterce at the beginning of each pattern is ignored and represent 2004 Linksys Proprietary See Copyright Notice on Page 2 32 wildcard characters It can contain up to 39 characters Examples 1408 1510 1408123 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 GG where Si D on 1 0ffi 1 0n 2 off 2 on 3 off s 0n 4 off 4 0n 5 off s 0ni 6 offi and is known as a section on j and oft are the on off duration in seconds of a segment and i 1 or 2 andj 1 to 6 D is the total duration of the section in seconds All durations can have up to 3 decimal places to provide 1 ms resolution The wildcard character
65. MF playback level are not affected by the lt FXS Port Output Gain gt 2 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_Short_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 3 PHONE ADAPTER has had polarity reversal feature since release 1 0 which can be applied to both the caller and the callee end This feature is generally used for answer supervision on the caller side to signal to the attached equipment when the call has been connected remote end has answered or disconnected remote end has hung up This feature should be disabled for the called party ie by using the same polarity for connected and idle state and the CPC feature should be used instead 4 Without CPC enabled reorder tone will is played after a configurable delay If CPC is enabled dial tone will be played when tip to ring voltage is restored 4 10 5 Miscellaneous Parameters Parameter Name Description Type Default Set Local Date Setting the local date year is optional and Str10 mm dd yyyy can be 2 digit or 4 digit Local Time HH mm ss Setting the local time second is optional Str8 Time Zone Number of hours to add to GMT to form local Choice GMT 07 00 time for ca
66. NAT Mapped SIP Port Call 1 2 Status State one 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 Tone playing for this call Dial 2 Dial Outside Dial Ring Back Ring Busy Reorder SIT1 4 Call Waiting Call Forward Conference 2004 Linksys Proprietary See Copyright Notice on Page 2 92 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
67. PHONE ADAPTER can be obtained from the profile compiler tool using the following command line arguments spc sample profile defaults txt In both the XML and SPC configuration formats Boolean parameter values that evaluate to true are any one of the values Yes yes Enable enable 1 Boolean values that evaluate to false are any one of the values No no Disable disable 0 3 1 1 Using the Supplemental Profile Compiler Once a plain text file has been generated with the desired parameter settings it needs to be compiled into a binary CFG file The profile compiler can generate a generic unencrypted CFG file a targeted CFG file encrypted for a unique PHONE ADAPTER a generic key encrypted CFG file or a targeted and key encrypted CFG file A generic CFG file non targeted is accepted as valid by any PHONE ADAPTER device A targeted CFG file is only accepted as valid by the PHONE ADAPTER device bearing the target MAC address Targeted CFG files are encrypted with a 128 bit algorithmically generated key and therefore do not require a key to be issued explicitly Targeted CFG files provide a basic level of security for remotely locking an otherwise unprovisioned PHONE ADAPTER The binary configuration format supports RC4 and AES symmetric key algorithms with keys of up to 256 bits The key can be specified explicitly as a hex string or it can be generated from a password or a quoted pass phrase In the case
68. Port TCP port through which the PHONE ADAPTER web Uns8 80 server will communicate Enable Web Admin Enable disable Admin pages of web server of PHONE Bool Yes Access ADAPTER 2004 Linksys Proprietary See Copyright Notice on Page 2 46 Protect IVR Factory Bool No Reset Admin Password The password for administrator Str63 User Password The password for User Str63 4 3 Basic Networking Configuration Configuration parameters in this list are used for setting up basic network connectivity In general many of these parameters are set automatically for example using DHCP or are configured by the end user of the device Note that the RT31P2 ignores the following parameters DHCP Static_IP NetMask and Gateway Other than the DNS_Server_Order and DNS_Query_Mode the rest these parameters also can be configured from the RT31P2 User GUI Network Configuration Parameter Name Description Type Default DHCP Enable Disable DHCP Bool Yes Host Name Host Name of PHONE ADAPTER Str31 Domain The network domain of PHONE ADAPTER Str127 Static IP Static IP address of PHONE ADAPTER which will take IP 0 0 0 0 effect if DHCP is disabled NetMask The NetMask used by PHONE ADAPTER when DHCP IP 255 255 255 is disabled A Gateway The default gateway used by PHONE ADAPTER when IP 0 0 0 0 DHCP is disabled Primar
69. 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 6 2 Enabling Logging and Debugging The PHONE ADAPTER uses the following parameters to enable logging and debugging both using the syslog protocol over UDP e Syslog Server e Debug_Server e Debug_Level 6 3 Error and Log Reporting The PHONE ADAPTER Error Status Code ESC is used to indicate the current operation status of the PHONE ADAPTER 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 PHONE ADAPTER web page 6 4 Internal Error Codes The PHONE ADAPTER defines a number of internal error codes X00 X99 to facilitate configuration in providing finer control over the behavior of the unit under certain error conditions They can be viewed as extensions to the SIP response codes 100 699 The definitions are shown below Error Code Description X00 Transport layer or ICMP error when sending a SIP request X20 SIP request times out wh
70. 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 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 Blind Attn Xfer If enabled the PHONE ADAPTER performs an Bool No Enable attended transfer operation by terminating the current call leg and blind transferring the other call leg If disabled the PHONE ADAPTER performs an attended transfer by referring the other call leg to the current call leg while maintaining both call legs Notes 1 If proxy responded to REGISTER with a smaller Expires value the PHONE ADAPTER will renew registration based on this smaller value instead of the configured value If registration failed with an Expires too brief error response the PHONE ADAPTER 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 holding MOH to work That is the SIP 2xx response from the remote party in reply to the re INVITE from the PHONE ADAPTER to put the call on h
71. STN Interworking The PHONE ADAPTER is designed to provide a transparent interworking relationship with the PSTN Service providers can deploy the PHONE ADAPTER in such a way that PSTN endpoints wired or wireless communicating with PHONE ADAPTER endpoints do so without modification to their configuration or network settings The service provider may choose to deploy a multi protocol VoIP network much the same way the PSTN supports multiple signaling schemes today Most telecommunication providers operate equipment that supports CAS or channel associated signaling ISDN signaling and SS7 signaling When VolP is introduced or used in the telecommunications landscape it is likely that the service provider will implement a signaling gateway that supports multiple IP Telephony protocols along with legacy PSTN protocols The signaling gateway is commonly referred to as a Softswitch Architecture and functionality can vary greatly amongst the different softswitch vendors The protocols used will depend on the types of connections that will be set up across the service provider s network If the provider is simply providing transport of calls to from their network to another provider s network but not originating or terminating calls with the endpoints SIP will likely be used for softswitch to softswitch communication 2004 Linksys Proprietary See Copyright Notice on Page 2 14 If the service provider is offering origination and or terminat
72. Selective 5 PhTmplt Cfwd Sel6 Caller Caller number pattern to trigger Call Forward Selective 6 PhTmplt Cfwd Sel7 Caller Caller number pattern to trigger Call Forward Selective 7 PhTmplt Cfwd Sel8 Caller Caller number pattern to trigger Call Forward Selective 8 PhTmplt Cfwd Sel1 Dest Forward number for Call Forward Selective 1 Phone Cfwd Sel2 Dest Forward number for Call Forward Selective 2 Phone Cfwd Sel3 Dest Forward number for Call Forward Selective 3 Phone Cfwd Sel4 Dest Forward number for Call Forward Selective 4 Phone Cfwd Sel5 Dest Forward number for Call Forward Selective 5 Phone Cfwd Sel6 Dest Forward number for Call Forward Selective 6 Phone Cfwd Sel7 Dest Forward number for Call Forward Selective 7 Phone Cfwd Sel8 Dest Forward number for Call Forward Selective 8 Phone Block Last Caller ID of caller blocked via the Block Last Caller service Phone Accept Last Caller ID of caller accepted via the Accept Last Caller service Phone Cfwd Last Caller The Caller number that is actively forwarded to lt Cfwd Phone Last Dest gt by using the Call Forward Last activation code Cfwd Last Dest Forward number for the lt Cfwd Last Caller gt Phone 4 7 3 Supplementary Services implemented in the service provider network For services which are activated or deactivated in the service provider network for example in an application server instead o
73. Server SAS This feature allows one to attach an audio source to one of the PHONE ADAPTER FXS ports and use it aS a streaming audio source device The corresponding Line 1 or 2 can be configured as a streaming audio server SAS such that when the Line is called the PHONE ADAPTER answers the call automatically and starts streaming audio to the calling party provided the FXS port is off hook If the FXS port is on hook when the incoming call arrives the PHONE ADAPTER replies with a SIP 503 response code to indicate Service Not Available If an incoming call is auto answered but later the FXS port becomes on hook the PHONE ADAPTER does not terminate the call but continues to stream silence packets to the caller If an incoming call arrives when the SAS line has reached full capacity the PHONE ADAPTER replies with a SIP 486 response code to indicate Busy Here The SAS line can be setup to refresh each streaming audio session periodically via SIP re INVITE to detect if the connection to the caller is down If the caller does not respond to the refresh message the SAS line will terminate the call so that the streaming resource can be used for other callers 7 2 25 Music On Hold MOH On a connected call the PHONE ADAPTER may place the remote party on call the only way to do this on te PHONE ADAPTER is to perform a hook flash to initiate a 3 way call or to swap 2 calls during call waiting If the remote party indicates that the
74. Service Bool Yes Cfwd Last Serv Enable Forward Last Call Service Bool Yes Block Last Serv Enable Block Last Call Service Bool Yes Accept Last Serv Enable Accept Last Call Service Bool Yes DND Serv Enable Do Not Disturb Service Bool Yes CID_Serv Enable Caller ID Service Bool Yes 2004 Linksys Proprietary See Copyright Notice on Page 2 57 CWCID Serv Enable Call Waiting Caller ID Service Bool Yes Call Return Serv Enable Call Return Service Bool Yes Call Back Serv Enable Call Back Service Bool Yes Three Way Call Serv Enable Three Way Calling Service Bool Yes Three Way Conf Enable Three Way Conference Service Bool Yes Serv Attn Transfer Serv Enable Attended Call Transfer Service Bool Yes Unattn Transfer Serv Enable Unattended Blind Call Transfer Bool Yes Service MWI Serv Enable MWI Service Bool Yes VMWI Serv Enable VMWI Service FSK Bool Yes Speed Dial Serv Enable Speed Dial Service Bool Yes Secure Call Serv Enable Secure Call Service Bool Yes Referral Serv Enable Referral Service See lt Referral Bool Yes Services Codes gt for more details Feature Dial Serv Enable Feature Dial Service See lt Feature Bool Yes Dial Services Codes gt for more details Notes 1 Three Way Calling is required for Three Way Conference and Attended Transfer 2 Three Way Conference is required for Attended Transfer 3 MWI is available only if a Voice Mail Service
75. Speed Dial 7 1 Speed Dial 8 1 Speed Dial 9 1 wu wu wu wu wu wu wu Supplementary Service Settings CW Setting 1 Yes Block CID Setting 1 No Block ANC Setting 1 No DND_Setting 1 No CID Setting 1 Yes CWCID Setting 1 Yes Dist_Ring_Setting 1 Yes 2004 Linksys Proprietary See Copyright Notice on Page 2 107 Secure Call Setting 1 Distinctive Ring Settings Ringl Caller 1 Ring2 Caller 1 Ring3 Caller 1 Ring4 Caller 1 Ring5 Caller 1 Ring6 Caller 1 Ring7 Caller 1 Rings Caller 1 Ring Settings Default Ring 1 Default _CWT 1 Hold Reminder _Ring 1 Call Back _Ring 1 Cfwd_Ring Splash_Len 1 Cb1lk_ Ring Splash_Len 1 VMWI_Ring Splash _Len 1 VMWI_ Ring Policy 1 Ring On No New_VM 1 KKK Line Enable 2 Streaming Audio Server SAS Enable 2 SAS DLG Refresh _Intvl1 2 SAS Inbound _RTP_ Sink 2 NAT Settings NAT Mapping Enable 2 MAT Keep Alive Enable 2 NAT_Keep Alive Msg 2 NAT_Keep Alive Dest 2 Network Settings SIP_TOS DiffServ_Value 2 Network Jitter Level 2 RTP_TOS DiffServ_ Value 2 SIP Settings SIP Port 2 SIP_100REL Enable 2 EXT SIP Port 2 Auth_Resync Reboot 2 SIP Debug Option 2 OPT 1 line excl NTFY 1 line excl REG 1 line excl OPT NTFY REG full full excl SAS No G wu wu wu wu wu wu wu wu In
76. The technical challenges in deploying and operating a residential IP Telephony service however are not small One of the main challenges is to make the service transparent to subscribers The subscribers shall expect to use their existing phones to make or receive calls in the same way as with the existing PSTN service To enable this level of transparency the IP Telephony solution has to be tightly integrated A key element in this end to end IP Telephony solution is the provision of an endpoint device that sits at a subscriber s premises that serves as an IP Telephony gateway or telephone adapter This phone adapter offers one or more standard telephone RJ 11 phone ports identical to the phone wall jacks at home where the subscriber can plug in their existing telephone equipment to access phone services The IP Telephony gateway may connect to the IP network like the Internet through an uplink Ethernet connection Important Please note The information contained herein is not a warranty from Linksys Customers planning to use the PHONE ADAPTER in a VoIP service deployment are warned to test all functionality they plan to support in conjunction with the PHONE ADAPTER before putting the PHONE ADAPTER in service Some information in Section 1 of this guide is written for educational purposes and describes functionality not yet implemented in the PHONE ADAPTER 1 1 The Session Initiation Protocol There are many excellent articles and books th
77. a middle man server or media gateway controller Examples of proper reactions are playing dial tone collecting DTMF digits comparing them against a dial plan and terminating a call With intelligent endpoints at the edges of a network performing the bulk of the call processing duties the deployment of a large network with thousands of subscribers can scale quickly without the introduction of complicated expensive servers As described later in this section the Session Initiation Protocol SIP is a good choice of call signaling protocol for the implementation of such a device in this type of network The phenomenal growth of broadband Internet access DSL Cable FTTH etc has brought the realization of reliable packet switched IP Telephony Services with circuit switched toll quality and subscriber feature transparency with that of the PSTN s CLASS feature set In addition to basic offerings comparable to traditional PSTN services many service providers have integrated their IP Telephony offering with a large number of web based productivity applications like unified messaging and call management features such as remote call forward configuration via the web Such advances over traditional phone services with equal or better voice quality and lower per minute prices have made IP Telephony service a viable business In fact IP Telephony service providers in the US and abroad have seen their subscriber base growing at a rapid pace
78. ad the switch hook or flash button while the PSTN presents instructions information confirmation to the subscribers through a variety of audio tones beeps and or announcements The PHONE ADAPTER supports a comparable range of services via a similar user interface in order to make the IP Telephony service transparent to subscribers The PHONE ADAPTER is fully programmable and can be custom provisioned to emulate just about any traditional telephony service available today This ability to transparently deliver legacy services over an IP network coupled with the availability of Internet connected devices PCs PDA etc and browsers opens up a new world of potential offerings that a provider can use to differentiate their service and grow their business The following is a list of commonly supported phone services 2004 Linksys Proprietary See Copyright Notice on Page 2 11 1 1 4 1 Basic Services 1 1 4 1 1 Making Calls to PSTN and IP Endpoints This is the most basic service When the user picks up the handset the PHONE ADAPTER provides dial tone and is ready to collect dialing information via DTMF digits from a touch tone telephone While it is possible to support overlapped dialing within the context of SIP the PHONE ADAPTER collects a complete phone number and sends the full number in a SIP INVITE message to the proxy server for further call processing In order to minimize dialing delay the PHONE ADAPTER maintains a dial plan and mat
79. all calls Bool No Block ANC Setting Block Anonymous Calls on or off Bool No DND Setting Do Not Disturb on or off Bool No CID Setting Caller ID Generation on or off Bool Yes CWCID Setting Call Waiting Caller ID Generation on or off Bool Yes Dist Ring Setting Distinctive Ring on or off Bool Yes 2004 Linksys Proprietary See Copyright Notice on Page 2 59 Secure Call Setting If yes all outbound calls are secure calls by default Bool No 4 7 2 Call Forwarding Implemented internally The PHONE ADAPTER supports local call forwarding services Call Forward All Call Forward Busy Call Forward No Answer and Selective Call Forwarding for up to 8 numbers Parameter Name Description Type 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 Cfwd Gel Caller Caller number pattern to trigger Call Forward Selective 1 PhTmplt Cfwd Sel2 Caller Caller number pattern to trigger Call Forward Selective 2 PhTmplt Cfwd Sel3 Caller Caller number pattern to trigger Call Forward Selective 3 PhTmplt Cfwd Sel4 Caller Caller number pattern to trigger Call Forward Selective 4 PhTmplt Cfwd Sel5 Caller Caller number pattern to trigger Call Forward
80. 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 ActCode_ 02729 4 7 Supplementary Services Each line of the PHONE ADAPTER has settings which enable or disable each of the supplementary services implemented directly in the PHONE ADAPTER The expected behavior when a specific service is enabled is described in Section 5 The PHONE ADAPTER provides native support of a large set of enhanced or supplementary services All of these services are optional The parameters listed in the following table are used to enable or disable a specific supplementary service A supplementary service should be disabled if a the user has not subscribed for it or b the Service Provider intends to support similar service using other means than relying on the PHONE ADAPTER Parameter Name Description Type Default Call Waiting Serv Enable Call Waiting Service Bool Yes Block CID Serv Enable Block Caller ID Service Bool Yes Block ANC Serv Enable Block Anonymous Calls Service Bool Yes Dist Ring Serv Enable Distinctive Ringing Service Bool Yes Cfwd All Serv Enable Call Forward All Service Bool Yes Cfwd Busy Serv Enable Call Forward Busy Service Bool Yes Cfwd No Ans Serv Enable Call Forward No Answer Service Bool Yes Cfwd Sel Serv Enable Call Forward Selective
81. at discuss the advantages of SIP Here are some of the more popular details e SIP message constructs are very similar to those of HTTP which is well known to be IP Network Internet friendly e SIP is transport agnostic meaning it can be used over TCP IP or UDP IP with or without security e SIP has a better chance of traversing NATs than other control protocols 2004 Linksys Proprietary See Copyright Notice on Page 2 6 e SIP enables the implementation of intelligent endpoints to support scalable advanced services In a nutshell SIP is a distributed signaling protocol as opposed to a centralized protocol such as SS7 MGCP or MEGACO H 248 With a distributive protocol the intelligence does not necessarily reside on a central server but can be built into the individual endpoints By moving the intelligence to reside within the endpoints at the edge of the network the processing load of the network application and associated call servers are significantly reduced thus making the network a very scalable solution 2004 Linksys Proprietary See Copyright Notice on Page 2 7 1 1 1 Components of a SIP Network Subscriber Service Database Provider Domain Broadband i Modem Router NAT Subscriber Domain IP Network Internet Private IP Network Figure 1 Components of a SIP IP Telephony Network IP Telephony Gateway PHONE ADAPTER The PHONE ADAPTER is a
82. ation menu may be accessed 1 The PHONE ADAPTER IVR uses the following conventions By factory default there is no password and no password authentication is prompted for all the IVR settings If administrator password is set password authentication will be prompted for certain IVR settings See 3 4 2 for detailed information about administrator password To input the password using the phone keypad the following translation convention applies o To input A B C a b c press 2 o To input D E F d e f press 3 o To input G H l g h i press 4 o To input J K L j k press 5 o To input M N O m n o press 6 o To input P Q R S p q r s press 7 o To input T U V t u v press 8 o To input W X Y Z w x y Z press 9 o To input all other characters in the administrator password press 0 Note This translation convention only applies to the password input For example to input password test 1234 by phone keypad you need to press the following sequence of digits 8378001234 2 After entering a value press the pound key to indicate end of input o To Save value press 1 o To Review the value press 2 o To Re enter the value press 3 o To Cancel the value entry and return to the main configuration menu press star o The final key won t be counted into value o Saved settings will
83. call by pressing the switch hook or flash button creating a three way conference When the user hangs up the phone the transferee and the called party 2004 Linksys Proprietary See Copyright Notice on Page 2 84 remain in a call User Action Required to Deactivate or End Not applicable 5 11 Unattended or Blind Call Transfer Service Description Unattended or Blind Call Transfer lets a customer use their Touchtone phone to send a call to any other phone inside or outside their business including a wireless phones User Action Required to Activate or Use While in a call with the party to be transferred Press the switch hook or flash button on the phone to place the party on hold Enter Dial the number to which you will transfer the caller The call is transferred when a complete number is entered You will hear a short confirmation tone followed by regular dial tone Expected Call and Network Behavior When the user presses the switch hook or flash button the transferee is placed on hold When the user successfully dials the transfer number the transferee will automatically call the dialed number User Action Required to Deactivate or End No applicable 5 12 Call Hold Service Description Call Hold lets you put a caller on hold for an unlimited period of time It is especially useful on phones without the hold button Unlike a h
84. ch gateway to use depending on the destination and the time of the call The IP Telephony service provider will assign each subscriber an E164 telephone number so that it may be reached from the PSTN just like any other telephone Billing Servers Billing servers are used to generate billing data per usage of the IP Telephony service Typically the service provider will charge a flat fee for unlimited calls between IP Telephony subscribers on net to on net calls Per use or minute chargers will be incurred only when the subscriber makes calls to PSTN numbers on net to off net calls through one of the PSTN gateways CDR call detail record data are generated by the PSTN gateway and sent to the billing servers Provisioning Servers Provisioning servers are used to provision the subscriber user agent devices e g the PHONE ADAPTER When a subscriber signs up for IP Telephony service he selects an appropriate service level and enters his personal information including billing information This information is processed by the provisioning server and stored into the service provider s customer database The provisioning server generates a device profile based on the subscriber s choice of options The device profile which is list of configuration parameters is downloaded into the PHONE ADAPTER from the provisioning server The PHONE ADAPTER can be configured to contact the provisioning server periodically to check for any update of the device profile
85. ches it against the cumulative number entered by the user The PHONE ADAPTER also detects invalid phone numbers not compatible with the dial plan and alerts the user via a configurable tone reorder or announcement 1 1 4 1 2 Receiving Calls from PSTN and IP Endpoints The PHONE ADAPTER can receive calls from the PSTN or other IP Telephony subscribers Each subscriber is assigned an E 164 phone number so that they may be reached from wired or wireless callers on the PSTN The PHONE ADAPTER supplies ring voltage to the attached telephone set to alert the user of incoming calls 1 1 4 2 Enhanced Services Enhanced Services are provided in addition to Basic calling services and accessed by way of a touchtone phone through a series of menus Since the service enabled by the PHONE ADAPTER are Internet in nature these enhanced services can be made better by offering users a web browser based interface to control certain aspects of some or all services 1 1 4 2 1 Caller ID In between ringing bursts the PHONE ADAPTER can generate a Caller ID signal to the attached phone when the phone is on hook Calling Line Identification Presentation CLIP Some subscribers will elect to always block their Caller ID information yet there may be a circumstance where sending Caller ID information for a particular call is desired i e trying to reach a party that does not accept Caller ID blocked calls The subscriber activates this service to send his Call
86. crypted XML configuration profiles This can be used for subsequent configuration files stored on or generated by either TFTP or HTTP servers When used in concert with HTTPS for initial config this provides complete security but only uses the HTTPS server for initial enrollment For example an example configuration file in XML setup to download an encrypted XML file via HTTP looks like this lt flat profile gt lt Profile_Rule gt key B http config provider net linksys established MA xml lt Profile_Rule gt lt Resync_Periodic gt 86400 lt Resync_Periodic gt lt GPP_B gt 9b4cef5677a129 lt GPP_B gt lt Admin_Passwd gt 9b4cef5677a1 29 lt Admin_Passwd gt lt Proxy_1_ gt sip provider net lt Proxy_1_ gt lt User_ID_1_ gt 1234567890 lt User_ID_1_ gt lt Password_1_ gt YhJ89_Luk4E lt Password_1_ gt lt Display_Name_1_ gt 1234567890 lt Display_Name_1_ gt lt Line_Enable_2 gt 0 lt Line Enable 2 gt lt flat profile gt An XML configuration file can be encrypted using the openssl command line utility as shown below Note that aes encryption is available beginning with OpenSSL versions 0 9 7 OpenSSL is freely available from http www openssl org openssl aes 256 cbc e in cleartextconfig out encryptedconfig k 9b4cef5677a129 2004 Linksys Proprietary See Copyright Notice on Page 2 24 This utility generates 8 bytes of salt which is prepended to the encrypted configuration file and then calculates an Initial
87. d Make Call Without Allow making outbound calls without successful Bool No Reg 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 Timed 3600 PHONE ADAPTER will 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 DNS SRV Auto Prefix If enabled the PHONE ADAPTER will Bool No automatically prepend the Proxy 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 TimeO 3600 PHONE ADAPTER will retry from the highest priority proxy or outbound proxy servers after it has failed over to a lower priority server This 2004 Linksys Proprietary See Copyright Notice on Page 2 48 parameter is useful only if the primary and backup proxy server list is provided to the PHONE ADAPTER 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 PHONE ADAPT
88. d parameters General Purpose Secure Parameters ParName GPP_SA through GPP_SD Default empty GPP_SA through GPP_SD are the 4 Secure General Purpose Parameters usable by both the provisioning and the upgrade logic Each secure parameter can be configured to hold any string value Such a value can then be incorporated in other scripted parameters The secure parameters are not accessible through the PHONE ADAPTER web interface and can only be set viaa configuration profile Also the parameters cannot be incorporated as part of a syslog message a SIP command Parameter Name Description Type Default Provision Enable Master enable for configuration profile Bool yes resync operations Resync On Reset Resyncs configuration profile from Bool yes configuration server whenever the PHONE ADAPTER resets Resync Random Spread interval for resync requests TimeO 2 Delay Resync Periodic Resyncs configuration profile periodically Time 3600 after reset Resync Error Retry Retry interval following resync failure Timed 3600 Delay Resync From SIP Enables resync of configuration profile from Bool Yes Resync After Upgrade Attempt Resync Trigger 1 Bool 2004 Linksys Proprietary See Copyright Notice on Page 2 42 Resync Trigger 2 Profile Rule Configuration profile URL script ProfileScri
89. e Success Msg successful upgrade provisioning discussion section Log Upgrade Syslog message generated after a failed UpgradeMsg See Failure Msg upgrade provisioning discussion section Note In a customized PHONE ADAPTER the upgrade rule would point to a service provider s server 4 2 2 Provisioning Server Redundancy The Provisioning Server PS may be specified as an IP address or a FQDN PS redundancy is not available in the former case For the latter PHONE ADAPTER shall attempt to resolve the IP address of the PS via DNS SRV then DNS A Record In either case the DNS server may return a number of IP addresses with priority priority can be indicated in the case of SRV record for A records all IP addresses have the same priority The PHONE ADAPTER then contacts the IP address with the highest priority If that fails the PHONE ADAPTER shall contact the next available IP address The PHONE ADAPTER shall continue the process until one of the PS responds If all PS fail to respond the PHONE ADAPTER shall log an error to the Syslog server 4 2 3 Configuring the Web Server and IVR 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 PHONE ADAPTER Bool Yes This feature should only be used on firmware version 1 0 9 or later Web Server
90. e number configured in the PHONE ADAPTER Speed dials are specified per line Parameter Name Description Type Default 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 2004 Linksys Proprietary See Copyright Notice on Page 2 66 4 9 Progress Tone and Ring Configuration The progress tones and ring tones on the PHONE ADAPTER are extremely configurable There are 18 configurable call progress tones 8 configurable ringing cadences and 8 configurable call waiting cadences Progress tones and Ring cadences are configured using FreqScipts and CadScripts respectively described in Section 4 1 4 9 1 Distinctive Ring and Other Ring Settings Distinctive Ringing and Distinctive Call Waiting Tones can be associated with specific callers configured directly into the PHONE ADPATER by setting the appropr
91. eam 4B gt 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 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 both 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 gt User ID or Phone Number 16B 2004 Linksys Proprietary See Copyright Notice on Page 2 54 S Expiration Date 12B S Public Key 512b or 64B S Signature 1024b or 512B The signing agent is implicit and must be the same for all PHONE ADAPTER s that intended to communicate securely with each other The public key of the signing agent is pre configured into the PHONE ADAPTER s by the administrator and will be used by the PHONE ADAPTER to verify the Mini Certificate of its peer The Mini Cert
92. ediately 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 TimeO 30 sec 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 PHONE ADAPTER 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 TimeO 10 dialing Range 0 64 sec Interdigit Short Timer Short timeout between entering digits when TimeO 3 dialing Range 0 64 sec 2004 Linksys Proprietary See Copyright Notice on Page 2 75 CPC Delay Delay in seconds after caller hangs up when 2 the PHONE ADAPTER will start removing the tip and ring voltage to the attached equipment of the called party Range 0 to 255 s Resolution 1 s CPC Duration Duration in seconds for which the tip to ring 0 CPC voltage is removed after the caller hangs up disable After that tip to ring voltage is restored and dial d tone will apply if the attached equipment is still off hook CPC is disabled if this value is set to 0 Range 0 to 1 000 s Resolution 0 001 s Notes 1 The Call Progress Tones and DT
93. ee Figure 1 a c All PHONE ADAPTER CID methods can be applied for this type of caller id e On Hook Caller ID Not Associated with Ringing In the PHONE ADAPTER this feature is used for send VMWI signal to the phone to turn the message waiting light on and off see Figure 1 d and e This is available only for FSK based caller id methods Bellcore ETSI FSK and ETSI FSK With PR e Off Hook Caller ID This is used to delivery caller id on incoming calls when the attached phone is off hook see Figure 1 f This can be call waiting caller ID CIDCW or to notify the user that the far end party identity has changed or updated such as due to a call transfer This is only available if the caller id method is one of Bellcore ETSI FSK or ETSI FSK With PR 2004 Linksys Proprietary See Copyright Notice on Page 2 99 a Bellcore ETSI Onhook Post Ring FSK First gt Ring py tk b ETSI Onhook Post Ring DTMF First gt Ring gt DTMF c ETSI Onhook Pre Ring FSK DTMF Polarity CAS DTMF First Reversal gt DTAS Pi Fsk P Ring d Bellcore Onhook FSK w o Ring p OST p FSK CAS DTAS p FSK f Bellcore ETSI Offhook FSK CAS Wait For gt ras ack rb FSK PHONE ADAPTER Caller ID Delivery Architecture 7 2 24 Streaming Audio
94. empts to fetch a URL succeeds if the profile is received and parsed correctly If the Resync_Fails_On_FNF parameter is set to No a profile rule will also succeed if an attempted fetch for a URL returns a File Not Found error message A profile rule with only assignments always succeeds Optional qualifiers can be specified in brackets preceding each URL To ease testing and development the script syntax also supports using as a comment delimiter until end of parameter This allows a potentially long script to be temporarily commented out The syntax for the rule is as follows with standard conventions for URLs cule term zt teim term condition assignments options url eg em UI eg iesst WV WN conditionseq condelem conjunction condelem condelem numcond vercond strcond numcond number relop number vercond version relop version relop Wen lt I wo gt wow WE D D Es D ge A bel LD Lei A eq D Jos version major 4 miao Ha W ge 4 menner Sa Sieg Ol Canal GeO Mag line melo Mme hicum strcond gett egop gstr per Zem Vama asri DO wall DDT eme Waan WE Z WA Z He ii ne ASSaleimmeantes WC sescaeponmneyoe 19 UC assignment attribute expr 2004 Linksys Proprietary See Copyright Notice on Page 2 38 expr DQUOT val DQUOT EE eme WII eem 1713 option key opt alias opt post opt key opt
95. en 404 Not Found 405 Method Not Allowed 406 Not Acceptable 407 Proxy Authentication Required 408 Request Timeout 2004 Linksys Proprietary See Copyright Notice on Page 2 94 410 Gone 412 Conditional Request Failed 413 Request Entity Too Large 414 Request URI Too Long 415 Unsupported Media Type 416 Unsupported URI Scheme 420 Bad Extension 421 Extension Required 423 Interval Too Brief 429 Provide Referrer Identity 480 Temporarily Unavailable 481 Call Transaction Does Not Exist 482 Loop Detected 483 Too Many Hops 484 Address Incomplete 485 Ambiguous 486 Busy Here 487 Request Terminated 488 Not Acceptable Here 489 Bad Event 491 Request Pending 493 Undecipherable 494 Security Agreement Required Server Failure 5xx 500 Server Internal Error 501 Not Implemented 502 Bad Gateway 503 Service Unavailable 504 Server Time out 505 Version Not Supported 513 Message Too Large 580 Precondition Failure Global Failures 6xx 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere 606 Not Acceptable 7 Summary of Implemented Features and Specifications The PHONE ADAPTER is a full featured fully programmable phone adapter that can be custom provisioned within a wide range of configuration parameters The below feature descriptions are written as a high level overview to provide a basic understanding of the feature breadth and capabilities of the PHONE ADAPTER To understand the specific implementation of the below fea
96. er ID when making an outgoing call To activate the service the subscriber enters the corresponding or code prior to making the call This service is in effect only for the duration of the current call Calling Line Identification Restriction CLIR Caller ID Blocking The subscriber activates this service to hide his Caller ID when making an outgoing call To activate the service the subscriber enters the corresponding or code prior to making the call This service is in effect only for the duration of the current call 1 1 4 2 2 Call Waiting The subscriber can accept a call from a 3rd party while engaging in an active call The PHONE ADAPTER shall alert the subscriber for the 2nd incoming call by playing a call waiting tone Disable or Cancel Call Waiting By setting the corresponding configuration parameter on the PHONE ADAPTER the PHONE ADAPTER supports disabling of call waiting permanently or on a per call basis Call Waiting with Caller ID In between call waiting tone bursts the PHONE ADAPTER can generate a Caller ID signal to the attached phone when it is off hook 2004 Linksys Proprietary See Copyright Notice on Page 2 12 1 1 4 2 3 Voice Mail Message Waiting Indication Service Providers may provide voice mail service to their subscribers When voice mail is available for a subscriber a notification message will be sent from the Voice Mail server to the PHONE ADAPTER The PHONE ADAPTER indicates that a me
97. eries of elements which are individually matched to the keys pressed by the user Elements can be one of the following e Individual keys 0 1 2 9 e The letter x matches any one numeric digit 0 9 e A subset of keys within brackets allows ranges I set T e g 389 means 3 or 8 or 9 o Numeric ranges are allowed within the brackets digit digit e g 2 9 means 2 or 3 or or 9 o Ranges can be combined with other keys e g 235 8 means 2 or 3 or 5 or 6 or 7 ox or Dor 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 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 bar
98. es 21 3 1 1 Using the Supplemental Profile Compiler A 23 3 1 2 Encrypting and Compressing XML configuration files ecceeeeeseeeeeeeeeeeeeeeeeeeseaeeeeeeessaeeeeeeerias 24 3 2 Secure Initial CONFIQUIATION 0 0 2 c ceeeeceeeeeceteeeceaeeeenee seat eeeeaaeceaeesaeeesaaeeseaeeseeeetsaeeesaeeeenees 25 3 3 Web INterntace rroen adira andesite cates ha Gaar aAa a Aant aia chased Wag AAE Vaa AALA sees 26 3 3 1 Web Interface Conventions A 26 3 3 2 c Administration Gi Vue EE 27 3 3 3 Basic and Advanced VIEWS cccecsecereseeeceseseeneeeeseeeeseseneeeseseeenessaeeeseneesenseseneesesensensnensenenseesenes 27 3 4 Functional Configuration UL e 27 3AT Upgrade URE sid cakes ate tec i nen ee Hale ei ne sa 27 242 Resyne URL wives ioNierk e ENKEN dae ceeeeet cneesces vb evnie scene E 28 S Reboot RE gies Eeer ee dag te ile Sle eel 28 3 5 Configuration via the IVR PAP2 only cesceseececeeeeeeeaeeeeaeeseeeeeceaaeseeaeeseeeescaeeesaeenenees 29 4 Configuration Parameters ccccccccceesseeceeeeceeeeeceaeeeeaeeeceaeeeeaaeeeeaeeseaeeesaaeeseaaeseeeeesnaeeesiaeeeeeneesaas 32 4 1 Data Type Sinait a a ha A ee ee 32 AM ee 35 4 2 Provisioning Related Parameters 35 4 2 1 Firmware Upgrade tni sein eee Ae EE i ie e ee 43 4 2 2 Provisioning Server Redundancy s sssesseeeeeeieettetneetntetstttsittttnstinsttnntnnstntetnttenttnnntnaeenntenntnnnna 46 4 2 3 Configuring the Web Server and WD 46 System COntigurati
99. escription Type Default Preferred Codec Select a preferred codec for all calls However the Choice G711u actual codec used in a call still depends on the outcome of the codec negotiation protocol G711u G711a G726 16 G726 24 G726 32 G726 40 G729a G723 Use Pref Codec Only Only use the preferred codec for all calls The call will Bool No fail if the far end does not support this codec Silence Supp Enable Enable silence suppression so that silent audio Bool No frames are not transmitted Echo Canc Enable Enable the use of echo canceller Bool Yes Echo Canc Adapt Enable echo canceller to adapt Bool Yes Enable Echo Supp Enable Enable the use of echo suppressor If lt Echo Canc Bool Yes Enable gt is no this parameter is ignored 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 FAX Passthru Enable This parameter has been removed Bool Yes 2004 Linksys Proprietary See Copyright Notice on Page 2 52 FAX CED Detect Enable Enable detection of FAX tone Bool Yes FAX CNG Detect Bool Yes Enable FAX Passthru Codec Codec to use for fax passthru G711u G
100. esired service behavior To understand the specific implementation options of the below features including parameters requirements and contingencies please refer the section Configuration Parameters section Error Reference source not found 5 1 Originating a Phone Call Service Description Placing telephone a call to another telephone or telephony system IVR conference bridge etc This is the most basic service User Action Required to Activate or Use When the user picks up the handset the PHONE ADAPTER provides dial tone and is ready to collect dialing information via DTMF digits from the telephone Touchtone key pad Expected Call and Network Behavior While it is possible to support overlapped dialing within the context of SIP the PHONE ADAPTER collects a complete phone number and sends the full number in a SIP INVITE message to the proxy server for further call processing In order to minimize dialing delay the PHONE ADAPTER maintains a dial plan and matches it against the cumulative number entered by the user The PHONE ADAPTER also detects invalid phone numbers not compatible with the dial plan and alerts the user via a configurable tone Reorder or announcement User Action Required to Deactivate or End Hang up the telephone 5 2 Receiving a Phone Call Service Description User Action Required to Activate or Use The PHONE ADAPTER can receive calls from the PSTN or other IP Tele
101. ess user level and admin level to configuration parameters For standalone PHONE ADAPTERS which contain no router or NAT functionality an IVR Interactive Voice Response interface is also available for configuring basic networking 3 1 Configuration Profile Formats The PHONE ADAPTER configuration profile is an XML or binary file with encoded PHONE ADAPTER parameter values and optionally user access permissions for those parameters By convention the profile is named with the extension cfg e g pap2 cfg An administrator can easily generate the XML format and compress and or encrypt this file with off the shelf tools e g gzip openssl The XML configuration file always begins with the top level element lt flat profile gt Within this element are any number of the configuration elements which are visible in the GUI The XML tag names are case sensitive and are identical to the names in the GUI except that characters other than hyphen period underscore and alphanumeric characters from the GUI are replaced with an underscore in the XML names For example User ID 1 becomes lt User_ID_1_ gt Empty elements ex lt element gt or missing elements do not change the value already stored in memory An opening and closing tag ex lt element gt lt element gt with no included value deletes the value stored in memory Standard XML comments and arbitrary whitespace can be included in the file for readability purposes Note
102. f hook it will remove battery from the line since no audio session is in progress e Set up the Proxy and Subscriber Information for the SAS Line as you normally would with a regular user account e Call Forwarding Call Screening Call Blocking DND and Caller ID Delivery features are not available on an SAS line 4 10 3 Per Line Polarity Settings Parameter Name Description Type Default Idle Polarity Polarity before call connected Forward Reverse Forward Caller Conn Polarity Polarity after outbound call connected Forward Reverse Reverse Callee Conn Polarity Polarity after inbound call connected Forward Reverse Reverse 4 10 4 Additional Timer Values sec Parameter Name Description Type Default Hook Flash Timer Min Minimum on hook time before off hook to Time3 0 1 qualify as hook flash Less than this the on hook event is ignored Range 0 1 0 4 sec Hook Flash Timer Max Maximum on hook time before off hook to Time3 0 9 qualify as hook flash More than this the on hook event is treated as on hook no hook flash event Range 0 4 1 6 sec Callee On Hook Delay The phone must be on hook for at this time in Time 0 sec before the PHONE ADAPTER will tear down the current inbound call It does not apply to outbound calls Range 0 255 sec Reorder Delay Delay after far end hangs up before reorder Time 5 tone is played 0 plays imm
103. f internally in the PHONE ADAPTER The Feature_Dial_Services_Codes and Referral_Services_Codes parameters contain a list of dial strings that correspond to feature codes in the network after which the PHONE ADAPTER needs to collect a target number These codes are automatically appended to the dial plan so there is no need to explicitly include them in the dial plan For example if call forwarding is implemented in the network the code to activate call forwarding and collect the target number should be included in the Feature_Dial_Services_Codes parameter but the code to deactivate call forwarding should not since it does not require collection of a target phone number Feature Dial Services Codes 2004 Linksys Proprietary See Copyright Notice on Page 2 60 One or more code can be configured into this parameter such as 72 or 72 74 67 82 etc Max total length is 79 chars This parameter applies when the user has a dial tone 1st or 2nd dial tone Enter code and the following target number according to current dial plan entered at the dial tone triggers the PHONE ADAPTER to call the target number prepended by the code For example after user dials 72 the PHONE ADAPTER plays a prompt tone awaiting the user to enter a valid target number When a complete number is entered the PHONE ADAPTER 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
104. ference Calling Service Description This feature allows the user to conference up to two other numbers on the same line to create a three way call User Action Required to Activate or Use If you are already on a call and wish to add a third party Press the switch hook or flash button Listen for dial tone Dial the third party normally When the third party number starts to ring press the switch hook or flash button again You now have the original caller and the third party together with you on the same call If you want to initiate a new Three Way Call 2004 Linksys Proprietary See Copyright Notice on Page 2 86 Call the first party in the normal manner Follow the directions for adding a third party see instructions above Expected Call and Network Behavior The PHONE ADAPTER can host a 3 way conference and perform 3 way audio mixing without the need of an external conference bridge device or service If you also have Call Transfer you can also hang up at any time to transfer the original caller to the third party User Action Required to Deactivate or End 5 15 Call Return Service Description The PHONE ADAPTER supports a service that allows the PHONE ADAPTER to automatically dial the last caller s number User Action Required to Activate or Use Pick up the receiver Listen for dial tone Press __ to dial back the last caller that tried to reach you
105. 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 of a mapping from any external 2004 Linksys Proprietary See Copyright Notice on Page 2 15 address port to the corresponding private source address port These characteristics of a NAT can be exploited by an PHONE ADAPTER to let external entities send SIP messages and RTP packets to it when it is installed on a private network 1322 VoIP NAT Interworking In the case of SIP the addresses where messages data should be sent to an PHONE ADAPTER are embedded in the SIP messages sent by the device If the PHONE ADAPTER 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 PHONE ADAPTER must substitute the private IP address information with the proper external IP address port
106. ful and will not retry to upgrade unless some event triggers a reboot Upgrade Rule ParName Upgrade_Rule and Upgrade_Rule_B Default Empty The Upgrade_Rule and Upgrade_Rule_B parameters are scripts that identifies the upgrade server to contact during a firmware upgrade Upgrade_Rule_B is only executed if Upgrade_Rule executed successfully These strings support one level of macro expansion using a small set of variables Following macro substitution the rule is evaluated to obtain a URL of the firmware file to request from an upgrade server The URL can be partially specified in which case default values are assumed for the unspecified terms The filepath portion of the URL must be specified The Upgrade_Rule supports additional syntax that allows the URL to be a function of the firmware release currently running in the PHONE ADAPTER This mechanism can aid service providers sequence through a firmware upgrade by allowing them to automatically stage the upgrade sequence if so required by the firmware Also the Downgrade_Rev_Limit parameter can contain a version string below which the PHONE ADAPTER will not downgrade The conditional syntax consists of a sequence of condition url pairs separated by the character The condition component tests the current firmware version number against a specified value The sequence of conditions is evaluated until one is satisfied The URL associated with that condition i
107. ful resync SSCHEME SSERVIP SPORTSPATH Log Resync Failure Msg SPN MAC Resync failed ERR Firmware Upgrade Upgrade_Enable Yes j Upgrade_Error Retry Delay 3600 Downgrade Bev Limit V Upgrade_Rule H E Log Upgrade Request Msg SPN SMAC Requesting upgrade SSCHEME SSERVIP SPORTSPATH Log Upgrade Success Msg SPN SMAC Successful upgrade SSCHEME SSERVIP SPORTSPATH SERR Log Upgrade Failure Msg SPN SMAC Upgrade failed ERR General Purpose Parameters GPP A wu GPP B wu GPP E wu GPP D wu GPP E wu GPP F wu 2004 Linksys Proprietary See Copyright Notice on Page 2 103 GPP o GPP_H GPP I GPP J GPP K GPP L GPP M GPP N GPP O GPP P GPP SA GPP SB GPP SC GPP SD SIP Parameters Max Forward Max Redirection Max Auth SIP_User_ Agent Name SIP Server Name SIP _Accept_Language DTMF Relay MIME Type Hook_Flash MIME Type Remove_Last_Reg Use_Compact_Header STP Timer Values sec SIP T1 SIP T2 SIP T4 SIP Timer RB SIP Timer F SIP Timer H SIP Timer D SIP Timer J INVITE Expires ReINVITE_ Expires Reg Min Expires Reg Max Expires Reg Retry Intvl Reg Retry Long Intvl wu wu wu wu wu wu wu wu wu wu wu wu wu wu 70 wow 2 SVERSION SVERSION wu application dtmf relay application hook flash W No No 25 g wou 32 32 32 32 32 240 s 30 q 7200
108. ging voltage 60 90 V Uns8 70 CWT Frequency Frequency script of the call waiting tone All FreqScript 440 10 distinctive CWT is based on this tone 4 9 2 Progress Tones Most of the 18 progress tones in the PHONE ADAPTER are played automatically in response to fixed stimuli However the administrator can select which SIP response codes correspond to the 4 SIT tones Response Status Code Handling SIT1 RSC SIP response status code to INVITE on which RscTmplt to play the SIT1 Tone SIT2 RSC SIP response status code to INVITE on which RscTmplt to play the SIT2 Tone SIT3 RSC SIP response status code to INVITE on which RscTmplt to play the SIT3 Tone SIT4 RSC SIP response status code to INVITE on which RscTmplt to play the SIT4 Tone The Frequencies of the actual progress tones are configurable to accommodate local and regional conventions end is ringing 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 ToneScript 420 19 520 tries to dial a 3 way call 19 10 0 1 2 Outside Dial Tone An alternative to lt Dial Tone gt usually used ToneScript 420 16 10 0 1 to prompt the user to enter an external phone number versus an internal extension This is t
109. git sequences As more digits are entered by the user the set of candidates diminishes until only one or none are valid 2004 Linksys Proprietary See Copyright Notice on Page 2 62 Any one of a set of terminating events triggers the PHONE ADAPTER to either accept the user dialed sequence and transmit it to initiate a call or else reject it as invalid The terminating events are e No candidate 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 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 White space is ignored and may be used for readability Digit Sequence Syntax Each digit sequence within the dial plan consists of a s
110. he 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 Comfort Noise Generation provides artificially generated background white noise sounds designed to reassure callers that their calls are still connected during silent periods H Comfort Noise Generation is not used the caller may think the call has been disconnected because of the dead silence periods created by the VAD and Silence Suppression feature 7 2 16 Attenuation Gain Adjustment 7 2 17 Signaling Hook Flash Event 2004 Linksys Proprietary See Copyright Notice on Page 2 98 The PHONE ADAPTER can signal hook flash events to the remote party on a connected call This feature can be used to provide advanced mid call services with third party call control Depending on the features that the service provider will offer using third party call control the following three PHONE ADAPTER features may be disabled to correctly signal a hook flash event to the softswitch 1 Call Waiting Service 2 Three Way Call Service 3 Three Way Conf Service 7 2 18 Configurable Flash Switch Hook Timer 7 2 19 Configurable Dial Plan with Interdigit Timers The PHONE ADAPTER has three configurable interdigit timers e Initial timeout T
111. he 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 VolP telephone system that records information about the telephone number s activity Carriers 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 capital investment or maintenance charges The LEC charges a monthly fee to the customer who must agree to sign a term agreement 2004 Linksys Proprietary See Copyright Notice on Page 2 115 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
112. hficattons tetter rest tn rnst nnnnnerrn nnne 95 TA Data Networking Features 95 FAA MAC Address IEEE GO 21 EEN aeae aii Se iee iei itokane aie iiaee tekidir 95 7 1 2 IPv4 Internet Protocol Version A RFC 791 upgradeable to v6 RFC 19082 96 7 1 3 ARP Address Resolution Protocols aene a eean n r i a ag 96 7 1 4 DNS A Record RFC 1706 SRV Record RFC 20921 96 7 1 5 DiffServ RFC 2475 and ToS Type of Service RFC 01 12401 96 7 1 6 DHCP Client Dynamic Host Configuration Protocol RFC 21231 96 7 1 7 ICMP Internet Control Message Protocol DEC 202 96 7 1 8 TCP Transmission Control Protocol DEC oO 96 7 1 9 UDP User Datagram Protocol HEC ZG8 96 7 1 10 RTP Real Time Protocol RFC 1889 RFC 1900 96 7 1 11 RTCP Real Time Control Protocol RFC 18901 96 7 2 Volite EE 96 7 2 1 SIPv2 Session Initiation Protocol Version 2 RFC 2261 226b 96 7 2 1 1 SIP Proxy Redundancy Static or Dynamic via DNS SPV A 96 7 2 1 2 Re registration with Primary SIP Proxy Server ccceeccesseeeeeeeeeeeeeeneeeeeeseaeeseeeseeeseeeeeaees 96 2004 Linksys Proprietary See Copyright Notice on Page 2 4 7 2 1 3 SIP Support in Network Address Translation Networks NAT 96 7 2 2 Codec Name Assignments eenei ebi e iei iieri raires ae aioa dressers aitas 96 72 3 Secure Calg iienan e ena e Menta eis aetna tae nd einen ie 97 LS E e ee le ln EEN 97 7 2 4 1 G711 A laW and Bau EE 9
113. his check after receiving the callee s Mini Certificate Service Provider Requirements The PHONE ADAPTER 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 PHONE ADAPTER to verify the MC received from the other end If the MC is invalid the PHONE ADAPTER 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 SRTP Private Key gt parameter which should be hidden from the PHONE ADAPTER 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 fi
114. hone and has Call Waiting active the new callers Caller ID information will be displayed on the users phone display screen at the same time the user is hearing the Call Waiting beeps tones User Action Required to Activate or Use The telephone equipment connected to the PHONE ADAPTER must support Call Waiting with Caller ID Expected Call and Network Behavior In between call waiting tone bursts the PHONE ADAPTER can generate a Caller ID signal to the attached phone when it is off hook User Action Required to Deactivate or End Not applicable 5 9 Voice Mail Service Description Service Providers may provide voice mail service to their subscribers Users have the ability to retrieve voice mail via the telephone connected to the PHONE ADAPTER User Action Required to Activate or Use The PHONE ADAPTER indicates that a message is waiting by playing stuttered dial tone when the user picks up the handset To retrieve messages Lift the receiver Listen for dial tone Dial the phone number assigned to the PHONE ADAPTER You will be connected to the voice mail server and prompted by a voice response system with 2004 Linksys Proprietary See Copyright Notice on Page 2 83 instructions to listen to your messages Expected Call and Network Behavior User Action Required to Deactivate or End When voice mail is available for a subscriber a notification mes
115. iate callers in the Ring_n_Caller parameters The Ring_1_Caller parameter specifies which callers will trigger ring cadence 1 and so forth If a provider wishes to offer a distinctive ringing service by providing hints from the network the provider can insert an Alert Info SIP header into incoming calls If the value in the Alert Info header matches one of the strings in the Ring_n_Name set of parameters the corresponding ring cadence will be used In addition to ordinary and distinctive rings there are number of other situations where the PHONE ADAPTER can provide a short burst of ringing These ring settings are described below Parameter Name Description Type Default Ring 1 Caller Caller number pattern to play Distinctive Ring CWT 1 PhTmplt Ring 2 Caller Caller number pattern to play Distinctive Ring CWT 2 PhTmplt Ring 3 Caller Caller number pattern to play Distinctive Ring CWT 3 PhTmplt Ring 4 Caller Caller number pattern to play Distinctive Ring CWT 4 PhTmplt Ring 5 Caller Caller number pattern to play Distinctive Ring CWT 5 PhTmplt Ring 6 Caller Caller number pattern to play Distinctive Ring CWT 6 PhTmplt Ring 7 Caller Caller number pattern to play Distinctive Ring CWT 7 PhTmplt Ring 8 Caller Caller number pattern to play Distinctive Ring CWT 8 PhTmplt Default Ring Default ringing pattern 1 8 for all callers 1 2 3 4 1 5 6 7 8 Default CWT Default CWT
116. icit resync command from the web interface discussed earlier in the Function URLs section of this document Resync on Reset ParName Resync_On_Reset Default Enable Resync_On_Reset determines whether the PHONE ADAPTER will attempt to resync with the provisioning server on power up and following explicit reboot requests Resync Random Delay ParName Resync_Random_Delay Default 2 Resync_Random_Delay helps to scatter resync requests from multiple devices uniformly over a period of time whose duration in seconds is indicated by this parameter Hence if a number of PHONE ADAPTER devices were to power up at the same time their resync requests would be distributed over time lessening the impact on the provisioning servers Resync Periodic ParName Resync_Periodic Default 3600 The PHONE ADAPTER attempts to resync with the provisioning server periodically provided the Resync_Periodic parameter is configured with a non zero value The value in seconds indicates the interval between resync attempts Normally the PHONE ADAPTER will not start the resync while an 2004 Linksys Proprietary See Copyright Notice on Page 2 36 active call is in progress The PHONE ADAPTER will wait up to Forced_Update_Delay seconds for both lines to become idle If the adapter still is not idle the adapter will perform a resync anyway Resync Error Retry Delay ParName Resync_Error_Re
117. ietary See Copyright Notice on Page 2 64 IL SOK 2 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 distance 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 8 gt i soo soomomwx hi o ee 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 Sea Eu Willi seen 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 C O BHL T 2 9 Z ss The following allows US style long distance but blocks 9xx area codes i 2 Slsen 2 9 seas The following allows arbitrary long distance dialing but explicitly blocks the 947 area code i i eem 1 eme eme The following implements a Hot Line phone which automatically calls 1 212 5551234
118. ificate is valid if a it has not expired and b its signature checks out User Interface The PHONE ADAPTER 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 will be updated with the Name and Number extracted from the Mini Certificate sent by the other party provided CIDCW service and equipment are available the CID Name in this case will have a sign inserted at the beginning 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 PHONE ADAPTER 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 PHONE ADAPTER will perform t
119. ilable if the caller id method is one of Bellcore ETSI FSK or ETSI FSK With PR 2004 Linksys Proprietary See Copyright Notice on Page 2 77 a Bellcore ETSI Onhook Post Ring FSK First gt Ring Tak b ETSI Onhook Post Ring DTMF First gt Ring DTMF H c ETSI Onhook Pre Ring FSK DTMF Polarity CAS DTMF First DTAS FSK gt Ring d Bellcore Onhook FSK w o Ring OSI FSK e ETSI Onhook FSK w o Ring CAS DTAS CAS Wait For gt DTAS ACK ESK Figure PHONE ADAPTER Caller ID Delivery Architecture FSK f Bellcore ETSI Offhook FSK 2004 Linksys Proprietary See Copyright Notice on Page 2 5 Expected Feature Behavior The PHONE ADAPTER can be configured to the custom requirements of the service provider so that from the subscriber s point of view the service behaves exactly as the service provider wishes with varying degrees of control left with the end user This means that a service provider can leverage the programmability of the PHONE ADAPTER to offer sometimes subtle yet continually valuable and differentiated services optimized for the network environment or target market s This section of the Administration Guide describes how some of the supported basic and enhanced or supplementary services could be implemented The implementations described below by no means are the only way to achieve the d
120. ile waiting for a response 2004 Linksys Proprietary See Copyright Notice on Page 2 93 X40 General SIP Protocol Error e g unacceptable codec in SDP in 200 and ACK messages or times out while waiting for ACK X60 Dialed number invalid according to given dial plan 6 5 Provisioning and Upgrade result codes The PRVST and UPGST macro variables expand to integer codes which report the state of a resync or upgrade attempt They are typically used within triggers and resync upgrade conditions The values of these variables is as follows 1 explicit request resync upgrade url or sip 0 just rebooted resync only 1 triggered from configured trigger or rule 2 error retry 6 6 Table of SIP Response Codes Error Codes For convenience below is a list of SIP error codes at the time of this printing which incorporates response codes from the IANA Internet Assigned Numbers Authority SIP parameter registry http www iana org assignments sip parameters and additional response codes defined in Internet drafts which are implemented by the PHONE ADAPTER Provisional 1xx 100 Trying 180 Ringing 181 Call Is Being Forwarded 182 Queued 183 Session Progress Successful 2xx 200 OK 202 Accepted Redirection 3xx 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 305 Use Proxy 380 Alternative Service Request Failure 4xx 400 Bad Request 401 Unauthorized 402 Payment Required 403 Forbidd
121. in the PHONE ADAPTER configuration profile 2004 Linksys Proprietary See Copyright Notice on Page 2 10 e The SIP signaling messages The SIP messages exchanged between the SIP proxy server and the PHONE ADAPTER should be encrypted with a secret key This can be achieved for instance by transporting SIP over TLS e RTP packets The RTP payload exchanged between SIP user agents can be encrypted with a secret key to protect against eavesdropper The secret key can be negotiated with proper SIP signaling messages Hence the signaling path must be secured also 1 1 3 1 Proxy Servers Proxy servers handle two functions 1 Accept registrations from the SIP user agents 2 Proxy requests and responses between user agents Registration is the process by which a user agent tells the proxy who it is and at what IP address and port that it can be reached via SIP Registration usually expires within a finite period e g 60s or 3600s and the UA shall renew their registration periodically before the last registration expires When a user agent initiates a call it sends a SIP INVITE request to the proxy server and indicates the target recipient of the call The proxy server then consults a database to determine where to forward the request to the destination user agent The proxy server can request authentication credentials from the user agent before granting the service The credentials are computed by the user agent based on a pre provisio
122. in the mapping chosen by the underlying NAT to communicate with a particular public peer address port For this the PHONE ADAPTER needs to perform the following tasks e 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 PHONE ADAPTER can send such a request when it first attempts to communicate with a SIP 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 PHONE ADAPTER continues to send periodic keep alive packet
123. ion on endpoint equipment then it is very likely that the softswitch chosen for network operations will support multiple PSTN and VoIP signaling protocols The table below lists the most commonly accepted de facto standards used when implementing a VoIP signaling scheme based on the type of gateway or endpoint equipment being deployed VoIP Equipment Type Typical Port Density De Facto Signaling Standards Trunking Gateways Greater Than 500 Ports H 248 Megaco MGCP IPDC Access Gateways Between five and 500 Ports SIP H 323 PBX KTS Platforms Between ten and 500 Ports SIP H 323 SCCP PBX KTS Telephone Sets One Port SIP MGCP SCCP Phone Adapters and IP Centrex Up to four Ports SIP MGCP Phones The PHONE ADAPTER supports SIP today It has the capability to communicate with a variety of endpoints and signaling entities via SIP messages 1 2 Network Address Translation NAT Traversal 1 2 1 What is a NAT or NAPT Network Address Port Translator 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
124. ion on firmware upgrade The syntax of Upgrade URL is 2004 Linksys Proprietary See Copyright Notice on Page 2 27 http lt PAP2 ip addr gt admin upgrade protocol server namef port firmware pathname If no protocol is specified TFTP is assumed Note Only TFTP is supported in the current release If no server name is specified the host that requests the URL is used as server name If no port specified default port of the protocol is used 69 for TFTP 80 for http 443 for HTTPS The firmware pathname is typically the file name of the PHONE ADAPTER binary located in the root directory of the TFTP server If no firmware pathname is specified Phone Adapter bin is assumed For example http 192 168 2 217 upgrade tftp 192 168 2 251 PAP2 bin 3 4 2 Resync URL Through Resync URL you can force the PHONE ADAPTER to do a resync to a profile specified in the URL Note The PHONE ADAPTER will resync only when it is idle The syntax of Resync URL is http lt Phone Adapter ip addr gt resync protocol server name port profile pathname If no parameter follows resync the profile rule setting in provisioning is used See 4 2 for detailed information about profile rule in provisioning If no protocol is specified TFTP protocol is assumed Note Only TFTP is supported in the current release If no server name is specified the host that requests the URL is used as server name If n
125. is document Firmware upgrades are attempted only when the PHONE ADAPTER is idle since they trigger a software reboot Firmware upgrades are controlled by the following parameters which operate in a manner similar to but independent of the provisioning parameters Upgrade_Enable Upgrade_Rule Upgrade Enable Upgrade_Error_Retry_Delay Downgrade_Rev_Limit Log_Upgrade_Request_Msg Log_Upgrade_Success_Msg Log_Upgrade_Failure_Msg ParName Upgrade_Enable Default Enable 2004 Linksys Proprietary See Copyright Notice on Page 2 43 The firmware file must be requested by the PHONE ADAPTER and cannot be pushed from an upgrade server although a service provider can effectively push a new firmware load by triggering the request operation remotely via the CFG file The functionality is controlled by the Upgrade_Enable parameter The parameter enables the functionality encompassed by the remaining upgrade parameters In addition Upgrade_Enable also gates the ability to issue an explicit upgrade command from the web interface discussed in section 3 4 1 of this document Upgrade Error Retry Delay ParName Upgrade_Error_Retry_Delay Default 3600 If an upgrade attempt fails the PHONE ADAPTER will retry with a delay indicated by the Upgrade_Error_Retry_Delay parameter specified in seconds If the value is zero the PHONE ADAPTER treats upgrade failures as though they were success
126. ited by a character up to the end of line Blank lines can be used for readability Parameter name II J quoted_parameter_value_string ici Example of plain text file entries These parameters are for illustration only Feature_Enable Ligen 5 user read writ Another Parameter B SSE OO s user read only Hidden_Parameter SIDES A user not accessibl Some_Entry Ls user read write leave value unchanged Multiple plain text files can be spliced together to generate the source for each CFG file This is accomplished by the import directive the literal string import placed at the start of a new line followed by one or more spaces and the file name to splice into the stream of parameter value pairs The following example illustrates File splicing can be nested several files deep base txt contains Paraml base value 1 Param2 base value 2 Phone Adapter1234 txt contains import base txt Paraml new value overrides base Param particular value 7 p 2004 Linksys Proprietary See Copyright Notice on Page 2 22 The Phone Adapterl234 txt file above is equivalent to Parami base value 1 Param2 base value 2 Paraml new value overrides base Baam Tea iess valve 7 p A sample plain text file containing default parameter value and access settings for the
127. ity 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 7 2 1 3 SIP Support in Network Address Translation Networks NAT 7 2 2 Codec Name Assignment 2004 Linksys Proprietary See Copyright Notice on Page 2 96 Negotiation of the optimal voice codec is sometimes dependent on the PHONE ADAPTER device s ability to match a codec name with the far end device gateway codec name The PHONE ADAPTER allows the network administrator to individually name the various codecs that are supported such that the correct codec successfully negotiates with the far end the equipment 7 2 3 Secure Calls A user if enabled by service provider or administrator has the option to make an outbound call secure in the sense that the audio packets in both directions are encrypted 7 2 4 Voice Algorithms 7 2 4 1 G 711 A law and mu law This very low complexity 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 7 2 4 2 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 pe
128. ively the user can activate this feature from a web browser interface Note The forward delay is entered from the web interface Default is 20s Expected Call and Network Behavior This feature allows a user the option to divert forward calls to their telephone number to any other dialable number when their phone is not answered by using the touchtone keypad of their telephone or web browser interface This service is activated or deactivated from the phone being forwarded or the web browser interface User Action Required to Deactivate or End Lift the receiver Listen for dial tone Drees You will hear a confirmation tone signaling your change has been accepted Alternatively the user can deactivate this feature from a web browser interface 5 20 Anonymous Call Blocking Service Description By setting the corresponding configuration parameter on the PHONE ADAPTER the subscriber has the option to block incoming calls that do not reveal the caller s Caller ID User Action Required to Activate or Use Pick up the receiver Listen for dial tone To Activate Press Expected Call and Network Behavior When activated by the user callers will hear busy tone User Action Required to Deactivate or End To De activate Press 5 21 Distinctive Priority Ringing and Call Waiting Tone Service Description The PHONE ADAPTER supports a number of
129. ization Vector IV and an 256 bit encryption key using the key phrase provided on the command line The TA recognizes the leading characters Salted__ as a hint to find the salt and decrypt the configuration file Linksys XML configuration files can be compressed using the gzip compression algorithm Gzip is available from http Awww gzip org gzip cleartextconfig xml If both compression and encryption are used the clear text version must be compressed before it is encrypted The PHONE ADAPTER does not recognize files which are encrypted and then compressed since encrypted files are uncompressible The Linksys PHONE ADAPTER automatically detects if a file is compressed or encrypted 3 2 Secure Initial Configuration Linksys recommends a secure configuration system to providers to protect them from theft of service account forgery and denial of service To that end Linksys Terminal Adapters are provisioned at the factory with a public key certificate signed by the Linksys certificate authority The first step in this process is for the Linksys terminal adapters to use HTTPS to initially contact the configuration server specified in the Profile Rule The initial URL can be configured into the TA at manufacturing time for order over a certain size it can be added during a staging process or it can be provided via the web interface as described in the next section The PHONE ADAPTER opens a TCP connection to the initial configuration server
130. knSjjjOy8c 1ITCd2t44MhOvmwNg4fDck2 YdmTMBR516xut4 uQ LJQIni2kwqlm7scDvll5 k232EvwVtCKOAYa4eWd6fQOpiESCO9CC9aY U1 X5 JUU EBZmi3AmcqE9U1LxEQOGwopaGyGOh3 VyhKgi6JaVtQZt87 PiJINKW8XQj3B9Qqe3VgYxWCQNa335Y CnDsenASeBxuMIEaBCYd11I1fVEodJZ OGwXwfAdeOMhcbDOkj7LVizcsTyk2TZYTccnZ75TuTjj13qvYs lt SRTP Private Key gt b DWc96X4Y QraCnYzl5en1 ClIUhVQQarvcr6Qd 8R52IEvJjOw e Kim4xiiF EPaKmU8UbooxKG36SEd Kusp0OAQ Mini Certificate Base64 encoded of Mini Certificate concatenated Str508 Empty with the 1024 bit public key of the CA signing the MC of all subscribers in the group SRTP Private Key Base64 encoded of the 512 bit private key per Str88 Empty subscriber for establishment of a secure call 4 6 5 Outbound Call Codec Selection Codes The User can use additional feature codes on the PHONE ADAPTER to force or prefer specific codecs These codes are automatically appended to the dial plan There is no need to include them explicitly in dial plan 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 ActCode_ 027110 codec that 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
131. l of redirection involving a random CFG file path and encryption key Hence each of the first stage CFG files above would point to a second stage CFG file with entries such as the following 2004 Linksys Proprietary See Copyright Notice on Page 2 40 Profile Rule key B ps global com profiles active S A pap2 cfg GPP A Dz3P2q9sVgx7LmWbvu GPP_B 83cle792bcb6a824c0d18 42 9bea52d8483f2a24b32d75bc965d05e38c163d5ef In practice the first provisioning stage which individualizes each PHONE ADAPTER into fetching a unique CFG file could be preconfigured during manufacturing For added security the second stage which introduces strong encryption may be performed in house prior to shipping an PHONE ADAPTER to each end user Release 2 0 supports SSL based key exchanges alleviating the need for this in house step while preserving strong security for the provisioning process A provisioning flow chart from the point of view of the PHONE ADAPTER endpoint is presented in a later section Log Resync Request Message ParName Log_Resync_Request_Msg Default SPN SMAC Requesting resync SCHEME SSERVIP SPORTS PATH The Log_Resync_Request_Msg is a script that defines the message sent to the configured Syslog server whenever the PHONE ADAPTER attempts to resync with the provisioning server The string supports one level of macro substitution with the same va
132. l_ Plan 1 and 2 Enable_IP_Dialing Other timers are configurable via parameters but do not directly pertain to the dial plan itself They are discussed elsewhere in this document Interdigit Long Timer ParName Interdigit_Long_Timer 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 Dal Plan parameter for an explanation of candidate digit sequences 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 1 and Dial Plan 2 ParName Dial_Plan 1 and Dial_Plan 2 Default xx 3469 11 O 00 lt 1408 gt 2 9 xxxxxx 1 2 9 xx 2 9 xxxxxx 011x The Dial_Plan parameters contain the actual dial plan scripts for each of lines 1 and 2 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 di
133. le 2 no MOH Server 2 1002 7 3 Security Features 7 3 1 Multiple Administration Layers Levels and Permissions 7 3 2 HTTP Digest Encrypted Authentication via MD5 RFC 1321 7 3 3 HTTPS with Client Certificate 7 4 Administration and Maintenance Features 7 4 1 Web Browser Administration and Configuration via Integral Web Server 7 4 2 Telephone Key Pad Configuration with Interactive Voice Prompts 7 4 3 Automated Provisioning amp Upgrade via TFTP HTTP and HTTPS 7 4 4 Periodic Notification of Upgrade Availability via NOTIFY or HTTP 7 4 5 Non Intrusive In Service Upgrades 7 4 6 Report Generation and Event Logging The PHONE ADAPTER 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 HTTP with digested authentication for instance The information may be organized as an XML page or HTML page 7 4 7 Syslog and Debug Server Records The PHONE ADAPTER 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 PHONE ADAPTER allows some settings to select which type of activity events should be logged for instance a debug level setting 8 List of all configuration parameters Below is a list of all the configuration parameters for this software version 2 0
134. le with the base64 encoded 1024 bit CA private public key pairs for signing verifying the MC such as 9CC9aYU1 X5 JUU EBZmi3AmcqE9U 1LxEOGwopaGyGOh3VyhkKgi6JaVtQZt87PiJINKW8XQj3B9Qq 2004 Linksys Proprietary See Copyright Notice on Page 2 55 e3VgYxWCQNa335Y CnDsenASeBxuMIEaBCYd111fVEodJZOGwXwfAde0MhcbDOkj7LVizcsTyk2TZ YTccnZ75TuTjji3qvYs 5nEtOrkCa84 mEwl3D9tSvVLyliwQ u Hd C8u5SNk7hsAUZaA9TqH8Iw0d IqSrsf6scsmundY5j7Z5m K5J9uBxSB8t8vamFGDOpF4zhNtbrVvIXKI9kmp4vph 1 C5jzO9gDfs3MF zjy YrVUFdM pxtDBxmM f GUfrpAuxXb7 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 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 PHONE ADAPTER For Example gen_mc ca_key Joe Smith 14085551234 00 00 00 1 1 34 Produces lt Mini Certificate gt Sm9llIFNtaXRoAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAXNDAANT U1 MTIZNAAAAAAAMDAWM DAWMDEwMTMO00OwWakde2vVMF3Rw4pPXL7IAglagMpbLSAG2 YISqt1 98Cp9rP xMGFfoPmDK Gx6JFtkQ5sxLcuwgxpxpxkeXvpZKIYlpsb28L4Rhg5qZA Gol hDFCmG6dffZ9SJhxES767GOJIS N8 QBLr0Auemot
135. ller id generation Choices GMT 12 00 GMT 11 00 GMT GMT 01 00 GMT 02 00 GMT 13 00 FXS Port Impedance Electrical impedance of the FXS port 600 600 900 600 2 16uF 900 2 16uF 270 750 150nF 220 820 120nF 220 820 115nF 370 620 310nF FXS Port Input Gain Input Gain in dB Valid values are 6 0 to dB 3 infinity Up to 3 decimal places FXS Port Output Gain Similar to lt FXS Port Input Gain gt but apply to dB 3 the output signal 2004 Linksys Proprietary See Copyright Notice on Page 2 76 DTMF Playback Level Local DTMF playback level in dBm up to 1 PwrLevel 10 0 decimal place DTMF Playback Length Local DTMF playback duration in ms Time3 d Detect ABCD Enable local detection of DTMF ABCD Bool Yes Playback ABCD Enable local playback of OOB DTMF ABCD Bool Yes Caller ID Method The following choices are available Choice Bellcore e Bellcore N Amer China CID CIDCW and VMWI FSK sent after 1st ring same as ETSI FSK sent after 1st ring no polarity reversal or DTAS e DTMF Finland Sweden CID only DTMF sent after polarity reversal and no DTAS and before 1st ring e DTMF Denmark CID only DTMF sent after polarity reversal and no DTAS and before 1st ring e ETSI DTMF CID only DTMF sent after DTAS and no polarity reversal and before 1st ring e ETSI DTMF With PR CID only DTMF sent after polarity reversal and DTAS and before 1st ri
136. me 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 7 2 13 Full Duplex Audio Full duplex is the ability to communicate in two directions simultaneously so that more than one person can speak at a time Half duplex means that only one person can talk at a time like a CB radio or walkie talkie which is unnatural in normal free flowing two way communications The PHONE ADAPTER supports full duplex audio 7 2 14 Echo Cancellation Up to 8 ms Echo Tail The PHONE ADAPTER supports hybrid line echo cancellation This feature uses the G 165 echo canceller to eliminate up to 8 ms of line echo This feature does not provide acoustic echo cancellation on endpoint devices that is an end user s speakerphone 7 2 15 Voice Activity Detection with Silence Suppression amp Comfort Noise Generation Voice Activity Detection VAD and Silence Suppression is a means of increasing 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 t
137. mplete number to be stored just as if you were going to dial it yourself Listen for Confirmation tone two short beeps Hang up or repeat the sequence Note To enter IP addresses a graphical user interface like a web browser must be used 2004 Linksys Proprietary See Copyright Notice on Page 2 91 Expected Call and Network Behavior Pick up the receiver Listen for dial tone Press single digit code assigned to the stored number 2 9 Press to signal dialing complete The number is automatically dialed normally User Action Required to Deactivate or End None 6 Troubleshooting 6 1 Call Statistics Reporting The following lists the statistics collected by the PHONE ADAPTER during normal operation These statistics are presented in the PHONE ADAPTER 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
138. mware running on the PHONE ADAPTER Check Primary DNS 160 None IVR will announce the Server Setting current setting in the Primary DNS field Set Primary DNS Server 161 Enter IP address Requires Password using numbers on the telephone key pad Use the star key when entering a decimal point Check PHONE 170 None IVR will announce the ADAPTER s Web Server port that the web server Port is listening on Default is 80 Enable Disable Web 7932 Enter 1 to enable Requires Password Server of PHONE Enter 0 to disable ADAPTER Manual Reboot of Unit 732668 None After you hear Option Successful hang up Unit will reboot automatically 2004 Linksys Proprietary See Copyright Notice on Page 2 31 User Factory Reset of Unit 877778 Enter 1 to confirm PHONE ADAPTER will WARNING Enter star to prompt for confirmation DEER sienio cancel operation After confirming you will User Changeable DEFAULT SETTINGS WILL BE hear Option Successful LOST Hang up Unit will reboot S and all User Changeable This might include network and configuration parameters service provider data will be reset to factory default values Factory Reset of Unit 73738 Enter 1 to confirm PHONE ADAPTER will WARNING ALL NON DEFAULT SETTINGS WILL BE LOST This includes network and Enter star to cancel operation prompt for confirmation After confirming you will hear Option S
139. n et GE Dawn Om Grease SA to SD some value Content mor GRP _SVA Cen II 210 Note that the UPGCOND term is particularly useful in the Upgrade_Rule discussed later in this document but applies equally as a resync condition It shows which term of the rule triggered the operation See section 6 5 for the values of these macro variables Upon successful firmware upgrade the ERR variable carries the version of the newly installed load 2004 Linksys Proprietary See Copyright Notice on Page 2 39 In addition the contents of the general purpose parameters GPP_A through GPP P are available as macro variables A through P respectively A secondary set of general purpose parameters is also available for macro substitution GPP_SA GPP_SB GPP_SC GPP_SD using the respective expressions SA SB SC and SD These parameters are not accessible through the web interface and can only be set via a configuration profile Strings identified above as val values are strings which can include variable substitution The macro variables are invoked by prefixing the name with a character e g MAC The substitution works even within a quoted string without requiring additional escapes If the variable name is immediately followed by an alphanumeric character enclose the variable name in parentheses e g MAC config xml To include a dollar sign in the rule escape it with another dollar sign That is map
140. n Initiation Protocol SLIC Subscriber Line Interface Circuit SP Service Provider PAP2 Phone Adaptor Ports 2 Linksys Phone Adaptor 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 10 Glossary 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 Area Code A 3 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 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 T
141. ne 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 subscribers mail box 0 1 2 Cfwd Dial Tone Special dial tone played when call forward ToneScript 350 19 440 all is activated 19 2 2 2 1 2 10 0 1 2 Holding Tone Indicate to the local user that the far end ToneScript 600 has placed the call on hold 16 1 1 1 1 1 1 1 9 5 1 Conference Tone Plays to all parties when a 3 way ToneScript 350 conference is in progress 16 30 1 1 1 1 9 7 1 Secure Call This tone is played when a call is ToneScript 397 19 507 Indication Tone successfully switched to secure mode It 19 15 0 2 0 2 1 1 should be played only for a short while lt 1 2 1 2 30s and at a reduced 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 2 Off Hook Warning Tone also called Howler Tone is played when Reorder Tone times out 4 10 Less Frequently Used Paramters 4 10 1 Advanced Protocol Parameters Parameter Name Description Type Default SIP Parameters Max Forward SIP Max Forward value Range 1 255 Uns8 70 2004 Linksys Proprietary See Copyright Notice on Page
142. ned password and a challenge nonce dynamically generated by the proxy server per request This mechanism prevents unauthorized user agents from getting IP Telephony services through the proxy server SIP proxy servers are operated by the IP Telephony service provider and resides at the service provider s domain They may be implemented in many different ways They can be stateless stateful or B2BUA Stateless proxies do not maintain states of each call they simply proxy the requests and responses between the user agents Hence they are the simplest most scalable but provide the least types of services Advanced IP Telephony services are possible with these proxies only with intelligent user agent devices that are capable of delivering these services without proxy intervention Stateful proxies maintain the call state of each call and can provide more intelligent services at the expense of more processing load per call B2BUA proxies process every request and response from the user agents and are capable of providing very advance services even with relatively simple user agent devices Obviously B2BUA proxies have the highest processing load per call 1 1 4 SIP Services Today s PSTN offers a large number of enhanced services in addition to basic phone services Most of the services offered by the PSTN are accessed by the subscribers through their telephone sets The subscribers provide their input by talking into the handset pressing the keyp
143. ng e ETSI DTMF After Ring CID only DTMF sent after 1st ring no polarity reversal or DTAS e ETSI FSK CID CIDCW and VMWI FSK sent after DTAS but no polarity reversal and before 1st ring Will wait for ACK from CPE after DTAS for CIDCW e ETSI FSK With PR UK CID CIDCW and VMWI FSK is sent after polarity reversal and DTAS and before 1st ring Will wait for ACK from CPE after DTAS for CIDCW Polarity reversal is applied only if equipment is on hook FXS Port Power Limit Options 1 2 3 4 5 6 7 8 Choice 3 Notes 1 It should be noted that the choice of CID method will affect the following features e On Hook Caller ID Associated with Ringing This type of Caller ID is used for incoming calls when the attached phone is on hook See figure below a c All CID methods can be applied for this type of caller id e On Hook Caller ID Not Associated with Ringing This feature is used for send VMWI signal to the phone to turn the message waiting light on and off see Figure 1 d and e This is available only for FSK based caller id methods Bellcore ETSI FSK and ETSI FSK With PR e Off Hook Caller ID This is used to delivery caller id on incoming calls when the attached phone is off hook See figure below f This can be call waiting caller ID CIDCW or to notify the user that the far end party identity has changed or updated such as due to a call transfer This is only ava
144. nking Phone is in use Incoming Call detected 2 2 Broadband Router RT31P2 LED Status LED Color s Activity Description Off Power OFF Solid Green Power On Green S Power Green 5 Blinking Booting System Self Test Firmware upgrade POST Power On Self Test failure not bootable Red On or Device malfunction Off No Connection on Ethernet Ethernet Blue So Green Ethernet Connection established reen S e Blinking Data Sending Receiving Off Phone is not in use not provisioned or registered Phone 1 Green On Registered provisioned Blue Phone 2 Green ae ere Phone is in use Incoming Call detected Blinking 4 One 5 Volt Power Adapter Interface Figure 3 above for PAP2 Phone Adapter and 12 Volt Power Adapter for the Broadband Router RT31P2 This interface accepts the PHONE ADAPTER power adapter that came with the unit Linksys does not support the use of any other power adapters other then the power adapter that was shipped with the PHONE ADAPTER unit or the Broadband Router RT31P2 2004 Linksys Proprietary See Copyright Notice on Page 2 Please check to make sure that you have the following package contents 1 Linksys Phone Adapter Unit or Linksys Broadband Router RT31P2 2 Ethernet Cable 3 5 Volt PAP2 or 12 Volt RT31P2 Power Adapter 4 CD with User Guide You will also need 1 One or Two Analog Touch Tone Telephones or Fax Machine 2 Access to an IP
145. o port specified default port of the protocol is used 69 for TFTP 80 for http 443 for HTTPS 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 PAP2 sctf 3 4 3 Reboot URL Through the Reboot URL you can reboot the PHONE ADAPTER Note Upon request the PHONE ADAPTER will reboot only when it is idle The Reboot URL is http lt Phone Adapter ip addr gt admin reboot 3 4 4 Factory Reset URL Through the Reset URL you can perform a factory reset of the PHONE ADAPTER Note Upon request the PHONE ADAPTER will reset and then reboot only when it is idle The Reset URL is http lt Phone Adapter ip addr gt admin reset 2004 Linksys Proprietary See Copyright Notice on Page 2 28 3 5 Configuration via the IVR PAP2 only Administrators and or users can check read and set write basic network configuration settings via a touchtone telephone connected to one of the RJ 11 phone ports of the PAP2 model PHONE ADAPTER Please Note Service Providers offering service using the PHONE ADAPTER may restrict protect or turn off certain aspects of the unit s IVR and web configuration capabilities The Interactive Voice Response IVR capabilities of the PHONE ADAPTER are designed to give the administrator and or user basic read write capabilities such that the unit can attain basic IP network connectivity and the more advanced browser based configur
146. od of provisioning may be applied by an administrator when the device is at the Service Provider s office or by the subscriber under the guidance of trained personnel during over the phone troubleshooting A third method of entering provisioning information into the PHONE ADAPTER is by way of its integral web server via a browser on a PC The subscriber has the option to set and adjust configuration parameters via an easy to use password protected graphical user interface This method of provisioning might be preferred by administrators who wish to access the PHONE ADAPTER over a secure corporate institutional LAN or by the residential subscriber who is a power user 1 1 3 Security Overview Security may be applied at many levels in the context of the PHONE ADAPTER The following are examples of information that should be secured e The configuration profile pulled from the provisioning server The downloading of the profile should be secured since it contains authentication password user name ID number information for accessing subscriber telephony services It may also contain other passwords and or encryption keys used for a variety of management and service operations e The administration password to the PHONE ADAPTER unit The unit must disallow access to administrative functions to unauthorized users This access can be controlled with an administrator password The administrator password can be one of the parameters
147. ofile from the provisioning server is very scalable and flexible Using this provisioning method a large number of PHONE ADAPTER units can be provisioned simultaneously and updated periodically However some basic information must be provided to the PHONE ADAPTER before it can be provisioned in this fashion a the IP address or domain name of the provisioning server to contact and b an ID and or a password to send to the provisioning server such that it can associate it with a specific subscriber and obtain the corresponding profile This information can be sent out of band to the subscriber via secured email or in a letter inside a welcome kit for example The subscriber might need to punch in some numbers using a telephone connected to the PHONE ADAPTER in order to enter this information into the unit The PHONE ADAPTER provides an easy to use interface with audio instructions to make this initial configuration process as painless as possible An alternative is for the unit to be provisioned with this basic information by the Service Provider before the unit is shipped to the subscriber In addition to the batch mode of remote provisioning the PHONE ADAPTER allows an interactive mode of local provisioning One way to offer this feature is through the use of an IVR system accessed through an attached telephone set The user can access a diagnostic or configuration menu to check the status of the device or to change some of the settings This meth
148. old button this feature provides access to a dial tone while the call is being held User Action Required to Activate or Use Press the switch hook or flash button on the phone to place the first party on hold You will hear a dial tone To make another call Enter the new number To return to call on hold Hang up and the phone set will ring with the first call on the line or Hook Flash again 2004 Linksys Proprietary See Copyright Notice on Page 2 85 Expected Call and Network Behavior User Action Required to Deactivate or End Hang up the telephone 5 13 Three Way Calling Service Description The user can originate a call to a 3rd party while engaging in an active call User Action Required to Activate or Use Press the switch hook or flash button on the phone to place the first party on hold Listen for three short tones followed by dial tone Dial the number of the 3 party When the 3 party answers you may have a conversation with them while the other party is on hold To hold a conference with the party on hold and the 3 party simply press the switch hook or flash button Expected Call and Network Behavior The PHONE ADAPTER supports up to two calls per line The PHONE ADAPTER can conference two calls by bridging the 277 and 3 parties User Action Required to Deactivate or End Hang up the telephone 5 14 Three Way Ad Hoc Con
149. old must have the SDP indicate a sendrecv or recvonly attribute and the remote destination address and port must not be 0 3 SAS Notes e Either or both of lines 1 and 2 can be configured as an SAS server e Each server can maintain up to 5 simultaneous calls If the second line on the PHONE ADAPTER is disabled then the SAS line can maintain up to 10 simultaneous calls Further incoming calls will receive a busy signal SIP 486 Response e The streaming audio source must be off hook for the streaming to occur Otherwise incoming calls will get a error response SIP 503 Response The SAS line will not ring for incoming calls even if the attached equipment is on hook e If no calls are in session battery is removed from tip and ring of the FXS port Some audio source devices have an LED to indicate the battery status This can be used as a visual indication whether any audio streaming is in progress 2004 Linksys Proprietary See Copyright Notice on Page 2 74 e IVR can still be used on an SAS line but the user needs to follow some simple steps a Connect a phone to the port and make sure the phone is on hook b power on the PHONE ADAPTER and c pick up handset and press to invoke IVR in the usual way The idea behind this is that if the PHONE ADAPTER boots up and finds that the SAS line is on hook it will not remove battery from the line so that IVR may be used But if the PHONE ADAPTER boots up and finds that the SAS line is of
150. on EE 46 4 3 Basic Networking Configuration ccccccccseceeseeceeeeeceaeeeeeeeeeeeeeecaaeeesaaesseeeeseaeeesaeesseeeenees 47 Network Configuration 0 820 cd aid ied a dine ced di pei a Denke 47 4 4 Basic Account Config rato Msie aienea ih tie iced dd a eee eg 48 4 5 Configuration for NAT Traversal cccccceeececeeeeeeeeceeeeecaeeeeaaeeseaeeesaaesseaeeseeeesaeessaeeeenees 49 4 6 Media and SDP Session Description Protocol Configuration ccccecceeeeneeeeeees 51 46 1 DTM an Hooktlash 29 ccveace gan aaa aara a aa ar aaa a aeaa Ea Teden paeronia 51 4 6 2 Co decand Audio Setting Srn neip Eet DEENEN eevee die te dated en eta teenie 52 4 6 3 Dynamic Payload Types and SDP Codec Names cccceecceseeeeeeeeeeeeeeeeeeeseaeeeeeeeteaeeeeeeessaeeseaeeteas 53 4 6 4 Secure Media Implementaton tetetete tte ttetstttttratnttinnttnttetstnnstnntnnnttnenneenneenne 54 4 6 5 Outbound Call Codec Selection Codes A 56 4 7 Supplementary Servis cccccccecceceeeeeceeeeeeeaeeeeeeeceeaeeeeaaeeeeeeeseeeesaaeeseaeeseeeeesaeessieeeneaees 57 4 7 1 Supplementary Services activated internally eecceeeseeeeeeeeeeeeeeeeeseeeeeeeeseaeeeeeeeseaeeseaeeseeeseaeeneas 58 4 7 2 Call Forwarding Implemented internally cccceeeeeesseeeeeeeeceeeeeeeeeeeaeeeeeeeseaeeseaeeteaeeeeaeeseeeseaeenias 60 4 7 3 Supplementary Services implemented in the service provider NetWork seessesssessesereerreeeerrnees 60
151. or BLock Caller ID 67 Notes The codes should not conflict with any of the other vertical service codes internally processed by the PHONE ADAPTER You can empty the corresponding code that you do not want to PHONE ADAPTER to process You can add a parameter 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 e lt Second Dial Tone gt X No tones are place x is any digit not used above If no tone parameter is specified the PHONE ADAPTER plays Prompt tone by default 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 simply add that code in the dial plan and the PHONE ADAPTER will send INVITE 73 as usual when user dials 73 Referral Services Codes One or more code can be configured into this 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 trigge
152. ous calls ActCode_ 77 Block AN Deact Code Unblock all anonymous calls ActCode_ 87 DND Ac Code Enable Do Not Disturb ActCode_ 78 DND_Deact_Code Disable Do Not Disturb ActCode_ 79 CID Ac Code Enable Caller ID Generation ActCode_ 65 CID Deact Code Disable Call ID Generation ActCode_ 85 CWCID_Act_Code Enable Call Waiting Caller ID generation ActCode_ 25 CWCID_Deact_Code Disable Call Waiting Caller ID generation ActCode_ 45 Dist_Ring Ac Code Enable Distinctive Ringing ActCode_ 61 Dist_Ring_Deact_Code Disable Distinctive Ringing ActCode_ 81 Speed Dial Act Code Assign a speed dial number ActCode_ 74 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 ActCode_ 19 This operation is redundant if all outbound calls are not secure by default In addition to the dynamic activation and deactivation codes the following parameters control the default activation or deactivation of internal parameters Parameter Name Description Type Default CW Setting Call Waiting on off by default for all calls Bool Yes Block CID Setting Block Caller ID on off by default for
153. ow it as the lt Call Round Trip Delay gt value ms in the Info section of PHONE ADAPTER web page 4 10 2 Additional User Account Information carrying a RTP data Enables the FXS Line to act as a Streaming Audio Source SAS If enabled the line cannot be used for making outgoing calls Instead it auto answers incoming calls and streams audio RTP packets to the calling party Bool Parameter Name Description Type Default Line Enable Enable this line for service Bool Yes MOH Server The User ID or URL of the auto answering SAS to Str127 Empty 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 regular outbound call MOH is disabled if this parameter is not specified empty SIP Port SIP message listening port and transmission port Port 5060 SIP TOS DiffServ TOS DiffServ field value in UDP IP Packets Byte 0x68 Value carrying a SIP Message RTP TOS DiffServ TOS DiffServ field value in UDP IP Packets Byte Oxb8 No SAS DLG Refresh Intvi If non zero this is the interval at which SAS sends out session refresh SIP re INVITE messages to detect if connection to the caller is still up If the caller does not respond to refresh message PHONE ADAPTER will terminate this call with a SIP BYE message The default 0 Session refresh disabled Range 0
154. owing parameters Outbound_Proxy Use_Outbound_Proxy NAT_Keep_Alive_Dest NAT_Keep_Alive_Msg NAT_Keep_Alive_Intvl and NAT_Keep_Alive_Enable If the NAT_Keep_Alive_Msg parameter is set to blank the PHONE ADAPTER will send a Carriage Return Line Feed as the Keep Alive Message The STUN approach works through more than 95 of home NATs when there is only a single PHONE ADAPTER in use behind the same NAT The STUN approach requires a STUN server setup by the provider but uses very few resources The actual media flows directly between the PHONE ADAPTER and its peer To configure STUN set the following parameters STUN_Enable STUN_Test_Enable STUN_Server NAT_Mapping_Enable Substitute_VIA_Addr NAT_Keep_Alive_Dest NAT Keep Ave Mao NAT_Keep_Alive_Intvl and NAT keep Alive_Enable The Manual Configuration approach requires coordinated administration of the NAT and the PHONE ADAPTER It is not practical for general retail use but can be used behind symmetric NATs occasionally found in larger businesses for troubleshooting and in circumstances where other mechanisms have been exhausted The configure the PHONE ADAPTER for manual NAT traversal set the EXT_IP parameter to the public translated outside external IP address the EXT_SIP_Port parameters per line to the translated port number for this line and PHONE ADAPTER and the EXT_RTP_Port_Min parameter to the first translated port number reserved for this PHONE 2004 Linksys Proprietary See
155. 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 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 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 Registration State 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 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 called Last Caller Number The number of the last caller Mapped SIP Port
156. pattern 1 8 for all callers 1 2 3 4 1 5 6 7 8 Hold Reminder Ring Ring pattern for reminder of a holding call when the 1 2 3 4 None phone is on hook 5 6 7 8 None Call Back Ring Ring pattern for call back notification 1 2 3 4 None 5 6 7 8 Ring1 Name Name in an INVITE s Alert Info Header to pick Str31 Bellcore r1 distinctive ring CWT 1 for the inbound call Ring2 Name Name in an INVITE s Alert Info Header to pick Str31 Bellcore r2 distinctive ring CWT 2 for the inbound call Ring3 Name Name in an INVITE s Alert Info Header to pick Str31 Bellcore r3 distinctive ring CWT 3 for the inbound call Ring4 Name Name in an INVITE s Alert Info Header to pick Str31 Bellcore r4 distinctive ring CWT 4 for the inbound call Ring5 Name Name in an INVITE s Alert Info Header to pick Str31 Bellcore r5 distinctive ring CWT 5 for the inbound call Ring6 Name Name in an INVITE s Alert Info Header to pick Str31 Bellcore r6 distinctive ring CWT 6 for the inbound call Ring7 Name Name in an INVITE s Alert Info Header to pick Str31 Bellcore r7 distinctive ring CWT 7 for the inbound call 2004 Linksys Proprietary See Copyright Notice on Page 2 67 Ring8 Name Name in an INVITE s Alert Info Header to pick distinctive ring CWT 8 for the inbound call Str31 Bellcore r8 Cfwd Ring Splash Len Duration of ring splash when a call is forwarded 0 10 0s Time3 0 Cblk Ring Spla
157. pecified Syslog messages are also logged to the Debug Server 4 4 Basic Account Configuration Basic SIP Account Configuration is typically straightforward involving only a handful of key parameters All of these parameters are configured on a per line basis The Line_Enable parameters control whether a line is enabled or not The Proxy setting is the address of the SIP Registrar usually collocated with a SIP Proxy for the account The User_ID is the username or phone number of the SIP account The Proxy and User_ID together form the SIP URI For example User_ID alice Proxy sip provider net 5060 the SIP URI used for registration would be sip alice sip provider net 5060 The Password is the password used for Digest authentication With some providers the username used for authentication is different from the User_ID used in the SIP From header For example Alice Smith could have a User_ID of 1234 and a Digest username of alice smith In this situation set the Auth_ID to alice smith and set Use_Auth_ID to yes The Display_Name is the string that will appear in quotes in the From header It can be an arbitrary string such as a name for example Alice Smith or a local phone number for example 5551212 Proxy and Registration Proxy SIP Proxy Server for all outbound requests FQDN Register Enable periodic registration with the lt Proxy gt This Bool Yes parameter is ignored if lt Proxy gt is not specifie
158. pect to 1 milliwatt Dynamic Host Configuration Protocol Domain Name Server Dynamic Random Access Memory Digital Subscriber Loop Digital Signal Processor Data Terminal Alert Signal same as CAS Dual Tone Multiple Frequency European Telecommunication Standard Fully Qualified Domain Name Frequency Shift Keying Foreign eXchange Station Gateway International Telecommunication Union Hypertext Markup Language Hypertext Transfer Protocol HTTP over SSL Internet Control Message Protocol Internet Group Management Protocol Incumbent Local Exchange Carrier Internet Protocol Internet Service Provider IP Telephony Service Provider Interactive Voice Response Local Area Network Low Bit Rate Low Bit Rate Codec Mini Certificate Media Gateway Control Protocol Music On Hold Mean Opinion Score 1 5 the higher the better Millisecond Music Source Adaptor Message Waiting Indication Open Switching Interval Printed Circuit Board Polarity Reversal Provisioning Server Perceptual Speech Quality Measurement 1 5 the lower the better Public Switched Telephone Network Network Address Translation Out of band SIP Request Message SIP Response Message SIP Response Status Code such as 404 302 600 Real Time Protocol KS Se 2004 Linksys Proprietary See Copyright Notice on Page 2 114 RTT Round Trip Time SAS Streaming Audio Server SDP Session Description Protocol SDRAM Synchronous DRAM sec seconds SIP Sessio
159. peed Dial 5 2 Speed Dial 6 2 Speed Dial 7 2 Speed Dial 8 2 Speed Dial 9 2 wu wu wu wu wu wu wu Supplementary Service Settings CW Setting 2 Yes Block CID Setting 2 No Block ANC Setting 2 No DND_Setting 2 No CID Setting 2 Yes CWCID Setting 2 Yes Dist_Ring_Setting 2 Yes Secure_Call_Setting 2 No 2004 Linksys Proprietary See Copyright Notice on Page 2 Distinctive Ring Settings Rina Caller 2 Ring2 Caller 2 Ring3 Caller 2 Ring4 Caller 2 Ring5 Caller 2 Ring6 Caller 2 Ring7 Caller 2 Ring8 Caller 2 Ring Settings wu wu wu wu wu wu wu wu Default Ring 2 Default _CWT 2 Hold Reminder _Ring 2 Call Back_Ring 2 Cfwd_Ring Splash_Len 2 Cb1lk_ Ring Splash_Len 2 VMWI_Ring Splash Lentz VMWI_ Ring Policy 2 Available New VM Becomes Available New Ring On No New_VM 2 Call Progress Tones Dial_Tone Second _Dial_ Tone Outside Dial Tone Prompt_Tone Busy _Tone Reorder_Tone Off Hook Warning Tone Ring_Back_Tone Confirm_Tone SIT1_Tone 20 380 0 1 380 0 2 SIT2_Tone 20 274 0 1 274 0 2 SIT3_Tone 20 380 0 1 380 0 2 SIT4_ Tone 20 380 0 1 274 0 2 MWI_Dial_ Tone Cfwd_Dial_Tone Holding Tone Conference Tone Secure_Call_ Indication_Tone Distinctive Ring Patterns Ringl_ Cadence Ring Cadence Ring3_ Cadence Ring4 Cadence Ring5 Cadence
160. phony subscribers When the telephone rings pick up the handset and begin talking Expected Call and Network Behavior Each subscriber is assigned an E 164 ID phone number so that they may be reached 2004 Linksys Proprietary See Copyright Notice on Page 2 from wired or wireless callers on the PSTN or IP network The PHONE ADAPTER supplies ring voltage to the attached telephone set to alert the user of incoming calls User Action Required to Deactivate or End Hang up the telephone 5 3 Caller ID Service Description If available the PHONE ADAPTER supports the generation and pass through of Caller ID information User Action Required to Activate or Use No user action required The user s telephone equipment must support Caller ID to display the caller s name and or number Expected Call and Network Behavior In between ringing bursts the PHONE ADAPTER can generate a Caller ID signal to the attached phone when the phone is on hook As part of the INVITE message the PHONE ADAPTER sends the callers name and number as it is configured in the profile User Action Required to Deactivate or End No user action required See CLIP and CLIR 5 4 Calling Line Identification Presentation CLIP Service Description Some users will elect to block their Caller ID information for all outgoing calls However there may be circumstances where sending
161. pt Phone Adapter cfg Profile Rule B ProfileScript Profile Rule C ProfileScript Profile Rule D ProfileScript Log Resync Request Syslog message generated when attempting ProfileMsg See Msg a resync provisioning discussion section Log Resync Success Syslog message generated after a ProfileMsg See Msg successful resync provisioning discussion section Log Resync Failure Syslog message generated after a failed ProfileMsg See Msg resync provisioning discussion section GPP A thru GPP P General purpose parameter String empty GPP SA thru GPP SD General purpose parameter String empty Note In a customized PHONE ADAPTER the profile rule would point to a service provider s server 4 2 1 Firmware Upgrade The PHONE ADAPTER is firmware upgradeable via TFTP and 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 pap2 bin The PHONE ADAPTER can be configured to upgrade to a specific version possibly staging through intermediate releases if necessary This process can be automated for a pool of devices through configuration profile parameters Alternatively an individual PHONE ADAPTER can be directed to perform an upgrade to a specific firmware load via its built in web server interface this mechanism is discussed in section 3 4 1 of th
162. r the jitter buffer the more jitter it can absorb but this also introduces bigger delay Therefore the jitter buffer size should be kept to a relatively small size whenever possible If jitter buffer size is too small then many late packets may be considered as lost and thus lowers the Voice Quality The PHONE ADAPTER can dynamically adjust the size of the jitter buffer according to the network conditions that exist during a call Echo Impedance mismatch between the telephone and the IP Telephony gateway phone port can lead to near end echo The PHONE ADAPTER has a near end echo canceller with at least 8 ms tail length to compensate for impedance match The PHONE ADAPTER also implements an echo suppressor with comfort noise generator CNG so that any residual echo will not be noticeable Hardware Noise Certain levels of noise can be coupled into the conversational audio signals due to the hardware design The source can be ambient noise or 60Hz noise from the power adaptor The PHONE ADAPTER hardware design minimizes noise coupling End to End Delay End to end delay does not affect Voice Quality directly but is an important factor in determining whether subscribers can interact normally in a conversation taking place over an IP network Reasonable delay figure should be about 50 100ms End to end delay larger than 300ms is unacceptable to most callers The PHONE ADAPTER supports end to end delays well within acceptable thresholds 2
163. r packet This codec provides the high voice quality 7 2 4 3 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 7 2 4 4 G 723 1 The PHONE ADAPTER supports the use of ITU G 723 1 audio codec at 6 4 kbps Up to 2 channels of G 723 1 can be used simultaneously For example Line 1 and Line 2 can be using G 723 1 simultaneously or Line 1 or Line 2 can initiate a 3 way conference with both call legs using G 723 1 7 2 5 Codec Selection The administrator can select which low bit rate codec to be used for each line G711a and G711u are always enabled 7 2 6 Dynamic Payload When no static payload value is assigned per RFC 1890 the PHONE ADAPTER can support dynamic payloads for G 726 7 2 7 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 7 2 8 Fax Tone Detection Pass Through Users can connect a fax terminal to the PHONE ADAPTER telephone port s Fax terminals transmit a single tone when they answer a call The PHONE
164. red rejected by placing a character at the end of the sequence Thus 1 900xxxxxxx automatically rejects all 900 area code numbers from being dialed 2004 Linksys Proprietary See Copyright Notice on Page 2 63 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 3649 e Long interdigit timer L delay value e Short interdigit timer S 7 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 Similarly to change the short timer override use S delay value lt space gt These overrides are especially useful to terminate dialing in countries with predictable but variable length numbering plans or to provide an exception when a rule with fewer digits is known to override a rule waiting for more digits For example assuming a generic international calling sequence of 011Xxxxxxxxx in North
165. riables as for the Profile Rule above An empty string does not generate a syslog message Log Resync Success Message ParName Log_Resync_Success_Msg Default SPN SMAC Successful resync SSCHEME S SERVIP SPORTSPATH The Log_Resync_Success_Msg is a script that defines the message sent to the configured Syslog server whenever the PHONE ADAPTER successfully completes a resync with the provisioning server The string supports one level of macro substitution with the same variables as for the Profile Rule above An empty string does not generate a syslog message Log Resync Failure Message ParName Log_Resync_Failure_Msg Default SPN SMAC Resync failed SERR The Log_Resync_Failure_Msg is a script that defines the message sent to the configured Syslog server whenever the PHONE ADAPTER fails to complete a resync with the provisioning server The 2004 Linksys Proprietary See Copyright Notice on Page 2 41 string supports one level of macro substitution with the same variables as for the Profile_Rule above An empty string does not generate a syslog message General Purpose Parameters ParName GPP_A through GPP_P Default empty GPP_A through GPP_P are the 16 General Purpose Parameters usable by both the provisioning and the upgrade logic Each general purpose parameter can be configured to hold any string value Such a value can then be incorporated in other scripte
166. riggered by a character encountered in the dial plan Prompt Tone Played when prompting the user to enter a ToneScript 520 19 620 call forward phone number 19 10 0 1 2 Busy Tone Played when a 486 RSC is received for an ToneScript 480 19 620 outbound call 19 10 5 5 1 2 Reorder Tone Played when an outbound call has failed ToneScript 480 19 620 or after the far end hangs up during an 19 10 25 25 1 2 established call Off Hook Warning Played when the subscriber does not ToneScript 480 Tone place the handset on the cradle properly 10 620 0 10 125 125 142 Ring Back Tone Played for an outbound call when the far ToneScript 440 19 480 19 2 4 1 2 2004 Linksys Proprietary See Copyright Notice on Page 2 69 Confirm Tone This should be a brief tone to notify the ToneScript 600 user that the last input value has been 1631 25 25 1 accepted SIT1 Tone An alternative to lt Reorder Tone gt played ToneScript 985 16 1428 when an error occurs while making an 16 1777 outbound call The RSC to trigger this tone 16 20 380 0 1 380 is configurable see Section 0 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 To
167. rivilege use this URL for the PAP2 http IP_Address_ Of PHONE ADAPTER admin and this URL for the RT31P2 http IP_Address_ Of PHONE ADAPTER Voice_adminPage htm The default IP address for the LAN interface of the RT31P2 is 192 168 15 1 See the next section for more information about administration privileges o The PHONE ADAPTER supports Internet Explorer 5 5 and above and Netscape 7 0 and above o The web configuration pages can be password protected See 3 3 2 for more information about password protect o The user name of web Administrator is admin o The user name of web User is user o Note The user names for both administrator and User are fixed and cannot be changed o After making changes to PHONE ADAPTER configuration parameters pressing Submit All Changes button will apply all the changes and if necessary automatically reboot the device Multiple changes may be made on multiple page tabs of the web interface at the same time Pressing Submit All Changes will apply all the modifications Important Note switching between page tabs won apply the changes to PHONE ADAPTER The only way to apply the changes is to press the Submit All Changes button o 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 2004 Linksys Proprietary See Copyright Notice on Page 2 26 NOTE Pressing the
168. rom the User pages you 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 3 3 3 Basic and Advanced Views The PAP2 web configuration interface provides a Basic and an advanced view from which the various configuration parameters can be accessed The PHONE ADAPTER 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 already been made on the web pages 3 4 Functional Configuration URLs The web interface of the PHONE ADAPTER supports several functions through special URLs Upgrade Reboot Profile Resync and Factory Reset Administrator privilege is needed for these functions Note that on the RT31P2 these URLs are only accessible from the LAN interface unless the Admin_Passwd has been set and the Enable_Web_Admin_Access parameter is set 3 4 1 Upgrade URL Through upgrade URL you can upgrade the PHONE ADAPTER to a firmware specified by the URL Note If the value of upgrade enable parameter in Provisioning tab is no you cannot upgrade the PHONE ADAPTER even if the web page tells you that the upgrade will be done when it is not in use See 4 2 1 to get more informat
169. rs the PHONE ADAPTER to perform a blind transfer to a target number that is prepended by the service code For example after the user dials 98 the PHONE ADAPTER 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 normal dialing When a complete number is entered the PHONE ADAPTER sends a blind REFER to the holding party with the Refer To target equals to 98 lt target_number gt This feature allows the PHONE ADAPTER 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 PHONE ADAPTER You can empty the corresponding code that you do not want to PHONE ADAPTER to process 4 8 Dial Plan Configuration The PHONE ADAPTER allows each line 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 PHONE ADAPTER syntax for the dial plan closely resembles the corresponding syntax specified by MGCP and MEGACO Some extensions are added that are useful in an end point 2004 Linksys Proprietary See Copyright Notice on Page 2 61 The dial plan functionality is regulated by the following configurable parameters Interdigit_Long_Timer Interdigit_Short_Timer Dia
170. ry must contain a configuration file or a CGI application needs to generate the appropriate config file if that MAC address is configured in your system The Apache web server is freely available at http www apache org Once an initial XML configuration file is downloaded from the provider web server subsequent configuration can be downloaded from the same server Alternatively the individual configuration files can be encrypted using AES 256 bit encryption as described previously using a key that was conveyed in the initial configuration file These encrypted configuration files then can be downloaded safely using HTTP or TFTP Linksys recommends using an encrypted configuration file In the unlikely event that the private key of a terminal adapter or the Linksys certificate authority is compromised terminal adapters which have already enrolled with a provider and use an encrypted configuration file would be unaffected by such a compromise 3 3 Web Interface The PHONE ADAPTER provides a built in web server Configuration and administration can be performed through this convenient web interface 3 3 1 Web Interface Conventions The PHONE ADAPTER line uses the following conventions with the web administration capabilities o The PHONE ADAPTER 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 PHONE ADAPTER to use the administrator p
171. s GMT 12 00 GMT 11 00 GMT 10 00 GMT 09 00 GMT 08 00 GMT 07 00 GMT 06 00 GMT 05 00 GMT 04 00 GMT 03 30 GMT 03 00 GMT 02 00 GMT 01 00 GMT GMT 01 00 GMT 02 00 GMT 03 00 GMT 03 30 GMT 04 00 GMT 05 00 GMT 05 30 GMT 05 45 GMT 06 00 GMT 06 30 GMT 07 00 GMT 08 00 GMT 09 00 GMT 09 30 GMT 10 00 GMT 11 00 GMT 12 00 GMT 13 00 FXS Port Impedance 600 options 600 900 600 2 16uF 900 2 16uF 270 750 150nF 220 820 120nF 220 820 115nF 370 620 310nF FXS Port Input Gain VS HN FXS_ Port Output Gain erh DTMF_ Playback Level ten DTMF Playback Length ER 3 Detect_ABCD Yes Playback_ABCD Yes Caller ID Method Bellcore N Amer China options Bellcore N Amer China DTMF Finland Sweden DTMF Denmark ETSI DTMF ETSI DTMF With PR ETSI DTMF After Ring ETSI FSK ETSI FSK With PR UK FXS_ Port Power Limit 3 options 1 2 3 4 5 6 7 8 Protect_IVR_FactoryReset No 9 Acronyms A D Analog To Digital 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 CAS CPE Alert Signal CDR Call Detail Record CID Caller ID 2004 Linksys Proprietary See Copyright Notice on Page 2 113 RTP Call Waiting Caller ID Comfort Noise Generation Calling Party Control Customer Premises Equipment Call Waiting Caller ID Call Waiting Tone Digital to Analog Converter decibel dB with res
172. s 0 the PHONE ADAPTER will stop retrying This value should be much larger than lt Reg Retry Intvl gt which should not be 0 Timed 1200 Response Status Co de Handling SIT1 RSC SIP response status code to INVITE on which to play the SIT1 Tone RscTmplt SIT2 RSC SIP response status code to INVITE on which to play the SIT2 Tone RscTmplt SIT3 RSC SIP response status code to INVITE on which to play the SIT3 Tone RscTmplt SIT4 RSC SIP response status code to INVITE on which to play the SIT4 Tone RscTmplt Try Backup RSC Retry Reg RSC SIP response status code on which to retry a backup server for the current request Interval to wait before the PHONE ADAPTER retries registration again after encountering a failure condition during last registration RscTmplt Time 30 RTP Parameters RTP Port Min Minimum port number for RTP transmission and reception Port 16384 RTP Port Max Maximum port number for RTP transmission and reception Port 16482 RTP Packet Size Packet size in sec Valid values must be multiple of 0 01s Range 0 01 0 16 Time3 0 02 RTCP Tx Interval Controls the interval sec to send out RTCP sender report on an active connection Range 0 255 s Time Notes 1 2 3 Reorder or Busy Tone will be played by default for all unsuccessful response status code lt RTP Port Min g
173. s then used to upgrade the PHONE ADAPTER No additional URLs in the rule are considered 2004 Linksys Proprietary See Copyright Notice on Page 2 44 The upgrade will fail if the new firmware load does not satisfy the upgrade rule condition that suggested the URL This alleviates the possibility of infinite upgrade loops in case the device has been misconfigured The rule syntax is the same as for the Profile Rule described in a previous section except that there are no supported optional qualifiers for upgrades at this time That is the bracketed options preceding the URL are not supported in the Upgrade_ Rule Upgrade Rule Syntax Examples each line is a separate example 1 0 2 Phone Adapter2000 1 00 02 Phone Adapter bin lt 1 0 tftp pserv myvoice com 42001 upg Phone Adapter2000 1 0 2 Phone Adapter bin lt 0 99 52 Phone Adapter09952 bin lt 1 0 2 Phone Adapterl0002 bin Log Upgrade Request Message ParName Log_Upgrade_Request_Msg Default SPN SMAC Requesting upgrade SSCHEME SSERVIP SPORTSPATH The Log_Upgrade_Request_Msg is a script that defines the message sent to the configured Syslog server whenever the PHONE ADAPTER attempts an upgrade from the upgrade server The string supports one level of macro substitution with the same variables as for the Upgrade_Rule above An empty string does not generate a syslog message Log Upgrade Success Message
174. s through the mapping to extend its validity as necessary 1 3 Voice Quality Overview Voice Quality perceived by the subscribers of the IP Telephony service should be indistinguishable from that of the PSTN Voice Quality can be measured with such methods as Perceptual Speech Quality Measurement PSQM 1 5 lower is better and Mean Opinion Score MOS 1 5 higher is better The table below displays speech quality metrics associated with various audio compression algorithms Algorithm Bandwidth Complexity MOS Score G 711 64 kbps Very Low 4 5 G 726 16 24 32 40 kbps Low 4 1 32 kbps G 729a 8 kbps Low Medium 4 G 729 8 kbps Medium 4 G 723 1 6 3 5 3 kbps High 3 8 Please note The PHONE ADAPTER supports all the above voice coding algorithms 2004 Linksys Proprietary See Copyright Notice on Page 2 16 Several factors that contribute to Voice Quality are described below Audio compression algorithm Speech signals are sampled quantized and compressed before they are packetized and transmitted to the other end For IP Telephony speech signals are usually sampled at 8000 samples per second with 12 16 bits per sample The compression algorithm plays a large role in determining the Voice Quality of the reconstructed speech signal at the other end The PHONE ADAPTER supports the most popular audio compression algorithms for IP Telephony G 711 a law and u law G 726 G 729a and G 723 1
175. s to Profile_Rule syntax examples each line is a separate example pap2 cfg pserv myvoice com 42000 sip SMA pap2 cfg key 6e4f2a8733ba7c90aal3250bde4f6927 ur well com Gj2fLx3Nqbg a cfg Lilien Ee cre Zemee ee Profile Example Scenarios Enterprise LAN with DHCP Supplied TFTP Server Name The DHCP server automatically advertises a TFTP server name to service the local network Each PHONE ADAPTER in the network is supplied a unique CFG file based on its MAC address The TFTP server would also contain a generic Phone Adapter2000 cfg in its tftp root directory that contains the Profile_Rule indicated below It would additionally carry individualized CFG files one per device within a tree below the tftp root node Each of these files would then individualize the devices profiles SMA pap2 cfg When first powered on unprovisioned devices would download the pap2 cfg file from the TFTP server indicated by DHCP following their manufacturing default setting for the Profile_Rule parameter The downloaded file would then direct the PHONE ADAPTER to resync to the server and fetch the individualized CFG file as per the rule above which completes the provisioning sequence VoIP Service Provider Conceptually a service provider solution would follow the steps as in the above example In addition it would then proceed to enable stronger encryption by implementing one more provisioning step with one more leve
176. sage will be sent from the Voice Mail server to the PHONE ADAPTER When the user dials their own phone number the PHONE ADAPTER connects the subscriber their voice mail system which can then connect them to their individual voice mail box Follow instructions of the voice mail system or simply hang up the telephone 5 10 Attendant Call Transfer Service Description Attendant Call Transfer lets a customer use their Touchtone phone to send a call to any other phone inside or outside their business including a wireless phones User Action Required to Activate or Use While in a call with the party to be transferred Press the switch hook or flash button on the phone to place the party on hold Listen for three short tones followed by dial tone Dial the number to which you will transfer the caller Stay on the line until the called number answers Announce the call Press the switch hook or flash button adding the held party to the call Hang up to connect the two parties and transfer the call Note You can hook flash while the 3 party is ringing to start an early conference Then hang up to complete the transfer without waiting for the 3 party to answer first Expected Call and Network Behavior When the user presses the switch hook or flash button the transferee is placed on hold When the user successfully dials the transfer number and the party answers the transferee can be added to the
177. sh Len Duration of ring splash when a call is blocked 0 10 0s Time3 0 VMWI Ring Splash Len Duration of ring splash when new messages arrive before the VMWI signal is applied 0 10 0s Time3 5 VMWI Ring Policy The parameter controls when a ring splash is played when a the VM server sends a SIP NOTIFY message to the PHONE ADAPTER indicating the status of the subscriber s mail box 3 settings are available New VM Available ring as long as there is 1 or more unread voice mail New VM Becomes Available ring when the number of unread voice mail changes from 0 to non zero New VM Arrives ring when the number of unread voice mail increases Choice New VM Available Ring On No New VM If enabled the PHONE ADAPTER will play a ring splash when the VM server sends SIP NOTIFY message to the PHONE ADAPTER indicating that there are no more unread voice mails Some Bool No signal to turn off VMWI lamp equipment requires a short ring to precede the FSK Notes 1 Caller number patterns are matched from Ring 1 to Ring 8 The first match not the closest match will be used for alerting the subscriber Parameter Name Description Type Default Ring1 Cadence Cadence script for distinctive ring 1 CadScript 60 2 4 Ring2 Cadence Cadence script for distinctive ring 2 CadScript 60 3 2 1 2 3
178. small device that sits at the subscriber s premises It converts between analog telephone signals and IP Telephony signals It has up to two RJ 11 ports where standard analog telephones can be directly attached and an RJ 45 interface for the Ethernet connection to the home or business LAN Intelligence can be built into this device to provide a wide range of features to the subscribers in association with the other elements in the service The PHONE ADAPTER functions as a SIP User Agent UA Home SOHO Routers with NAT Functionality A home SOHO router is used for routing IP packets between the subscriber s private network and the ISP s public network If the ISP provides only one public IP address to the subscriber the devices attached to the private network will be assigned private IP addresses and the router will perform network address translation NAT on packets sent from the private network to the public network via the router Home routers offer the following features e An R J45 WAN interface for connection to the ISP s public network and one or more RJ 45 LAN interfaces for connection to the subscriber s private network The router directs packets between the private network and the public network e A PPPOE client to connect with the ISP through a DSL modem e A DHCP client where the router will obtain an IP address subnet mask default router assignment etc for its WAN interface from a DHCP server on the public network e
179. ssage is waiting by playing stuttered dial tone or other configurable tone when the user picks up the handset Checking Voice Mail The PHONE ADAPTER allows the subscriber to connect to their voice mail box by dialing their personal phone number 1 1 4 2 4 Call Transfer Three parties are involved in Call Transfer The transferor transferee and transfer target There are 2 flavors of call transfer Attended Transfer Transfer with consultation and Unattended Transfer Blind Transfer Attendant Transfer The transferor dials the number of the transfer target then he hangs up or enters some or code when the transfer target answers or rings to complete the transfer Unattended or Blind Transfer The transferor enters some or code and then dials the number of the transfer target to complete the transfer without waiting for the target to ring or answer 1 1 4 2 5 Call Hold Call Hold lets you put a caller on hold for an unlimited period of time It is especially useful on phones without the hold button Unlike a hold button this feature provides access to a dial tone while the call is being held 1 1 4 2 6 Three Way Calling The subscriber can originate a call to a 3rd party while engaging in an active call 1 1 4 2 7 Three Way Ad Hoc Conference Calling The PHONE ADAPTER can host a 3 way conference and perform 3 way audio mixing without the need of an external conference bridge device or service 1
180. t 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 PHONE ADAPTER 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 PHONE ADAPTER 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 PHONE ADAPTER can be programmed to send out compound RTCP packet on the connection Each compound RTP 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 2004 Linksys Proprietary See Copyright Notice on Page 2 72 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 PHONE ADAPTER2000 1 0 31 b The NTP timestamp used in the SR is a snapshot of the PHONE ADAPTER s local time not the time reported by an NTP server If the PHONE ADAPTER receives a RR from the peer it will attempt to compute the round trip delay and sh
181. t Ring Serv 2 Cfwd_ All Serv 2 Cfwd_Busy Serv 2 Cfwd_No Ans Serv 2 Cfwd_ Sel Serv 2 Cfwd_Last_Serv 2 Block Last _Serv 2 Accept Last Serv 2 DND_Serv 2 CID Serv 2 CWCID_Serv 2 Call Return _Serv 2 Call Back Serv 2 Three Way Call Serv 2 Three Way Conf Serv 2 Attn Transfer Serv 2 Unattn_ Transfer Serv 2 MWI_ Serv 2 VMWI_ Serv 2 Speed Dial Serv 2 Secure Call Serv 2 Referral Serv 2 Feature Dial Serv 2 Audio Configuration Preferred Codec 2 Yes Yes n Yes W Yes W W Yes W W Yes W W Yes Yes W Yes W Yes W Yes Yes Yes W Yes W W Yes Yes W ut Yes Yes n Yes Yes W Yes n W Yes Yes n Yes n Yes Yes G711u G726 24 G726 32 G726 40 G729a G723 Silence_Supp_Enable 2 Use_Pref_Codec_Only 2 Echo_Canc_Enable 2 G729a_Enable 2 Echo_Canc_Adapt_Enable 2 G723_Enable 2 Echo Supp Enable 2 G726 16 Enable 2 FAX CED Detect_Enable 2 G726 24 Enable 2 FAX CNG Detect_Enable 2 G726 32 Enable 2 FAX Passthru_Codec 2 G726 40 Enable 2 FAX Codec_Symmetric 2 No W No W Ye s n Ye s W W Ye s W Ye s W W Ye s Ye s Ye s W n Ye s W Ye s Ye s G711u Ye s W Ye s 1 b b b b options G711u G711la G726 16 options G711u G71la 2004 Linksys Proprietary See Copyright Notice on Page 2 109 DTMF_ Tx Method 2 Auto options
182. te set of configuration parameters for an PHONE ADAPTER corresponding to an individual subscriber is referred to as a configuration profile or simply a Profile The Profile can be encoded as an XML file or a simple plain text file with a list of tag value pairs When the PHONE 2004 Linksys Proprietary See Copyright Notice on Page 2 9 ADAPTER unit is shipped from the factory it contains a default common Profile and is considered Unprovisioned To save costs and expedite delivery however it is very desirable that an Unprovisioned unit can be shipped directly from the factory to the subscriber s location without any preprocessing by the Service Provider The PHONE ADAPTER contacts the Service Providers provisioning server via the IP network or Internet when it is plugged into the subscribers home or business Local Area Network LAN assuming the provisioning server is reachable from the subscriber s home network to pull the designated profile to be installed in that particular PHONE ADAPTER unit Furthermore the PHONE ADAPTER unit will periodically contact the provisioning server to download an updated profile The protocol for downloading the configuration profile can be clear text TFTP or HTTP data or it can be encrypted TFTP HTTP or HTTPS data if security is required Security will be discussed in more details in a later section This type of autonomous remote provisioning where the individual PHONE ADAPTER unit pulls the pr
183. ter the telephone number you are forwarding your call to Activation will be confirmed with three short bursts of tone and your forwarding will be activated Alternatively the user can activate this feature from a web browser interface Expected Call and Network Behavior This feature allows a user the option to divert forward calls to their telephone number to any number when their phone is busy or in conference by using the touchtone keypad of their telephone or web browser interface This service is activated or deactivated from the phone being forwarded or the web browser interface User Action Required to Deactivate or End Lift the receiver Listen for dial tone Dress You will hear a confirmation tone signaling your change has been accepted Alternatively the user can deactivate this feature from a web browser interface 5 19 Call FWD No Answer Service Description Calls are forwarded to the designated forwarding number after a configurable time period elapses while the PHONE ADAPTER is ringing and does not answer User Action Required to Activate or Use Lift the receiver Listen for dial tone Drees 2004 Linksys Proprietary See Copyright Notice on Page 2 Listen for dial tone and enter the telephone number you are forwarding your call to Activation will be confirmed with three short bursts of tone and your forwarding will be activated Alternat
184. that in XML less than lt and ampersand amp characters within an element must be escaped using amp lt and amp amp respectively Element names in XML are case sensitive lt flat profile gt lt Profile_Rule gt https config provider net linksys MA cfg xml lt Profile_Rule gt lt Resync_Periodic gt 86400 lt Resync_Periodic gt lt Admin_Passwd gt 9b4cef5677a1 29 lt Admin_Passwd gt lt Proxy_1_ gt sip provider net lt Proxy_1_ gt lt User_ID_1_ gt 1234567890 lt User_ID_1_ gt lt Password_1_ gt YhJ89_Luk4E lt Password_1_ gt lt Display_Name_1_ gt 1234567890 lt Display_Name_1_ gt lt Line_Enable_2 gt 0 lt Line Enable 2 gt lt flat profile gt The Linksys Supplementary Profile Compiler tool SPC is provided for compiling a plain text file containing parameter value pairs into a binary cfg file which is optionally encrypted The spc tool is available from Linksys for the Win32 environment spc exe Linux i386 elf environment spc linux i386 static and for the OpenBSD environment The syntax of the plain text file accepted by the profile compiler is a series of parameter value pairs with the value in double quotes Each parameter value pair is followed by a semicolon e g parameter_name parameter_value If no quoted value is specified for a parameter or if a parameter specification is missing entirely from the plain text file the value of the parameter will remain unchanged in the PHONE ADAPTER
185. the only ActCode 027111 codec that can be used for the associated call 2004 Linksys Proprietary See Copyright Notice on Page 2 56 codec that 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 ActCode_ 02723 codec that 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 ActCode 0272616 codec that 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 ActCode 0272624 codec that 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 will make this codec the only ActCode_ 0272632 codec that 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 ActCode_ 0272640 codec that can be used for the
186. tion Restriction CLIR Caller ID Blocking sseeeeseeesseeseeeeeen 81 5 6 EW dl DE EE 81 5 7 Disable or Cancel Call Wang 82 5 8 Call Waiting with Caller ID 83 5 9 Viet MET EE E 83 5 10 Attendant Gall Transtar EE 84 5 11 Unattended or Blind Call Transier nnne tn rnst tnnnnsnnnnnnnnnnnnnnnnnnenn nnne nenn 85 Ee UR e e DE 85 5138 Three Way Callin EE 86 5 14 Three Way Ad Hoc Conference Calling c ccccccceceeeeeeseeceeeeeeeeeesaeeeeeeeeseaeeesaeeesaaeeeenees 86 E Call REL EE 87 5 16 Automatic Call Back 87 5 17 Call FWD Unconditional cc es en ki EEE rRNR E EERE E PERT AEREE T 88 5148 Cal FWD BUSY er e E E EE E TEENE N TENE 89 5 19 Call FWD No Answer 89 5 20 Anonymous Call Blockimng 90 5 21 Distinctive Priority Ringing and Call Waiting Tone ssssessessssseessiessiesrissriesrinsrrresrrssres 90 5 22 Speed Calling Up to Eight 8 Numbers or IP Addresees nenene 91 Geo THOUBIOSNOOUNG se eer rE EIE E E Eege eege Eege 92 6 1 Call Statistics e ul e DEE 92 6 2 Enabling Logging and Debugging s sssseesseessessesssressntstnnstnnstnnnttnnttnutnntnnnnnnnnnsnnsnnnnnnn 93 6 3 Error and Log Reporting EE 93 6 4 Internal Error Codes itis seevecdve ne favs Meee tapenade de eege edel 93 6 5 Provisioning and Upgrade result codes A 94 6 6 Table of SIP Response Codes Error Codes sssesssessssssressrrssrnssrnssrnssrnsssnnssnnsrnssrnsnnt 94 7 Summary of Implemented Features and Gpec
187. tion file the last such definition in the file is the one that will take effect in the PHONE ADAPTER e A parameter 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 4 2 Provisioning Related Parameters Provisioning is controlled by the following parameters firmware upgrades are discussed later in this section Provision_Enable Resync_On_Reset Resync_Random_Delay Resync_Periodic Resync_Error_Retry_Delay Forced_Resync_Delay Resync_From_SIP Resync_After_Upgrade_Attempt Resync_Trigger_1 Resync_Trigger_2 Resync_Fails_On_FNF Profile Rule Profile _Rule_B Profile _Rule_C Profile _Rule_D Log_Resync_Request_Msg Log_Resync_Success_Msg Log_Resync_Failure_Msg 2004 Linksys Proprietary See Copyright Notice on Page 2 35 e GPP A through GPP_P e GPP_SA through GPP_SD Provision Enable ParName Provision_Enable Default Enable The CFG profile must be requested by the PHONE ADAPTER and cannot be pushed from a provisioning server although a service provider can effectively push a profile by triggering the request operation remotely via a SIP NOTIFY The functionality is controlled by the Provision_Enable parameter The parameter enables the functionality encompassed by the remaining provisioning parameters In addition Provision_Enable also gates the ability to issue an expl
188. to non existent Static Redundancy A relatively simple way to support proxy redundancy is to configure a static list of SIP proxy servers to the PHONE ADAPTER in its configuration profile where the list is arranged in some order of priority The PHONE ADAPTER will attempt to contact the highest priority proxy server whenever possible When the currently selected proxy server is not responding the PHONE ADAPTER automatically retries the next proxy server in the list 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 PHONE ADAPTER One solution to this situation is through the use DNS SRV records The PHONE ADAPTER can be instructed to contact a SIP proxy server in a domain named in SIP messages The PHONE ADAPTER 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 PHONE ADAPTER shall try to contact the list of hosts in the order of their stated priority 7 2 1 2 Re registration with Primary SIP Proxy Server If the PHONE ADAPTER is currently using a lower prior
189. try_Delay Default 3600 If a resync attempt fails the PHONE ADAPTER will retry with a delay indicated by the Resync_Error_Retry_Delay parameter specified in seconds If the value is zero the PHONE ADAPTER treats resync failures as though they were successful and simply waits for the next periodic event to resync Resync From SIP ParName Resync_From_SIP Default Enable Resync_From_SIP gates the ability of a service provider to trigger a profile resync via a SIP NOTIFY message to the PHONE ADAPTER If the PHONE ADAPTER receives a SIP NOTIFY request with an Event header field value of resync reboot or restart the PHONE ADAPTER will attempt to Digest authenticate the notifier using the authentication password used for registrations on that line if the Auth_Resync Reboot parameter is set If this parameter is not set or if the NOTIFY request is authenticated the PHONE ADAPTER triggers a resync cold boot or warm boot respectively The actual resync reboot or restart will not take place until the PHONE ADAPTER is idle i e no calls are in progress Profile Rule ParName Profile Rule Default spa PSN cfg ParName Profile Rule_B through Profile _Rule_D Default Empty The Profile_Rule parameter is a script that identifies the provisioning server to contact when performing a profile resync The Profile _Rule_B Profile Rule_C and Profile Rule_D parameters are also scripts used
190. tures including parameters requirements and contingencies please refer the section PHONE ADAPTER Feature Configuration Parameters section Error Reference source not found 7 1 Data Networking Features 7 1 1 MAC Address IEEE 802 3 2004 Linksys Proprietary See Copyright Notice on Page 2 95 7 1 2 IPv4 Internet Protocol Version A RFC 791 upgradeable to v6 RFC 1883 7 1 3 ARP Address Resolution Protocol 7 1 4 DNS A Record RFC 1706 SRV Record RFC 2782 7 1 5 DiffServ RFC 2475 and ToS Type of Service RFC 791 1349 7 1 6 DHCP Client Dynamic Host Configuration Protocol RFC 2131 7 1 7 ICMP Internet Control Message Protocol RFC792 7 1 8 TCP Transmission Control Protocol RFC793 7 1 9 UDP User Datagram Protocol RFC768 7 1 10 RTP Real Time Protocol RFC 1889 RFC 1890 7 1 11 RTCP Real Time Control Protocol RFC 1889 7 2 Voice Features 7 2 1 SIPv2 Session Initiation Protocol Version 2 RFC 3261 3265 7 2 1 1 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 average SIP proxy server may handle 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 PHONE ADAPTER supports the use of backup SIP proxy servers so that service disruption should be next
191. uccessful Hang up Unit will reboot and all configuration parameters will be reset to service provider data factory default values Note If the Administrator password is not set or the user is allowed to change it the items marked with Requires Password will not require a password 4 Configuration Parameters 4 1 Data Types The data types for the PHONE ADAPTER configuration parameters are described below e Uns lt n gt 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 e Sig lt n gt 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 e Stren gt A generic string with up to n non reserved characters e Float lt n gt A floating point value with up to n decimal places Time lt n gt 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 a b c A choice among a be 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 UR
192. vice during the initial HTTPS provisioning stage As an alternative method for initial configuration an unprovisioned PHONE ADAPTER can receive an encrypted profile specifically targeted for that device without requiring an explicit key although this is not as secure as using HTTPS The PHONE ADAPTER can be configured to resync its internal configuration state to a remote profile periodically and on power up An administrator can also remotely trigger a profile resync by sending an authenticated SIP NOTIFY request to the PHONE ADAPTER Likewise remote upgrades are achieved via TFTP HTTP or HTTPS The PHONE ADAPTER upgrade logic is capable of automating multi stage upgrades in case intermediate upgrades are ever required to reach a future upgrade state from an older release General purpose parameters are provided as an additional aid to service providers in managing the provisioning process The administrator can configure simple comparisons translations concatenations and parameter substitution with the aid of these parameters All profile resyncs are attempted only when the PHONE ADAPTER is idle since they may trigger a software reboot User intervention is not required to initiate or complete a profile update or firmware upgrade In general most configuration changes take effect without requiring a reboot 2004 Linksys Proprietary See Copyright Notice on Page 2 20 The PHONE ADAPTER also provides a Web Interface with two level acc
193. y DNS DNS server used by PHONE ADAPTER in addition to IP 0 0 0 0 DHCP supplied DNS servers if DHCP is enabled when DHCP is disabled this will be the primary DNS server Secondary DNS DNS server used by PHONE ADAPTER in addition to IP 0 0 0 0 DHCP supplied DNS servers if DHCP is enabled when DHCP is disabled this will be the secondary DNS server 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 PHONE ADAPTER system information and critical events Debug Server The debug server name and port This feature FQDN specifies the server for logging PHONE ADAPTER 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 0 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 2004 Linksys Proprietary See Copyright Notice on Page 2 47 Parallel DNS query mode PHONE ADAPTER will send the same request to all the DNS servers at the same time when doing a DNS lookup the first incoming reply will be accepted by PHONE ADAPTER Tolog SIP messages Debug Level must be set to at least 2 lf both Debug Server and Syslog Server are s
194. y can still receive audio while the call is holding the PHONE ADAPTER can be setup to contact an auto answering SAS as described in Section 4 and have it stream audio to the holding party When used this way the SAS is referred to as a MOH Server 2004 Linksys Proprietary See Copyright Notice on Page 2 100 PAI IP 192 168 2 100 User ID 1 1001 SIP Port 1 5060 a User ID 2 1002 SIP Port 2 5061 Phone 1 1 Phone 2 CD Player Radio ete PA2 IP 192 168 2 200 User ID 1 2001 SIP Port 1 5060 ys H User ID 2 2002 SIP Port 2 5061 Phone 2 Example configuration for MOH application with a PHONE ADAPTER line configured as a SAS SAS Configuration Examples The following configuration examples are based on the setup as depicted in Figure Example 1 SAS Line not registered with the Proxy Server for the other subscribers On PHONE ADAPTER 1 SAS Enable 1 no MOH Server 1 1002 192 168 2 100 5061 or 1002 127 0 0 1 5061 SAS Enable 2 yes On PHONE ADAPTER 2 SAS Enable 1 no MOH Server 1 1002 192 168 2 100 5061 SAS Enable 2 no MOH Server 2 1002 192 168 2 100 5061 Example 2 SAS Line registered with the Proxy Server as the other subscribers On PHONE ADAPTER 1 SAS Enable 1 no MOH Server 1 1002 2004 Linksys Proprietary See Copyright Notice on Page 2 101 SAS Enable 2 yes On PHONE ADAPTER 2 SAS Enable 1 no MOH Server 1 1002 SAS Enab
195. ys Proprietary See Copyright Notice on Page 2 50 Ext SIP Port External port to substitute for the actual SIP port of Port 0 the unit in all outgoing SIP messages If O is specified no SIP port substitution is performed Ext RTP Port Min External port mapping of lt RTP Port Min gt If this Port 0 value is non zero the RTP port number in all outgoing SIP messages is substituted by the corresponding port value in the external RTP port range NAT Mapping Enable Enable the use of externally mapped of IP address Bool No and SIP RTP ports in SIP messages The mapping may be discovered by any of the supported methods NAT Keep Alive Enable If set to yes the configured lt NAT Keep Alive Bool No Msg gt is sent periodically every lt NAT Keep Alive Intvl gt seconds NAT Keep Alive Msg Contents of the keep alive message to be senttoa Str31 NOTIFY given 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 NAT Keep Alive Dest Destination to send NAT keep alive messages to If FQDN PROXY value is PROXY it will be sent to the current proxy or outbound proxy 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

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