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Yealink SIP-T40P Инструкция по эксплуатации онлайн [14/465] 166937
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Содержание
- We are striving to improve our documentation quality and we appreciate your feedback email 3
- Your opinions and comments to docsfeedback yealink com 3
- About this guide 5
- Documentations 5
- In this guide 5
- Changes for release 72 guide version 72 6
- Summary of changes 6
- Changes for release 71 guide version 71 50 7
- Changes for release 71 guide version 71 70 7
- Changes for release 71 guide version 71 71 7
- Changes for release 71 guide version 71 80 7
- Changes for release 71 guide version 71 81 7
- Changes for release 71 guide version 71 0 8
- About this guide v 9
- Getting started 9 9
- Product overview 1 9
- Table of contents 9
- Table of contents ix 9
- Configuring basic features 35 10
- Configuring advanced features 25 11
- Configuring audio features 91 11
- Configuring security features 03 12
- Resource files 19 12
- Troubleshooting 31 12
- Upgrading firmware 15 12
- Appendix 43 13
- Index 49 13
- Product overview 15
- Voip principle 15
- Sip components 16
- User agent client uac 16
- Sip ip phone models 17
- User agent server uas 17
- Physical features 18
- Physical features of sip t4x ip phones 18
- Sip t46g 18
- Physical features 19
- Sip t42g 19
- Key features of sip t4x ip phones 20
- Phone features 20
- Physical features 20
- Sip t41p 20
- Codecs and voice features 21
- Management 21
- Network features 21
- Security 21
- Connecting the ip phone 23
- Getting started 23
- Initialization process overview 26
- Configuration methods 28
- Phone user interface 28
- Verifying startup 28
- Web user interface 28
- Configuration files 29
- Reading icons 30
- Configuring basic network parameters 32
- Dhcp option 32
- Procedure 33
- Configuring network parameters manually 35
- Procedure 36
- Procedure 38
- Configuring transmission methods of the internet port and pc port 39
- Auto negotiation 40
- Full duplex 40
- Half duplex 40
- Procedure 41
- Creating dial plan 42
- Procedure 43
- Replace rule 43
- Delay time for dial now rule 44
- Dial now 44
- Procedure 44
- Area code 46
- Procedure 46
- Block out 47
- Procedure 47
- Administrator password 49
- Anonymous call 49
- Anonymous call rejection 49
- Auto answer 49
- Auto redial 49
- Backlight 49
- Basic features 49
- Busy tone delay 49
- Call completion 49
- Call log 49
- Call waiting 49
- Configuring basic features 49
- Contrast 49
- Do not disturb 49
- Early media 49
- Hotline 49
- Key as send 49
- Language 49
- Live dialpad 49
- Local directory 49
- Logo customization 49
- Missed call log 49
- Phone lock 49
- Power indicator led 49
- Return code when refuse 49
- Ring workaround 49
- Sip session timer 49
- Softkey layout 49
- This chapter provides information for making configuration changes for the following 49
- Time and date 49
- Use outbound proxy in dialog 49
- User password 49
- Power indicator led 50
- Procedure 51
- Contrast 52
- Procedure 52
- Procedure 53
- Wallpaper 53
- Backlight 55
- Procedure 55
- Procedure 56
- User password 56
- Administrator password 57
- Procedure 57
- Phone lock 58
- Procedure 59
- Daylight saving time 61
- Time and date 61
- Time zone 61
- Procedure 62
- Language 66
- Loading language packs 67
- Procedure 67
- Procedure 68
- Specifying the language to use 68
- Logo customization 69
- Procedure 69
- Softkey layout 70
- Procedure 72
- Key as send 73
- Procedure 73
- Hotline 75
- Procedure 75
- Call log 77
- Procedure 77
- Missed call log 78
- Procedure 78
- Local directory 79
- Procedure 79
- Call waiting 82
- Live dialpad 82
- Procedure 82
- Procedure 83
- Auto redial 84
- Procedure 85
- Auto answer 86
- Procedure 86
- Call completion 88
- Procedure 88
- Anonymous call 89
- Procedure 90
- Anonymous call rejection 91
- Procedure 92
- Do not disturb 93
- Procedure 93
- Return message when dnd 93
- Busy tone delay 97
- Procedure 97
- Procedure 98
- Return code when refuse 98
- Early media 99
- Procedure 100
- Ring workaround 100
- Procedure 101
- Use outbound proxy in dialog 101
- Procedure 102
- Sip session timer 102
- Procedure 103
- Session timer 103
- Call hold 104
- Procedure 105
- Call forward 106
- Forward international 107
- Procedure 107
- Call transfer 112
- Procedure 113
- Network conference 114
- Procedure 114
- Procedure 115
- Transfer on conference hang up 115
- Directed call pickup 116
- Procedure 116
- Group call pickup 119
- Procedure 120
- Dialog info call pickup 122
- Procedure 123
- Call return 124
- Procedure 124
- Call park 125
- Procedure 125
- Web server type 126
- Procedure 127
- Calling line identification presentation 128
- Procedure 128
- Connected line identification presentation 129
- Procedure 129
- Inband 130
- Rfc 2833 130
- Procedure 131
- Sip info 131
- Procedure 133
- Suppress dtmf display 133
- Procedure 134
- Transfer via dtmf 134
- Intercom 135
- Outgoing intercom calls 135
- Procedure 135
- Incoming intercom calls 136
- Procedure 137
- Action uri 139
- Action url 139
- Advanced features 139
- Automatic call distribution 139
- Busy lamp field 139
- Call recording 139
- Configuring advanced features 139
- Distinctive ring tones 139
- Distinctive ring tones allows particular incoming calls to trigger ip phones to play 139
- Distinctive ring tones the ip phone inspects the invite request for an alert info header 139
- Hot desking 139
- Ipv6 support 139
- Message waiting indicator 139
- Multicast paging 139
- Music on hold 139
- Network address translation 139
- Quality of service 139
- Remote phone book 139
- Ring tone 139
- Server redundancy 139
- The ip phone strips out the url and keyword parameter and maps it to the appropriate 139
- This chapter provides information for making configuration changes for the following 139
- Tr 069 device management 139
- When receiving an incoming call if the invite request contains an alert info header 139
- X authentication 139
- Procedure 141
- Australia 143
- Austria 143
- Belgium 143
- Brazil 143
- Configuring advanced features 143
- Czech etsi 143
- Denmark 143
- Finland 143
- France 143
- Germany 143
- Great britain 143
- Greece 143
- Hungary 143
- Indicate different conditions of the ip phone the default tones used on ip phones are 143
- Lithuania 143
- Mexico 143
- Netherlands 143
- New zealand 143
- Norway 143
- Portugal 143
- Russia 143
- Sweden 143
- Switzerland 143
- The us tone sets available tone sets for ip phones 143
- United states 143
- When receiving a message or recording a call the ip phone will play a warning tone 143
- You can customize tones or select specialized tone sets vary from country to country to 143
- Procedure 144
- Procedure 145
- Remote phone book 145
- Ldap attributes 148
- Procedure 148
- Busy lamp field 150
- Visual alert and audio alert for blf pickup 150
- Blf led mode 151
- Procedure 152
- Music on hold 155
- Procedure 155
- Automatic call distribution 156
- Procedure 157
- Message waiting indicator 158
- Procedure 159
- Multicast paging 161
- Procedure 161
- Sending rtp stream 161
- Procedure 163
- Receiving rtp stream 163
- Call recording 165
- Record 165
- Administrator s guide for sip t4x ip phones 166
- An http get message to the server 166
- Example of a 200 ok message 166
- Example of a sip info message 166
- Example of an http get message 166
- If the recording is successfully started the server will respond with a 200 ok message 166
- Info message to the server with the specific header record off and then the 166
- Recording stops 166
- Url record 166
- When a user presses a url record key for the first time during a call the ip phone sends 166
- When the user presses the record key for the second time the ip phone sends a sip 166
- Procedure 167
- Hot desking 169
- Procedure 169
- Action url 170
- Action uri 173
- Procedure 173
- Procedure 175
- Phone configuration for redundancy implementation 176
- Server redundancy 176
- Phone registration 177
- Procedure 178
- Sip server domain name resolution 178
- Outgoing call when the working server connection fails 180
- Procedure 180
- Lldp med media endpoint discovery 182
- Procedure 183
- Procedure 185
- Vlan discovery via dhcp 185
- Procedure 188
- Quality of service 190
- Procedure 191
- Sip qos 191
- Voice qos 191
- Nat traversal 192
- Network address translation 192
- Stun simple traversal of udp over nats 192
- Procedure 193
- Procedure 194
- X authentication 194
- Tr 069 device management 199
- Procedure 200
- Ipv6 address assignment method 201
- Ipv6 support 201
- Procedure 202
- Configuring audio features 205
- Headset prior 205
- Procedure 205
- Dual headset 206
- Procedure 206
- Audio codecs 207
- Packetization time 208
- Procedure 209
- Acoustic clarity technology 211
- Acoustic echo cancellation 211
- Procedure 211
- Procedure 212
- Voice activity detection 212
- Comfort noise generation 213
- Procedure 213
- Jitter buffer 214
- Procedure 214
- Configuring security features 217
- Transport layer security 217
- Administrator s guide for sip t4x ip phones 218
- Aes128 sha 218
- Aes256 sha 218
- Des cbc sha 218
- Des cbc3 sha 218
- Dhe dss aes128 sha 218
- Dhe dss rc4 sha 218
- Dhe rsa aes128 sha 218
- Edh dss des cbc sha 218
- Edh dss des cbc3 sha 218
- Edh rsa des cbc sha 218
- Edh rsa des cbc3 sha 218
- Exp des cbc sha 218
- Exp edh dss des cbc sha 218
- Exp edh rsa des cbc sha 218
- Exp rc4 md5 218
- Exp1024 des cbc sha 218
- Exp1024 dhe dss des cbc sha 218
- Exp1024 dhe dss rc4 sha 218
- Exp1024 rc4 md5 218
- Exp1024 rc4 sha 218
- Idea cbc sha 218
- Public key information in server key exchange message and concludes its part of the 218
- Rc4 md5 218
- Rc4 sha 218
- Step1 ip phone sends client hello message proposing ssl options 218
- Step2 server responds with server hello message selecting the ssl options sends its 218
- The following figure illustrates the tls messages exchanged between the ip phone and 218
- Tls server to establish an encrypted communication channel 218
- Certificates 219
- Procedure 220
- Secure real time transport protocol 223
- Procedure 224
- Encrypting configuration files 225
- Procedure to encrypt configuration files 226
- Procedure 227
- Upgrade via web user interface 229
- Upgrading firmware 229
- Procedure 230
- Upgrade firmware from the provisioning server 230
- Replace rule template 233
- Resource files 233
- Dial now template 234
- Procedure 234
- Procedure 235
- Softkey layout template 235
- Procedure 236
- Local contact file 237
- Procedure 237
- Remote xml phone book 238
- Procedure 239
- Directory template 240
- Procedure 240
- Procedure 241
- Super search template 241
- Specifying the access url of resource files 242
- Troubleshooting 245
- Troubleshooting methods 245
- Viewing log files 245
- Capturing packets 248
- Enabling the watch dog feature 248
- Analyzing configuration files 249
- Getting information from status indicators 249
- Troubleshooting solutions 250
- Why doesn t the ip phone get an ip address 250
- Why is the lcd screen blank 250
- How do i find the basic information of the ip phone 251
- Why do i get poor sound quality during a call 251
- Why doesn t the ip phone display time and date correctly 251
- Why doesn t the ip phone upgrade firmware successfully 251
- How to increase or decrease the volume 252
- How to reboot ip phone remotely 252
- What is the difference between a remote phone book and a local phonebook 252
- What is the difference between user name register name and display name 252
- What do on code and off code mean 253
- What is auto provisioning 253
- What is pnp 253
- What will happen if i connect both poe cable and power adapter which has the higher priority 253
- Why doesn t the ip phone update the configuration 253
- How to reset your phone to factory configurations 254
- How to solve the ip conflict problem 254
- How to restore the administrator password 255
- Appendix 257
- Appendix a glossary 257
- Appendix b time zones 259
- Appendix c configuration parameters 262
- Basic and advanced feature parameters 262
- Setting parameters in configuration files 262
- Static network settings 263
- Internet and pc ports transmission methods 267
- Internet por 267
- Pc port transmission method 267
- Dial plan 268
- Replace rule 268
- Dial now 269
- Area code 270
- Block out 271
- Power indicator led 272
- Contrast 275
- Backlight 276
- Administrator password 277
- User password 277
- Phone lock 278
- Time and date 280
- Language 286
- Logo customization 287
- Key as send 288
- Hotline 290
- Call log 291
- Missed call log 291
- Call waiting 292
- Live dialpad 292
- Auto redial 294
- Auto answer 295
- Anonymous call 296
- Call completion 296
- Anonymous call rejection 298
- Dnd mode 300
- Do not disturb 300
- Return message when dnd 300
- Dnd in phone mode 301
- Dnd in custom mode 302
- Busy tone delay 303
- Return code when refuse 303
- Ring workaround 304
- Use outbound proxy in dialog 304
- Sip session timer 305
- Session timer 306
- Call hold 307
- Call forward 308
- Call forward mode 308
- Always forward 309
- Call forward in phone mode 309
- Busy forward 310
- No answer forward 311
- Always forward 313
- Call forward in custom mode 313
- Busy forward 315
- No answer forward 316
- Call transfer 318
- Fwd international 318
- Network conference 319
- Transfer on conference hang up 320
- Directed call pickup 321
- Phone basis 321
- Group call pickup 322
- Per line basis 322
- Phone basis 322
- Dialog info call pickup 323
- Per line basis 323
- Web server type 324
- Calling line identification presentation 325
- Connected line identification presentation 326
- Suppress dtmf display 329
- Transfer via dtmf 329
- Incoming intercom calls 330
- Distinctive ring tones 332
- Appendix 335
- Belgium 335
- Configuration file 335
- Customizes the tone for each condition 335
- Czech etsi 335
- Denmark 335
- Description 335
- Example voice tone country austria 335
- Finland 335
- France 335
- Germany 335
- Great britain 335
- Greece 335
- Hungary 335
- Lithuania 335
- Mexico 335
- Netherlands 335
- New zealand 335
- Norway 335
- Parameter 335
- Portugal 335
- Russia 335
- Sweden 335
- Switzerland 335
- The parameter voice tone message is not 335
- United states 335
- Voice tone autoanswer 335
- Voice tone busy 335
- Voice tone callwaiting 335
- Voice tone congestion 335
- Voice tone dial 335
- Voice tone dialrecall 335
- Voice tone info 335
- Voice tone message 335
- Voice tone record 335
- Voice tone ring 335
- Voice tone stutter 335
- Remote phone book 336
- Visual and audio alert for blf pickup 343
- Blf led mode 344
- Music on hold 344
- Message waiting indicator 346
- Receiving rtp stream 348
- Sending rtp stream 348
- Action url 350
- Action uri 351
- Server redundancy 352
- Fallback mode 354
- Failover mode 355
- Sip server domain name resolution 357
- Internet port 358
- Pc port 360
- Dhcp vlan discovery 361
- Network address translation 363
- Tr 069 366
- Audio feature parameters 373
- Headset prior 373
- Audio codecs 374
- Dual headset 374
- Acoustic echo cancellation 377
- Comfort noise generation 378
- Jitter buffer 378
- Voice activity detection 378
- Security feature parameters 380
- Uploading certificates 382
- Configuring decryption method 383
- Upgrading firmware 385
- Access url of dial now template 388
- Access url of replace rule template 388
- Resource files 388
- Access url of softkey layout template 389
- Access url of local contact file 391
- Access url of directory template 392
- Access url of remote xml phone book 392
- Access url of super search template 392
- Access url of wallpaper image 393
- Log settings 393
- Troubleshooting 393
- Watch dog 394
- Appendix 395
- Call park 395
- Call return 395
- Conference 395
- Configuration file 395
- Configures the key feature for the line key 395
- Configuring dss key 395
- Description 395
- Detailed in the following 395
- Directed pickup 395
- Disabled 395
- Dss key can be assigned with various key features the parameters of the dss key are 395
- Enabled 395
- Example watch_dog enable 1 395
- Forward 395
- Group listening 395
- Group pickup 395
- Hot desking 395
- Intercom 395
- Line default for line key 1 6 of sip t46g 395
- Linekey x type 395
- Multicast paging 395
- N a default to line key 7 27 for 395
- Or line key 1 3 of sip t42g t41p 395
- Parameter 395
- Sip t42g t41p 395
- Sip t46g or line key 4 15 for 395
- Sms not applicable to sip t42g t41p 395
- Speed dial 395
- This section provides dss key parameters you can configure on the ip phone 395
- Transfer 395
- Valid types are 395
- Voice mail 395
- X ranges from 1 to 15 for sip t42g t41p 395
- X ranges from 1 to 27 for sip t46g 395
- Xml group 395
- Dnd key 400
- Keypad lock key 400
- Directed call pickup key 401
- Group call pickup key 402
- Call park key 403
- Call return key 403
- Intercom key 404
- Ldap key 405
- Blf key 406
- Acd key 407
- Multicast paging key 408
- Record key 408
- Hot desking key 409
- Url record key 409
- Appendix d sip session initiation protocol 410
- Rfc and internet draft support 410
- Appendix 411
- Bandwidth 411
- Network address translators nats 411
- Registration 411
- Rfc 3262 reliability of provisional responses in the session initiation protocol sip 411
- Rfc 3263 session initiation protocol sip locating sip servers 411
- Rfc 3264 an offer answer model with the session description protocol sdp 411
- Rfc 3265 session initiation protocol sip specific event notification 411
- Rfc 3266 support for ipv6 in session description protocol sdp 411
- Rfc 3310 http digest authentication using authentication and key agreement 411
- Rfc 3311 the session initiation protocol sip update method 411
- Rfc 3312 integration of resource management and sip 411
- Rfc 3313 private sip extensions for media authorization 411
- Rfc 3323 a privacy mechanism for the session initiation protocol sip 411
- Rfc 3324 requirements for network asserted identity 411
- Rfc 3325 sip asserted identity 411
- Rfc 3326 the reason header field for the session initiation protocol sip 411
- Rfc 3361 dhcp for ipv4 option for sip servers 411
- Rfc 3372 sip for telephones sip t context and architectures 411
- Rfc 3420 internet media type message sipfrag 411
- Rfc 3428 session initiation protocol sip extension for instant messaging 411
- Rfc 3455 private header p header extensions to the sip for the 3gpp 411
- Rfc 3486 compressing the session initiation protocol sip 411
- Rfc 3489 stun simple traversal of user datagram protocol udp through 411
- Rfc 3515 the session initiation protocol sip refer method 411
- Rfc 3550 rtp rtcp ietf rfc 3550 411
- Rfc 3556 session description protocol sdp bandwidth modifiers for rtcp 411
- Rfc 3581 an extension to the sip for symmetric response routing 411
- Rfc 3608 sip extension header field for service route discovery during 411
- Rfc 3665 session initiation protocol sip basic call flow examples 411
- Rfc 3666 sip public switched telephone network pstn call flows 411
- Rfc 3680 sip event package for registrations 411
- Rfc 3702 authentication authorization and accounting requirements for the sip 411
- Rfc 3711 the secure real time transport protocol srtp 411
- Rfc 3725 best current practices for third party call control 3pcc in the session 411
- Administrator s guide for sip t4x ip phones 412
- Draft ietf sip cc transfer 05 txt sip call control transfer 412
- Draft levy sip diversion 04 txt diversion indication in sip 412
- For the session initiation protocol sip 412
- Identifier uri parameter registry for the session initiation protocol sip 412
- Initiation protocol sip 412
- Parameter registry for the session initiation protocol sip 412
- Protocol sip 412
- Rfc 3842 a message summary and message waiting indication event package 412
- Rfc 3856 a presence event package for session initiation protocol sip 412
- Rfc 3890 a transport independent bandwidth modifier for the sdp 412
- Rfc 3891 the session initiation protocol sip replaces header 412
- Rfc 3892 the session initiation protocol sip referred by mechanism 412
- Rfc 3959 the early session disposition type for sip 412
- Rfc 3960 early media and ringing tone generation in sip 412
- Rfc 3968 the internet assigned number authority iana header field 412
- Rfc 3969 the internet assigned number authority iana uniform resource 412
- Rfc 4028 session timers in the session initiation protocol sip 412
- Rfc 4235 an invite initiated dialog event package for the session initiation 412
- Rfc 4244 an extension to the sip for request history information 412
- Rfc 4317 session description protocol sdp offer answer examples 412
- Rfc 4353 a framework for conferencing with the sip 412
- Rfc 4475 session initiation protocol sip torture 412
- Rfc 4485 guidelines for authors of extensions to the sip 412
- Rfc 4504 sip telephony device requirements and configuration 412
- Rfc 4566 sdp session description protocol 412
- Rfc 4568 session description protocol sdp security descriptions for media 412
- Rfc 4575 a sip event package for conference state 412
- Rfc 4579 sip call control conferencing for user agents 412
- Rfc 4662 a sip event notification extension for resource lists 412
- Rfc 5009 p early media header 412
- Rfc 5079 rejecting anonymous requests in sip 412
- Rfc 5359 session initiation protocol service examples 412
- Rfc 5589 session initiation protocol sip call control transfer 412
- Rfc3966 the tel uri for telephone number 412
- Streams 412
- Sip request 413
- Sip header 414
- Sip responses 415
- Xx response information responses 415
- Xx response redirection responses 415
- Xx response successful responses 415
- Xx response request failure responses 416
- Sip session description protocol sdp usage 417
- Xx response global responses 417
- Xx response server failure responses 417
- Appendix e sip call flows 418
- Successful call setup and disconnect 419
- Unsuccessful call setup called user is busy 421
- F1 invite b 422
- F2 invite b 422
- F3 100 trying 422
- F4 100 trying 422
- F5 486 busy here 422
- F6 486 busy here 422
- F7 ack 422
- F8 ack 422
- User a proxy server user b 422
- F1 invite b 424
- F2 invite b 424
- F3 180 ringing 424
- F4 180 ringing 424
- F5 cancel 424
- F6 cancel 424
- F7 200 ok 424
- F8 200 ok 424
- Unsuccessful call setup called user does not answer 424
- User a proxy server user b 424
- Successful call setup and call hold 426
- Successful call setup and call waiting 428
- Call transfer without consultation 433
- Administrator s guide for sip t4x ip phones 434
- Call is established between user a and user c 434
- User c answers the call 434
- Call transfer with consultation 437
- Administrator s guide for sip t4x ip phones 438
- Call is established between user b and user c 438
- User a transfers the call to user c 438
- Always call forward 442
- Appendix 443
- Call is established between user a and user c 443
- User c answers the call 443
- Busy call forward 446
- No answer call forward 449
- Call conference 452
- Appendix f sample configuration file 457
- Dial plan settings 457
- Network settings 457
- T4x sample configuration file 457
- Auto dst settings 458
- Auto redial 458
- Call hold 458
- Call waiting 458
- Language 458
- Phone lock 458
- Time settings 458
- Call forward 459
- Dtmf suppression 459
- Hotline 459
- In custom mode 459
- In phone mode 459
- Web server type 459
- Call conference 460
- Call transfer 460
- Distinctive ring tones 460
- Remote phone book 460
- Action url 461
- Access url of resource files 462
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