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Yealink CP860 [373/417] Sip responses
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Appendix
361
Method
Supported
Notes
Event
Yes
Expires
Yes
From
Yes
Max-Forwards
Yes
Min-SE
Yes
P-Asserted-Identity
Yes
P-Preferred-Identity
Yes
Proxy-Authenticate
Yes
Proxy-Authorization
Yes
RAck
Yes
Record-Route
Yes
Refer-To
Yes
Referred-By
Yes
Remote-Party-ID
Yes
Replaces
Yes
Require
Yes
Route
Yes
RSeq
Yes
Session-Expires
Yes
Subscription-State
Yes
Supported
Yes
To
Yes
User-Agent
Yes
Via
Yes
The following SIP responses are supported:
1xx Response—Information Responses
1xx Response
Supported
Notes
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
- About this guide v 7
- Getting started 7 7
- Product overview 1 7
- Table of contents 7
- Table of contents vii 7
- Configuring basic features 39 8
- Configuring advanced features 79 9
- Configuring audio features 91 9
- Configuring security features 05 10
- Resource files 23 10
- Troubleshooting 35 10
- Appendix 49 11
- Index 03 11
- Product overview 13
- Voip principle 13
- Sip components 14
- User agent client uac 14
- Introduction of cp860 ip conference phones 15
- User agent server uas 15
- Cp860 ip conference phone 16
- Key features of cp860 ip conference phones 16
- Phone features 16
- Physical features 16
- Physical features of cp860 ip conference phones 16
- Codecs and voice features 17
- Management 17
- Network features 17
- Security 17
- Connecting the ip phones 19
- Connecting the network and power 19
- Getting started 19
- Connecting the optional extension microphones 21
- Connecting the optional pc or mobile device 22
- Connecting the optional usb flash drive 22
- Initialization process overview 23
- Reading icons 25
- Verifying startup 25
- Configuration methods 26
- Central provisioning 27
- Common cfg file 27
- Configuration files 27
- Mac oriented cfg file 27
- Phone user interface 27
- Web user interface 27
- Provisioning server 28
- Supported provisioning protocols 28
- Deploying phones from the provisioning server 29
- Setting up the provisioning server 29
- Configuring basic network parameters 30
- Dhcp option 31
- Procedure 32
- Details of configuration parameters 33
- Configuring network parameters manually 36
- Details of configuration parameters 37
- Procedure 37
- Auto negotiation 42
- Configuring transmission methods of the internet port 42
- Half duplex 42
- Details of configuration parameters 43
- Full duplex 43
- Procedure 43
- Upgrading firmware 44
- Upgrade firmware from the provisioning server 45
- Upgrade via web user interface 45
- Details of configuration parameters 46
- Procedure 46
- Administrator password 51
- Anonymous call 51
- Anonymous call rejection 51
- Auto answer 51
- Auto redial 51
- Backlight 51
- Basic features 51
- Busy tone delay 51
- Call log 51
- Call waiting 51
- Configuring basic features 51
- Contrast 51
- Dial plan 51
- Directory 51
- Do not disturb 51
- Early media 51
- Hotline 51
- Key as send 51
- Language 51
- Live dialpad 51
- Local directory 51
- Logo customization 51
- Missed call log 51
- Phone lock 51
- Return code when refuse 51
- Ring workaround 51
- Search source list in dialing 51
- Softkey layout 51
- This chapter provides information for making configuration changes for the following 51
- Time and date 51
- User password 51
- Web server type 51
- Contrast 52
- Procedure 52
- Details of configuration parameters 53
- Backlight 54
- Details of configuration parameters 54
- Procedure 54
- Procedure 55
- Web server type 55
- Details of configuration parameters 56
- User password 58
- Details of the configuration parameter 59
- Procedure 59
- Administrator password 60
- Procedure 60
- Details of the configuration parameter 61
- Phone lock 62
- Procedure 62
- Details of configuration parameters 63
- Daylight saving time 66
- Time and date 66
- Time zone 66
- Procedure 67
- Details of configuration parameters 68
- Language 78
- Loading language packs 78
- Details of the configuration parameter 79
- Procedure 79
- Specifying the language to use 79
- Details of configuration parameters 80
- Procedure 80
- Details of configuration parameters 82
- Logo customization 82
- Procedure 82
- Softkey layout 84
- Procedure 86
- Details of configuration parameters 87
- Key as send 89
- Details of configuration parameters 90
- Procedure 90
- Dial plan 93
- Details of configuration parameters 94
- Procedure 94
- Replace rule 94
- Delay time for dial now rule 96
- Dial now 96
- Procedure 96
- Details of configuration parameters 97
- Area code 99
- Procedure 99
- Details of configuration parameters 100
- Block out 101
- Details of configuration parameters 101
- Procedure 101
- Hotline 102
- Details of configuration parameters 103
- Procedure 103
- Details of the configuration parameter 105
- Directory 105
- Procedure 105
- Procedure 106
- Search source list in dialing 106
- Details of the configuration parameter 107
- Call log 108
- Details of the configuration parameter 108
- Procedure 108
- Missed call log 109
- Procedure 109
- Details of the configuration parameter 110
- Details of the configuration parameter 111
- Local directory 111
- Procedure 111
- Details of configuration parameters 113
- Live dialpad 113
- Procedure 113
- Call waiting 115
- Procedure 115
- Details of configuration parameters 116
- Auto redial 119
- Details of configuration parameters 119
- Procedure 119
- Auto answer 121
- Procedure 121
- Details of configuration parameters 122
- Anonymous call 124
- Details of configuration parameters 125
- Procedure 125
- Anonymous call rejection 128
- Details of configuration parameters 128
- Procedure 128
- Do not disturb 130
- Procedure 131
- Return message when dnd 131
- Details of configuration parameters 132
- Busy tone delay 135
- Procedure 135
- Details of the configuration parameter 136
- Return code when refuse 136
- Details of the configuration parameter 137
- Procedure 137
- Early media 138
- Procedure 138
- Ring workaround 138
- Details of the configuration parameter 139
- Details of the configuration parameter 140
- Procedure 140
- Use outbound proxy in dialog 140
- Procedure 141
- Sip session timer 141
- Details of configuration parameters 142
- Call hold 143
- Procedure 144
- Details of configuration parameters 145
- Procedure 148
- Session timer 148
- Details of configuration parameters 149
- Call forward 151
- Forward international 151
- Procedure 151
- Details of configuration parameters 152
- Call transfer 159
- Details of configuration parameters 160
- Procedure 160
- Details of configuration parameters 162
- Network conference 162
- Procedure 162
- Procedure 164
- Transfer on conference hang up 164
- Details of the configuration parameter 165
- Directed call pickup 165
- Procedure 166
- Details of configuration parameters 167
- Group call pickup 169
- Procedure 169
- Details of configuration parameters 170
- Call return 172
- Details of configuration parameters 173
- Procedure 173
- Calling line identification presentation 174
- Procedure 174
- Details of the configuration parameter 175
- Connected line identification presentation 176
- Details of the configuration parameter 176
- Procedure 176
- Rfc 2833 177
- Inband 178
- Procedure 178
- Sip info 178
- Details of configuration parameters 179
- Suppress dtmf display 181
- Details of configuration parameters 182
- Procedure 182
- Details of configuration parameters 184
- Procedure 184
- Transfer via dtmf 184
- Details of configuration parameters 186
- Intercom 186
- Outgoing intercom calls 186
- Procedure 186
- Incoming intercom calls 187
- Details of configuration parameters 188
- Procedure 188
- Action uri 191
- Action url 191
- Advanced features 191
- Configuring advanced features 191
- Distinctive ring tones 191
- Ipv6 support 191
- Message waiting indicator 191
- Multicast paging 191
- Network address translation 191
- Quality of service 191
- Remote phone book 191
- Server redundancy 191
- Static dns cache 191
- This chapter provides information for making configuration changes for the following 191
- Tr 069 device management 191
- X authentication 191
- Distinctive ring tones 192
- Auto answer 194
- Procedure 194
- Details of configuration parameters 195
- Details of configuration parameters 200
- Procedure 200
- Remote phone book 204
- Procedure 205
- Details of configuration parameters 206
- Ldap attributes 209
- Procedure 209
- Details of configuration parameters 210
- Message waiting indicator 216
- Details of configuration parameters 217
- Procedure 217
- Multicast paging 220
- Procedure 220
- Sending rtp stream 220
- Details of the configuration parameter 221
- Receiving rtp stream 223
- Details of configuration parameters 224
- Procedure 224
- Action url 227
- Procedure 229
- Details of configuration parameters 230
- Action uri 242
- Details of the configuration parameter 243
- Procedure 243
- Server redundancy 244
- Phone configuration for redundancy implementation 245
- Phone registration 246
- Procedure 246
- Details of configuration parameters 247
- Sip server domain name resolution 251
- Outgoing call when the working server connection fails 253
- Details of configuration parameters 254
- Procedure 254
- Static dns cache 255
- Details of configuration parameters 256
- Procedure 256
- Lldp med media endpoint discovery 263
- Details of configuration parameters 265
- Procedure 265
- Procedure 267
- Vlan discovery via dhcp 267
- Details of configuration parameters 268
- Procedure 272
- Details of configuration parameters 273
- Quality of service 274
- Details of configuration parameters 276
- Procedure 276
- Sip qos 276
- Voice qos 276
- Nat traversal 278
- Network address translation 278
- Procedure 278
- Stun simple traversal of udp over nats 278
- Details of configuration parameters 279
- Procedure 281
- Details of configuration parameters 282
- Procedure 284
- X authentication 284
- Details of configuration parameters 285
- Tr 069 device management 290
- Procedure 291
- Details of configuration parameters 292
- Ipv6 address assignment method 295
- Ipv6 support 295
- Details of configuration parameters 296
- Procedure 296
- Audio codecs 303
- Configuring audio features 303
- Packetization time 304
- Details of the configuration parameter 305
- Procedure 305
- Acoustic clarity technology 310
- Acoustic echo cancellation 310
- Details of the configuration parameter 310
- Procedure 310
- Automatic gain control 311
- Background noise suppression 311
- Procedure 311
- Voice activity detection 311
- Details of the configuration parameter 312
- Comfort noise generation 313
- Details of the configuration parameter 313
- Procedure 313
- Jitter buffer 314
- Procedure 314
- Details of configuration parameters 315
- Configuring security features 317
- Transport layer security 317
- Administrator s guide for cp860 ip conference phones 318
- Aes128 sha 318
- Aes256 sha 318
- Des cbc sha 318
- Des cbc3 sha 318
- Dhe dss aes128 sha 318
- Dhe dss rc4 sha 318
- Dhe rsa aes128 sha 318
- Edh dss des cbc sha 318
- Edh dss des cbc3 sha 318
- Edh rsa des cbc sha 318
- Edh rsa des cbc3 sha 318
- Exp des cbc sha 318
- Exp edh dss des cbc sha 318
- Exp edh rsa des cbc sha 318
- Exp rc4 md5 318
- Exp1024 des cbc sha 318
- Exp1024 dhe dss des cbc sha 318
- Exp1024 dhe dss rc4 sha 318
- Exp1024 rc4 md5 318
- Exp1024 rc4 sha 318
- Idea cbc sha 318
- Public key information in server key exchange message and concludes its part of the 318
- Rc4 md5 318
- Rc4 sha 318
- Step1 ip phone sends client hello message proposing ssl options 318
- Step2 server responds with server hello message selecting the ssl options sends its 318
- The following figure illustrates the tls messages exchanged between the ip phone and 318
- Tls server to establish an encrypted communication channel 318
- Certificates 319
- Procedure 320
- Details of configuration parameters 321
- Secure real time transport protocol 326
- Details of the configuration parameter 327
- Procedure 327
- Encrypting configuration files 328
- Procedure to encrypt configuration files 329
- Details of configuration parameters 331
- Procedure 331
- Resource files 335
- Procedure 336
- Replace rule template 336
- Dial now template 337
- Procedure 337
- Procedure 338
- Softkey layout template 338
- Directory template 339
- Procedure 340
- Super search template 340
- Procedure 341
- Local contact file 342
- Procedure 342
- Remote xml phone book 343
- Procedure 344
- Troubleshooting 347
- Troubleshooting methods 347
- Viewing log files 347
- Details of configuration parameters 348
- Procedure 348
- Capturing packets 352
- Enabling the watch dog feature 353
- Procedure 353
- Details of the configuration parameter 354
- Getting information from status indicators 354
- Analyzing configuration files 355
- Troubleshooting solutions 355
- Why is the lcd screen blank 355
- How do i find the basic information of the ip phone 356
- Why doesn t the ip phone display time and date correctly 356
- Why doesn t the ip phone get an ip address 356
- Why doesn t the ip phone upgrade firmware successfully 356
- How to reboot the ip phone remotely 357
- What is the difference between a remote phone book and a local phone book 357
- What is the difference between user name register name and display name 357
- Why do i get poor sound quality during a call 357
- What is auto provisioning 358
- What is pnp 358
- What will happen if i connect both poe cable and power adapter which has the higher priority 358
- Why does the ip phone use dob format logo file instead of popular bmp jpg and so on 358
- How to reset your phone to factory configurations 359
- How to solve the ip conflict problem 359
- What do on code and off code mean 359
- Why doesn t the ip phone update the configuration 359
- How to restore the administrator password 360
- Appendix 361
- Appendix a glossary 361
- Appendix b time zones 363
- Appendix c configuring programable key 365
- Appendix d sip session initiation protocol 368
- Appendix 369
- Rfc 1321 the md5 message digest algorithm 369
- Rfc 1889 rtp media control 369
- Rfc 2112 multipart mime 369
- Rfc 2246 the tls protocol version 1 369
- Rfc 2327 sdp session description protocol 369
- Rfc 2543 sip session initiation protocol 369
- Rfc 2616 hypertext transfer protocol http 1 369
- Rfc 2617 http authentication basic and digest access authentication 369
- Rfc 2782 a dns rr for specifying the location of services dns srv 369
- Rfc 2806 urls for telephone calls 369
- Rfc 2833 rtp payload for dtmf digits telephony tones and telephony signals 369
- Rfc 3087 control of service context using sip request uri 369
- Rfc 3261 sip session initiation protocol replacement for rfc 2543 369
- Rfc 3262 reliability of provisional responses in the session initiation protocol 369
- Rfc 3263 session initiation protocol sip locating sip servers 369
- Rfc 3264 an offer answer model with the session description protocol sdp 369
- Rfc 3265 session initiation protocol sip specific event notification 369
- Rfc 3266 support for ipv6 in session description protocol sdp 369
- Rfc 3310 http digest authentication using authentication and key agreement 369
- Rfc 3311 the session initiation protocol sip update method 369
- Rfc 3312 integration of resource management and sip 369
- Rfc 3313 private sip extensions for media authorization 369
- Rfc 3323 a privacy mechanism for the session initiation protocol sip 369
- Rfc 3324 requirements for network asserted identity 369
- Rfc 3325 sip asserted identity 369
- Rfc 3326 the reason header field for the session initiation protocol sip 369
- Rfc 3361 dhcp for ipv4 option for sip servers 369
- Rfc 3372 sip for telephones sip t context and architectures 369
- Rfc 3420 internet media type message sipfrag 369
- Rfc and internet draft support 369
- Rfc2915 the naming authority pointer naptr dns resource record 369
- The following rfc s and internet drafts are supported 369
- Administrator s guide for cp860 ip conference phones 370
- Bandwidth 370
- For the session initiation protocol sip 370
- Identifier uri parameter registry for the session initiation protocol sip 370
- Initiation protocol sip 370
- Network address translators nats 370
- Parameter registry for the session initiation protocol sip 370
- Protocol sip 370
- Registration 370
- Rfc 3428 session initiation protocol sip extension for instant messaging 370
- Rfc 3455 private header p header extensions to the sip for the 3gpp 370
- Rfc 3486 compressing the session initiation protocol sip 370
- Rfc 3489 stun simple traversal of user datagram protocol udp through 370
- Rfc 3515 the session initiation protocol sip refer method 370
- Rfc 3550 rtp rtcp ietf rfc 3550 370
- Rfc 3556 session description protocol sdp bandwidth modifiers for rtcp 370
- Rfc 3581 an extension to the sip for symmetric response routing 370
- Rfc 3608 sip extension header field for service route discovery during 370
- Rfc 3665 session initiation protocol sip basic call flow examples 370
- Rfc 3666 sip public switched telephone network pstn call flows 370
- Rfc 3680 sip event package for registrations 370
- Rfc 3702 authentication authorization and accounting requirements for the sip 370
- Rfc 3711 the secure real time transport protocol srtp 370
- Rfc 3725 best current practices for third party call control 3pcc in the session 370
- Rfc 3842 a message summary and message waiting indication event package 370
- Rfc 3856 a presence event package for session initiation protocol sip 370
- Rfc 3890 a transport independent bandwidth modifier for the sdp 370
- Rfc 3891 the session initiation protocol sip replaces header 370
- Rfc 3892 the session initiation protocol sip referred by mechanism 370
- Rfc 3959 the early session disposition type for sip 370
- Rfc 3960 early media and ringing tone generation in sip 370
- Rfc 3968 the internet assigned number authority iana header field 370
- Rfc 3969 the internet assigned number authority iana uniform resource 370
- Rfc 4028 session timers in the session initiation protocol sip 370
- Rfc 4235 an invite initiated dialog event package for the session initiation 370
- Rfc 4244 an extension to the sip for request history information 370
- Rfc3966 the tel uri for telephone number 370
- Sip request 371
- Sip header 372
- Sip responses 373
- Xx response information responses 373
- Xx response redirection responses 374
- Xx response request failure responses 374
- Xx response successful responses 374
- Xx response server failure responses 375
- Appendix e sip call flows 376
- Sip session description protocol sdp usage 376
- Xx response global responses 376
- F1 invite b 377
- F10 ack 377
- F11 bye 377
- F12 bye 377
- F13 200 ok 377
- F14 200 ok 377
- F2 invite b 377
- F3 100 trying 377
- F4 100 trying 377
- F5 180 ringing 377
- F6 180 ringing 377
- F7 200 ok 377
- F8 200 ok 377
- F9 ack 377
- Successful call setup and disconnect 377
- User a proxy server user b 377
- Way rtp channel established 377
- Unsuccessful call setup called user is busy 379
- F1 invite b 380
- F2 invite b 380
- F3 100 trying 380
- F4 100 trying 380
- F5 486 busy here 380
- F6 486 busy here 380
- F7 ack 380
- F8 ack 380
- User a proxy server user b 380
- F1 invite b 382
- F2 invite b 382
- F3 180 ringing 382
- F4 180 ringing 382
- F5 cancel 382
- F6 cancel 382
- F7 200 ok 382
- F8 200 ok 382
- Unsuccessful call setup called user does not answer 382
- User a proxy server user b 382
- Successful call setup and call hold 384
- Successful call setup and call waiting 386
- Call transfer without consultation 391
- Administrator s guide for cp860 ip conference phones 392
- Call is established between user a and user c 392
- User c answers the call 392
- Call transfer with consultation 395
- Administrator s guide for cp860 ip conference phones 396
- Call is established between user b and user c 396
- User a transfers the call to user c 396
- Always call forward 400
- Appendix 401
- Call is established between user a and user c 401
- User c answers the call 401
- Busy call forward 404
- No answer call forward 407
- Call conference 410
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