Yealink SIP-T46G — настройка основных функций IP-телефонов для интеркома [137/465]
![Yealink SIP-T40P [137/465] Procedure](/views2/1192044/page137/bg89.png)
Configuring Basic Features
123
Intercom Mute
Intercom Mute allows the IP phone to mute the microphone for incoming intercom calls.
Intercom Tone
Intercom Tone allows the IP phone to play a warning tone before answering an intercom
call.
Intercom Barge
Intercom Barge allows the IP phone to automatically answer an incoming intercom call
while an active call is in progress. The active call will be placed on hold.
Procedure
Incoming intercom calls can be configured using the configuration files or locally.
Configuration File
<y0000000000xx>.cfg
Configure the incoming intercom
call feature.
For more information, refer to
Incoming Intercom calls on page
316.
Local
Web User Interface
Configure the incoming intercom
call feature.
Navigate to:
http://<phoneIPAddress>/servlet
?p=features-intercom&q=load
Phone User Interface
Configure the incoming intercom
call feature.
To configure intercom via web user interface:
1. Click on Features->Intercom.
Содержание
- Your opinions and comments to docsfeedback yealink com p.3
- We are striving to improve our documentation quality and we appreciate your feedback email p.3
- About this guide p.5
- In this guide p.5
- Documentations p.5
- Summary of changes p.6
- Changes for release 72 guide version 72 p.6
- Changes for release 71 guide version 71 81 p.7
- Changes for release 71 guide version 71 80 p.7
- Changes for release 71 guide version 71 71 p.7
- Changes for release 71 guide version 71 70 p.7
- Changes for release 71 guide version 71 50 p.7
- Changes for release 71 guide version 71 0 p.8
- Table of contents ix p.9
- Table of contents p.9
- Product overview 1 p.9
- Getting started 9 p.9
- About this guide v p.9
- Configuring basic features 35 p.10
- Configuring audio features 91 p.11
- Configuring advanced features 25 p.11
- Resource files 19 p.12
- Troubleshooting 31 p.12
- Configuring security features 03 p.12
- Upgrading firmware 15 p.12
- Index 49 p.13
- Appendix 43 p.13
- Voip principle p.15
- Product overview p.15
- User agent client uac p.16
- Sip components p.16
- User agent server uas p.17
- Sip ip phone models p.17
- Sip t46g p.18
- Physical features of sip t4x ip phones p.18
- Physical features p.18
- Sip t42g p.19
- Physical features p.19
- Key features of sip t4x ip phones p.20
- Sip t41p p.20
- Physical features p.20
- Phone features p.20
- Network features p.21
- Security p.21
- Management p.21
- Codecs and voice features p.21
- Getting started p.23
- Connecting the ip phone p.23
- Initialization process overview p.26
- Web user interface p.28
- Verifying startup p.28
- Phone user interface p.28
- Configuration methods p.28
- Configuration files p.29
- Reading icons p.30
- Dhcp option p.32
- Configuring basic network parameters p.32
- Procedure p.33
- Configuring network parameters manually p.35
- Procedure p.36
- Procedure p.38
- Configuring transmission methods of the internet port and pc port p.39
- Half duplex p.40
- Full duplex p.40
- Auto negotiation p.40
- Procedure p.41
- Creating dial plan p.42
- Replace rule p.43
- Procedure p.43
- Dial now p.44
- Delay time for dial now rule p.44
- Procedure p.44
- Procedure p.46
- Area code p.46
- Procedure p.47
- Block out p.47
- Auto redial p.49
- Administrator password p.49
- Early media p.49
- Sip session timer p.49
- Call waiting p.49
- Ring workaround p.49
- Call log p.49
- Return code when refuse p.49
- Call completion p.49
- Power indicator led p.49
- Busy tone delay p.49
- Phone lock p.49
- Basic features p.49
- Missed call log p.49
- Backlight p.49
- Logo customization p.49
- Local directory p.49
- Auto answer p.49
- Live dialpad p.49
- Anonymous call rejection p.49
- Language p.49
- Anonymous call p.49
- Key as send p.49
- User password p.49
- Hotline p.49
- Use outbound proxy in dialog p.49
- Time and date p.49
- Do not disturb p.49
- This chapter provides information for making configuration changes for the following p.49
- Contrast p.49
- Softkey layout p.49
- Configuring basic features p.49
- Power indicator led p.50
- Procedure p.51
- Procedure p.52
- Contrast p.52
- Wallpaper p.53
- Procedure p.53
- Procedure p.55
- Backlight p.55
- User password p.56
- Procedure p.56
- Administrator password p.57
- Procedure p.57
- Phone lock p.58
- Procedure p.59
- Time zone p.61
- Time and date p.61
- Daylight saving time p.61
- Procedure p.62
- Language p.66
- Procedure p.67
- Loading language packs p.67
- Specifying the language to use p.68
- Procedure p.68
- Procedure p.69
- Logo customization p.69
- Softkey layout p.70
- Procedure p.72
- Procedure p.73
- Key as send p.73
- Procedure p.75
- Hotline p.75
- Procedure p.77
- Call log p.77
- Procedure p.78
- Missed call log p.78
- Procedure p.79
- Local directory p.79
- Procedure p.82
- Live dialpad p.82
- Call waiting p.82
- Procedure p.83
- Auto redial p.84
- Procedure p.85
- Procedure p.86
- Auto answer p.86
- Procedure p.88
- Call completion p.88
- Anonymous call p.89
- Procedure p.90
- Anonymous call rejection p.91
- Procedure p.92
- Return message when dnd p.93
- Procedure p.93
- Do not disturb p.93
- Procedure p.97
- Busy tone delay p.97
- Return code when refuse p.98
- Procedure p.98
- Early media p.99
- Ring workaround p.100
- Procedure p.100
- Use outbound proxy in dialog p.101
- Procedure p.101
- Sip session timer p.102
- Procedure p.102
- Session timer p.103
- Procedure p.103
- Call hold p.104
- Procedure p.105
- Call forward p.106
- Procedure p.107
- Forward international p.107
- Call transfer p.112
- Procedure p.113
- Network conference p.114
- Procedure p.114
- Transfer on conference hang up p.115
- Procedure p.115
- Procedure p.116
- Directed call pickup p.116
- Group call pickup p.119
- Procedure p.120
- Dialog info call pickup p.122
- Procedure p.123
- Procedure p.124
- Call return p.124
- Procedure p.125
- Call park p.125
- Web server type p.126
- Procedure p.127
- Calling line identification presentation p.128
- Procedure p.128
- Procedure p.129
- Connected line identification presentation p.129
- Rfc 2833 p.130
- Inband p.130
- Sip info p.131
- Procedure p.131
- Suppress dtmf display p.133
- Procedure p.133
- Transfer via dtmf p.134
- Procedure p.134
- Procedure p.135
- Outgoing intercom calls p.135
- Intercom p.135
- Incoming intercom calls p.136
- Procedure p.137
- Network address translation p.139
- Music on hold p.139
- Multicast paging p.139
- Message waiting indicator p.139
- Ipv6 support p.139
- Hot desking p.139
- Distinctive ring tones the ip phone inspects the invite request for an alert info header p.139
- Distinctive ring tones allows particular incoming calls to trigger ip phones to play p.139
- X authentication p.139
- Distinctive ring tones p.139
- When receiving an incoming call if the invite request contains an alert info header p.139
- Configuring advanced features p.139
- Tr 069 device management p.139
- This chapter provides information for making configuration changes for the following p.139
- Call recording p.139
- The ip phone strips out the url and keyword parameter and maps it to the appropriate p.139
- Busy lamp field p.139
- Server redundancy p.139
- Automatic call distribution p.139
- Ring tone p.139
- Advanced features p.139
- Remote phone book p.139
- Action url p.139
- Quality of service p.139
- Action uri p.139
- Procedure p.141
- Russia p.143
- Czech etsi p.143
- Sweden p.143
- Configuring advanced features p.143
- Brazil p.143
- Portugal p.143
- Belgium p.143
- Norway p.143
- New zealand p.143
- Austria p.143
- Netherlands p.143
- Australia p.143
- Mexico p.143
- Lithuania p.143
- Indicate different conditions of the ip phone the default tones used on ip phones are p.143
- Hungary p.143
- Greece p.143
- You can customize tones or select specialized tone sets vary from country to country to p.143
- Great britain p.143
- When receiving a message or recording a call the ip phone will play a warning tone p.143
- Germany p.143
- United states p.143
- France p.143
- The us tone sets available tone sets for ip phones p.143
- Finland p.143
- Denmark p.143
- Switzerland p.143
- Procedure p.144
- Remote phone book p.145
- Procedure p.145
- Procedure p.148
- Ldap attributes p.148
- Visual alert and audio alert for blf pickup p.150
- Busy lamp field p.150
- Blf led mode p.151
- Procedure p.152
- Procedure p.155
- Music on hold p.155
- Automatic call distribution p.156
- Procedure p.157
- Message waiting indicator p.158
- Procedure p.159
- Sending rtp stream p.161
- Procedure p.161
- Multicast paging p.161
- Receiving rtp stream p.163
- Procedure p.163
- Record p.165
- Call recording p.165
- Example of an http get message p.166
- Example of a sip info message p.166
- Example of a 200 ok message p.166
- An http get message to the server p.166
- Administrator s guide for sip t4x ip phones p.166
- When the user presses the record key for the second time the ip phone sends a sip p.166
- When a user presses a url record key for the first time during a call the ip phone sends p.166
- Url record p.166
- Recording stops p.166
- Info message to the server with the specific header record off and then the p.166
- If the recording is successfully started the server will respond with a 200 ok message p.166
- Procedure p.167
- Procedure p.169
- Hot desking p.169
- Action url p.170
- Procedure p.173
- Action uri p.173
- Procedure p.175
- Server redundancy p.176
- Phone configuration for redundancy implementation p.176
- Phone registration p.177
- Sip server domain name resolution p.178
- Procedure p.178
- Procedure p.180
- Outgoing call when the working server connection fails p.180
- Lldp med media endpoint discovery p.182
- Procedure p.183
- Vlan discovery via dhcp p.185
- Procedure p.185
- Procedure p.188
- Quality of service p.190
- Voice qos p.191
- Sip qos p.191
- Procedure p.191
- Stun simple traversal of udp over nats p.192
- Network address translation p.192
- Nat traversal p.192
- Procedure p.193
- X authentication p.194
- Procedure p.194
- Tr 069 device management p.199
- Procedure p.200
- Ipv6 support p.201
- Ipv6 address assignment method p.201
- Procedure p.202
- Procedure p.205
- Headset prior p.205
- Configuring audio features p.205
- Procedure p.206
- Dual headset p.206
- Audio codecs p.207
- Packetization time p.208
- Procedure p.209
- Procedure p.211
- Acoustic echo cancellation p.211
- Acoustic clarity technology p.211
- Voice activity detection p.212
- Procedure p.212
- Procedure p.213
- Comfort noise generation p.213
- Procedure p.214
- Jitter buffer p.214
- Transport layer security p.217
- Configuring security features p.217
- Exp1024 rc4 sha p.218
- Des cbc3 sha p.218
- Exp1024 rc4 md5 p.218
- Des cbc sha p.218
- Exp1024 dhe dss rc4 sha p.218
- Aes256 sha p.218
- Exp1024 dhe dss des cbc sha p.218
- Aes128 sha p.218
- Administrator s guide for sip t4x ip phones p.218
- Exp1024 des cbc sha p.218
- Exp rc4 md5 p.218
- Exp edh rsa des cbc sha p.218
- Exp edh dss des cbc sha p.218
- Tls server to establish an encrypted communication channel p.218
- Exp des cbc sha p.218
- The following figure illustrates the tls messages exchanged between the ip phone and p.218
- Edh rsa des cbc3 sha p.218
- Step2 server responds with server hello message selecting the ssl options sends its p.218
- Edh rsa des cbc sha p.218
- Step1 ip phone sends client hello message proposing ssl options p.218
- Edh dss des cbc3 sha p.218
- Rc4 sha p.218
- Edh dss des cbc sha p.218
- Rc4 md5 p.218
- Dhe rsa aes128 sha p.218
- Public key information in server key exchange message and concludes its part of the p.218
- Dhe dss rc4 sha p.218
- Idea cbc sha p.218
- Dhe dss aes128 sha p.218
- Certificates p.219
- Procedure p.220
- Secure real time transport protocol p.223
- Procedure p.224
- Encrypting configuration files p.225
- Procedure to encrypt configuration files p.226
- Procedure p.227
- Upgrade via web user interface p.229
- Upgrading firmware p.229
- Upgrade firmware from the provisioning server p.230
- Procedure p.230
- Resource files p.233
- Replace rule template p.233
- Procedure p.234
- Dial now template p.234
- Softkey layout template p.235
- Procedure p.235
- Procedure p.236
- Procedure p.237
- Local contact file p.237
- Remote xml phone book p.238
- Procedure p.239
- Procedure p.240
- Directory template p.240
- Super search template p.241
- Procedure p.241
- Specifying the access url of resource files p.242
- Viewing log files p.245
- Troubleshooting methods p.245
- Troubleshooting p.245
- Enabling the watch dog feature p.248
- Capturing packets p.248
- Getting information from status indicators p.249
- Analyzing configuration files p.249
- Why is the lcd screen blank p.250
- Why doesn t the ip phone get an ip address p.250
- Troubleshooting solutions p.250
- Why doesn t the ip phone display time and date correctly p.251
- Why do i get poor sound quality during a call p.251
- How do i find the basic information of the ip phone p.251
- Why doesn t the ip phone upgrade firmware successfully p.251
- What is the difference between user name register name and display name p.252
- What is the difference between a remote phone book and a local phonebook p.252
- How to reboot ip phone remotely p.252
- How to increase or decrease the volume p.252
- Why doesn t the ip phone update the configuration p.253
- What will happen if i connect both poe cable and power adapter which has the higher priority p.253
- What is pnp p.253
- What is auto provisioning p.253
- What do on code and off code mean p.253
- How to solve the ip conflict problem p.254
- How to reset your phone to factory configurations p.254
- How to restore the administrator password p.255
- Appendix a glossary p.257
- Appendix p.257
- Appendix b time zones p.259
- Setting parameters in configuration files p.262
- Basic and advanced feature parameters p.262
- Appendix c configuration parameters p.262
- Static network settings p.263
- Pc port transmission method p.267
- Internet por p.267
- Internet and pc ports transmission methods p.267
- Replace rule p.268
- Dial plan p.268
- Dial now p.269
- Area code p.270
- Block out p.271
- Power indicator led p.272
- Contrast p.275
- Backlight p.276
- Administrator password p.277
- User password p.277
- Phone lock p.278
- Time and date p.280
- Language p.286
- Logo customization p.287
- Key as send p.288
- Hotline p.290
- Missed call log p.291
- Call log p.291
- Live dialpad p.292
- Call waiting p.292
- Auto redial p.294
- Auto answer p.295
- Call completion p.296
- Anonymous call p.296
- Anonymous call rejection p.298
- Return message when dnd p.300
- Do not disturb p.300
- Dnd mode p.300
- Dnd in phone mode p.301
- Dnd in custom mode p.302
- Return code when refuse p.303
- Busy tone delay p.303
- Use outbound proxy in dialog p.304
- Ring workaround p.304
- Sip session timer p.305
- Session timer p.306
- Call hold p.307
- Call forward mode p.308
- Call forward p.308
- Call forward in phone mode p.309
- Always forward p.309
- Busy forward p.310
- No answer forward p.311
- Call forward in custom mode p.313
- Always forward p.313
- Busy forward p.315
- No answer forward p.316
- Fwd international p.318
- Call transfer p.318
- Network conference p.319
- Transfer on conference hang up p.320
- Phone basis p.321
- Directed call pickup p.321
- Phone basis p.322
- Per line basis p.322
- Group call pickup p.322
- Dialog info call pickup p.323
- Per line basis p.323
- Web server type p.324
- Calling line identification presentation p.325
- Connected line identification presentation p.326
- Transfer via dtmf p.329
- Suppress dtmf display p.329
- Incoming intercom calls p.330
- Distinctive ring tones p.332
- Switzerland p.335
- Example voice tone country austria p.335
- Sweden p.335
- Description p.335
- Russia p.335
- Denmark p.335
- Voice tone stutter p.335
- Portugal p.335
- Czech etsi p.335
- Voice tone ring p.335
- Parameter p.335
- Customizes the tone for each condition p.335
- Voice tone record p.335
- Norway p.335
- Configuration file p.335
- Voice tone message p.335
- New zealand p.335
- Belgium p.335
- Voice tone info p.335
- Netherlands p.335
- Appendix p.335
- Mexico p.335
- Voice tone dialrecall p.335
- Lithuania p.335
- Voice tone dial p.335
- Hungary p.335
- Voice tone congestion p.335
- Greece p.335
- Voice tone callwaiting p.335
- Great britain p.335
- Voice tone busy p.335
- Voice tone autoanswer p.335
- Germany p.335
- United states p.335
- France p.335
- The parameter voice tone message is not p.335
- Finland p.335
- Remote phone book p.336
- Visual and audio alert for blf pickup p.343
- Blf led mode p.344
- Music on hold p.344
- Message waiting indicator p.346
- Sending rtp stream p.348
- Receiving rtp stream p.348
- Action url p.350
- Action uri p.351
- Server redundancy p.352
- Fallback mode p.354
- Failover mode p.355
- Sip server domain name resolution p.357
- Internet port p.358
- Pc port p.360
- Dhcp vlan discovery p.361
- Network address translation p.363
- Tr 069 p.366
- Audio feature parameters p.373
- Headset prior p.373
- Dual headset p.374
- Audio codecs p.374
- Acoustic echo cancellation p.377
- Voice activity detection p.378
- Jitter buffer p.378
- Comfort noise generation p.378
- Security feature parameters p.380
- Uploading certificates p.382
- Configuring decryption method p.383
- Upgrading firmware p.385
- Resource files p.388
- Access url of replace rule template p.388
- Access url of dial now template p.388
- Access url of softkey layout template p.389
- Access url of local contact file p.391
- Access url of super search template p.392
- Access url of remote xml phone book p.392
- Access url of directory template p.392
- Troubleshooting p.393
- Log settings p.393
- Access url of wallpaper image p.393
- Watch dog p.394
- Parameter p.395
- Detailed in the following p.395
- Or line key 1 3 of sip t42g t41p p.395
- Description p.395
- N a default to line key 7 27 for p.395
- Configuring dss key p.395
- Multicast paging p.395
- Configures the key feature for the line key p.395
- Linekey x type p.395
- Configuration file p.395
- Line default for line key 1 6 of sip t46g p.395
- Conference p.395
- Xml group p.395
- Intercom p.395
- Call return p.395
- X ranges from 1 to 27 for sip t46g p.395
- Hot desking p.395
- Call park p.395
- X ranges from 1 to 15 for sip t42g t41p p.395
- Appendix p.395
- Voice mail p.395
- Group pickup p.395
- Valid types are p.395
- Group listening p.395
- Transfer p.395
- Forward p.395
- This section provides dss key parameters you can configure on the ip phone p.395
- Example watch_dog enable 1 p.395
- Speed dial p.395
- Enabled p.395
- Sms not applicable to sip t42g t41p p.395
- Dss key can be assigned with various key features the parameters of the dss key are p.395
- Sip t46g or line key 4 15 for p.395
- Disabled p.395
- Sip t42g t41p p.395
- Directed pickup p.395
- Keypad lock key p.400
- Dnd key p.400
- Directed call pickup key p.401
- Group call pickup key p.402
- Call park key p.403
- Call return key p.403
- Intercom key p.404
- Ldap key p.405
- Blf key p.406
- Acd key p.407
- Record key p.408
- Multicast paging key p.408
- Url record key p.409
- Hot desking key p.409
- Rfc and internet draft support p.410
- Appendix d sip session initiation protocol p.410
- Rfc 3428 session initiation protocol sip extension for instant messaging p.411
- Rfc 3262 reliability of provisional responses in the session initiation protocol sip p.411
- Rfc 3420 internet media type message sipfrag p.411
- Registration p.411
- Rfc 3725 best current practices for third party call control 3pcc in the session p.411
- Rfc 3372 sip for telephones sip t context and architectures p.411
- Network address translators nats p.411
- Rfc 3711 the secure real time transport protocol srtp p.411
- Rfc 3361 dhcp for ipv4 option for sip servers p.411
- Bandwidth p.411
- Rfc 3702 authentication authorization and accounting requirements for the sip p.411
- Rfc 3326 the reason header field for the session initiation protocol sip p.411
- Appendix p.411
- Rfc 3680 sip event package for registrations p.411
- Rfc 3325 sip asserted identity p.411
- Rfc 3666 sip public switched telephone network pstn call flows p.411
- Rfc 3324 requirements for network asserted identity p.411
- Rfc 3323 a privacy mechanism for the session initiation protocol sip p.411
- Rfc 3665 session initiation protocol sip basic call flow examples p.411
- Rfc 3313 private sip extensions for media authorization p.411
- Rfc 3608 sip extension header field for service route discovery during p.411
- Rfc 3312 integration of resource management and sip p.411
- Rfc 3581 an extension to the sip for symmetric response routing p.411
- Rfc 3311 the session initiation protocol sip update method p.411
- Rfc 3556 session description protocol sdp bandwidth modifiers for rtcp p.411
- Rfc 3310 http digest authentication using authentication and key agreement p.411
- Rfc 3550 rtp rtcp ietf rfc 3550 p.411
- Rfc 3266 support for ipv6 in session description protocol sdp p.411
- Rfc 3515 the session initiation protocol sip refer method p.411
- Rfc 3489 stun simple traversal of user datagram protocol udp through p.411
- Rfc 3265 session initiation protocol sip specific event notification p.411
- Rfc 3486 compressing the session initiation protocol sip p.411
- Rfc 3264 an offer answer model with the session description protocol sdp p.411
- Rfc 3455 private header p header extensions to the sip for the 3gpp p.411
- Rfc 3263 session initiation protocol sip locating sip servers p.411
- Rfc 4028 session timers in the session initiation protocol sip p.412
- Draft ietf sip cc transfer 05 txt sip call control transfer p.412
- Rfc 5589 session initiation protocol sip call control transfer p.412
- Rfc 3969 the internet assigned number authority iana uniform resource p.412
- Administrator s guide for sip t4x ip phones p.412
- Rfc 5359 session initiation protocol service examples p.412
- Rfc 5079 rejecting anonymous requests in sip p.412
- Rfc 3968 the internet assigned number authority iana header field p.412
- Rfc 5009 p early media header p.412
- Rfc 3960 early media and ringing tone generation in sip p.412
- Rfc 4662 a sip event notification extension for resource lists p.412
- Rfc 3959 the early session disposition type for sip p.412
- Rfc 4579 sip call control conferencing for user agents p.412
- Rfc 3892 the session initiation protocol sip referred by mechanism p.412
- Rfc 4575 a sip event package for conference state p.412
- Rfc 3891 the session initiation protocol sip replaces header p.412
- Rfc 4568 session description protocol sdp security descriptions for media p.412
- Rfc 3890 a transport independent bandwidth modifier for the sdp p.412
- Rfc 4566 sdp session description protocol p.412
- Rfc 3856 a presence event package for session initiation protocol sip p.412
- Rfc 4504 sip telephony device requirements and configuration p.412
- Rfc 3842 a message summary and message waiting indication event package p.412
- Rfc 4485 guidelines for authors of extensions to the sip p.412
- Protocol sip p.412
- Rfc 4475 session initiation protocol sip torture p.412
- Parameter registry for the session initiation protocol sip p.412
- Rfc 4353 a framework for conferencing with the sip p.412
- Initiation protocol sip p.412
- Rfc 4317 session description protocol sdp offer answer examples p.412
- Identifier uri parameter registry for the session initiation protocol sip p.412
- Rfc 4244 an extension to the sip for request history information p.412
- For the session initiation protocol sip p.412
- Streams p.412
- Rfc 4235 an invite initiated dialog event package for the session initiation p.412
- Draft levy sip diversion 04 txt diversion indication in sip p.412
- Rfc3966 the tel uri for telephone number p.412
- Sip request p.413
- Sip header p.414
- Xx response successful responses p.415
- Xx response redirection responses p.415
- Xx response information responses p.415
- Sip responses p.415
- Xx response request failure responses p.416
- Xx response server failure responses p.417
- Xx response global responses p.417
- Sip session description protocol sdp usage p.417
- Appendix e sip call flows p.418
- Successful call setup and disconnect p.419
- Unsuccessful call setup called user is busy p.421
- F1 invite b p.422
- User a proxy server user b p.422
- F8 ack p.422
- F7 ack p.422
- F6 486 busy here p.422
- F5 486 busy here p.422
- F4 100 trying p.422
- F3 100 trying p.422
- F2 invite b p.422
- F8 200 ok p.424
- F7 200 ok p.424
- F6 cancel p.424
- F5 cancel p.424
- F4 180 ringing p.424
- F3 180 ringing p.424
- F2 invite b p.424
- F1 invite b p.424
- User a proxy server user b p.424
- Unsuccessful call setup called user does not answer p.424
- Successful call setup and call hold p.426
- Successful call setup and call waiting p.428
- Call transfer without consultation p.433
- User c answers the call p.434
- Call is established between user a and user c p.434
- Administrator s guide for sip t4x ip phones p.434
- Call transfer with consultation p.437
- User a transfers the call to user c p.438
- Call is established between user b and user c p.438
- Administrator s guide for sip t4x ip phones p.438
- Always call forward p.442
- User c answers the call p.443
- Call is established between user a and user c p.443
- Appendix p.443
- Busy call forward p.446
- No answer call forward p.449
- Call conference p.452
- T4x sample configuration file p.457
- Network settings p.457
- Dial plan settings p.457
- Appendix f sample configuration file p.457
- Time settings p.458
- Phone lock p.458
- Language p.458
- Call waiting p.458
- Call hold p.458
- Auto redial p.458
- Auto dst settings p.458
- Hotline p.459
- Dtmf suppression p.459
- Call forward p.459
- Web server type p.459
- In phone mode p.459
- In custom mode p.459
- Remote phone book p.460
- Distinctive ring tones p.460
- Call transfer p.460
- Call conference p.460
- Action url p.461
- Access url of resource files p.462
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Узнайте, как настроить функции интеркома на IP-телефонах, включая отключение микрофона, предупреждающий тон и автоматический ответ на вызовы.