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Yealink SIP-T42G — настройка SIP-телефонов и NAT для VoIP-систем [192/465]
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Administrator’s Guide for SIP-T4X IP Phones
178
3. Enter the desired value in the SIP Qos (0~63) field.
4. Click Confirm to accept the change.
A dialog box pops up to prompt that the settings will take effect after reboot.
5. Click OK to reboot the IP phone.
Network Address Translation (NAT) is essentially a translation table that maps public IP
address and port combinations to private ones. This reduces the need for a large
number of public IP addresses. The NAT feature ensures security since each outgoing or
incoming request must first go through a translation process. But in the VoIP environment,
NAT breaks end-to-end connectivity.
NAT Traversal
NAT traversal is a general term for techniques that establish and maintain IP
connections traversing NAT gateways, typically required for client-to-client networking
applications, especially for VoIP deployments. STUN is one of the NAT traversal
techniques supported by IP phones.
STUN (Simple Traversal of UDP over NATs)
STUN is a network protocol, used in NAT traversal for applications of real-time voice,
video, messaging, and other interactive IP communications. The STUN protocol allows
applications to operate behind a NAT to discover the presence of the network address
translator, and to obtain the mapped (public) IP address and port number that the NAT
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 SIP-телефоны и использовать NAT для обеспечения безопасности и стабильности VoIP-соединений. Подробное руководство по NAT Traversal и STUN.