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Yealink SIP-T48G [135/465] Intercom
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Configuring Basic Features
121
3. Enter the specified DTMF digits in the Tran Send DTMF field.
4. Click Confirm to accept the change.
Intercom allows establishing an audio conversation directly. The IP phone can answer
intercom calls automatically. This feature depends on support from a SIP server.
Intercom is a useful feature in office environments to quickly connect with an operator or
secretary. Users can press an intercom key to automatically initiate an outgoing
intercom call with a remote extension.
Procedure
Intercom key can be configured using the configuration files or locally.
Configuration File
<y0000000000xx>.cfg
Assign an intercom key.
For more information, refer to
Intercom Key on page 390.
Local
Web User Interface
Assign an intercom key.
Navigate to:
http://<phoneIPAddress>/servlet
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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