1 | изначальная версия редактировать | |
Здравствуйте, картина такая Hyper-V 2012 R2> Ubuntu Server 14.02 > Asterisk 13.13 Cisco 7941G прошивка SIP41.8-5-4S. Тел пишет регистрация, астериск говорит мол сип 222 зарегистрирован, на телефон можно звонить, но в трубке тишина и с телефона звонки не идут. Полагаю что это кодеки, но почему тел пишет Status-Line: SIP/2.0 401 Unauthorized
Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits) Ethernet II, Src: Microsof01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc53:10:9e (00:21:60:53:10:9e) Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222 User Datagram Protocol, Src Port: 35560, Dst Port: 35560 Session Initiation Protocol (401) Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 [Resent Packet: False] Message Header
Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits) Ethernet II, Src: Microsof01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc53:10:9e (00:21:60:53:10:9e) Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222 User Datagram Protocol, Src Port: 35560, Dst Port: 35560 Source Port: 35560 Destination Port: 35560 Length: 560 Checksum: 0x867b [unverified] [Checksum Status: Unverified] [Stream index: 2] Session Initiation Protocol (401) Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 192.168.1.222:35560;branch=z9hG4bK326b363a;received=192.168.1.222 From: <sip:222@192.168.1.11>;tag=00215553089c0002d90d93b8-afe29f02 To: <sip:222@192.168.1.11>;tag=as11c1032b Call-ID: 00215553-089c0002-13ee4830-b47113aa@192.168.1.222 CSeq: 101 REGISTER Server: FPBX-13.0.190.9(13.13.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79801381" Content-Length: 0
2 | No.2 Revision редактировать |
Здравствуйте, картина такая Hyper-V 2012 R2> Ubuntu Server 14.02 > Asterisk 13.13 Cisco 7941G прошивка SIP41.8-5-4S. Тел пишет регистрация, астериск говорит мол сип 222 зарегистрирован, на телефон можно звонить, но в трубке тишина и с телефона звонки не идут. Полагаю что это кодеки, но почему тел пишет Status-Line: SIP/2.0 401 Unauthorized
Frame `Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits)
Ethernet II, Src: Microsof01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc53:10:9e (00:21:60:53:10:9e)
Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222
User Datagram Protocol, Src Port: 35560, Dst Port: 35560
Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
Message Header
Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits)
Ethernet II, Src: Microsof01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc53:10:9e (00:21:60:53:10:9e)
Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222
User Datagram Protocol, Src Port: 35560, Dst Port: 35560
Source Port: 35560
Destination Port: 35560
Length: 560
Checksum: 0x867b [unverified]
[Checksum Status: Unverified]
[Stream index: 2]
Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.1.222:35560;branch=z9hG4bK326b363a;received=192.168.1.222
From: <sip:222@192.168.1.11>;tag=00215553089c0002d90d93b8-afe29f02
To: <sip:222@192.168.1.11>;tag=as11c1032b
Call-ID: 00215553-089c0002-13ee4830-b47113aa@192.168.1.222
CSeq: 101 REGISTER
Server: FPBX-13.0.190.9(13.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79801381"
Content-Length: 00
`
3 | No.3 Revision редактировать |
Здравствуйте, картина такая Hyper-V 2012 R2> Ubuntu Server 14.02 > Asterisk 13.13 Cisco 7941G прошивка SIP41.8-5-4S. Тел пишет регистрация, астериск говорит мол сип 222 зарегистрирован, на телефон можно звонить, но в трубке тишина и с телефона звонки не идут. Полагаю что это кодеки, но почему тел пишет Status-Line: SIP/2.0 401 Unauthorized
`Frame Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits)
Ethernet II, Src: Microsof01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc53:10:9e (00:21:60:53:10:9e)
Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222
User Datagram Protocol, Src Port: 35560, Dst Port: 35560
Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
Message Header
Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits)
Ethernet II, Src: Microsof01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc53:10:9e (00:21:60:53:10:9e)
Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222
User Datagram Protocol, Src Port: 35560, Dst Port: 35560
Source Port: 35560
Destination Port: 35560
Length: 560
Checksum: 0x867b [unverified]
[Checksum Status: Unverified]
[Stream index: 2]
Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.1.222:35560;branch=z9hG4bK326b363a;received=192.168.1.222
From: <sip:222@192.168.1.11>;tag=00215553089c0002d90d93b8-afe29f02
To: <sip:222@192.168.1.11>;tag=as11c1032b
Call-ID: 00215553-089c0002-13ee4830-b47113aa@192.168.1.222
CSeq: 101 REGISTER
Server: FPBX-13.0.190.9(13.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79801381"
Content-Length: 0
`0
4 | No.4 Revision редактировать |
Здравствуйте, картина такая Hyper-V 2012 R2> Ubuntu Server 14.02 > Asterisk 13.13 Cisco 7941G прошивка
SIP41.8-5-4S. SIP41.8-5-4S.
Тел пишет регистрация, астериск говорит мол сип 222 зарегистрирован, на телефон можно звонить, но в трубке тишина и с телефона звонки не идут. Полагаю что это кодеки, но почему тел пишет Status-Line: SIP/2.0 401
Unauthorized Unauthorized
Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits)
Ethernet II, Src: 5 | No.5 Revision редактировать |
Здравствуйте, картина такая Hyper-V 2012 R2> Ubuntu Server 14.02 > Asterisk 13.13 Cisco 7941G прошивка SIP41.8-5-4S.
Тел пишет регистрация, астериск говорит мол сип 222 зарегистрирован, на телефон можно звонить, но в трубке тишина и с телефона звонки не идут. Полагаю что это кодеки, но почему тел пишет Status-Line: SIP/2.0 401 Unauthorized
Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits)
Ethernet II, Src: Microsof_01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc_53:10:9e (00:21:60:53:10:9e)
Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222
User Datagram Protocol, Src Port: 35560, Dst Port: 35560
Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
Message Header
Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits)
Ethernet II, Src: Microsof_01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc_53:10:9e (00:21:60:53:10:9e)
Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222
User Datagram Protocol, Src Port: 35560, Dst Port: 35560
Source Port: 35560
Destination Port: 35560
Length: 560
Checksum: 0x867b [unverified]
[Checksum Status: Unverified]
[Stream index: 2]
Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.1.222:35560;branch=z9hG4bK326b363a;received=192.168.1.222
From: <sip:222@192.168.1.11>;tag=00215553089c0002d90d93b8-afe29f02
To: <sip:222@192.168.1.11>;tag=as11c1032b
Call-ID: 00215553-089c0002-13ee4830-b47113aa@192.168.1.222
CSeq: 101 REGISTER
Server: FPBX-13.0.190.9(13.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79801381"
Content-Length: 0
конфиг телефона
<device> <fullconfig>true</fullconfig> <deviceprotocol>SIP</deviceprotocol> <devicepool> <datetimesetting> <datetemplate>D.M.Y</datetemplate> <timezone>Russian Standard/Daylight Time</timezone> <ntps> <ntp> <name>192.168.1.11</name> <ntpmode>Unicast</ntpmode> </ntp> </ntps> </datetimesetting> <callmanagergroup> <tftpdefault>true</tftpdefault> <members> <member priority="0"> <callmanager> <name>192.168.1.11</name> <description>CallManager 5.0</description> <ports> <ethernetphoneport>2000</ethernetphoneport> <sipport>35560</sipport> <securedsipport>5061</securedsipport> </ports> <processnodename>192.168.1.11</processnodename> </callmanager> </member> </members> </callmanagergroup> </devicepool> <commonprofile> <phonepassword></phonepassword> <backgroundimageaccess>true</backgroundimageaccess> <calllogblfenabled>0</calllogblfenabled> </commonprofile> <loadinformation>SIP41.8-5-4S</loadinformation> <loadinformation434 model="Cisco 7941">SIP41.8-5-4S</loadinformation434> <vendorconfig> <disablespeaker>false</disablespeaker> <disablespeakerandheadset>false</disablespeakerandheadset> <pcport>0</pcport> <settingsaccess>1</settingsaccess> <garp>0</garp> <voicevlanaccess>0</voicevlanaccess> <videocapability>0</videocapability> <autoselectlineenable>0</autoselectlineenable> <daysdisplaynotactive>1,7</daysdisplaynotactive> <displayontime>10:30</displayontime> <displayonduration>06:05</displayonduration> <displayidletimeout>00:05</displayidletimeout> <webaccess>0</webaccess> <spantopcport>1</spantopcport> <loggingdisplay>1</loggingdisplay> <loadserver></loadserver> </vendorconfig> <userlocale> <name>RussianRussianFederation</name> <uid></uid> <langcode>ruRU</langcode> <version>8.4.3.1000-1</version> <wincharset>utf-8</wincharset> </userlocale> <networklocale>RussianFederation</networklocale> <networklocaleinfo> <name>Russian_Federation</name> <uid></uid> <version>8.4.3.1000-1</version> </networklocaleinfo> <devicesecuritymode>1</devicesecuritymode> <idletimeout>0</idletimeout> <directoryurl></directoryurl> <servicesurl></servicesurl> <idleurl></idleurl> <messagesurl></messagesurl> <proxyserverurl></proxyserverurl> <dscpforsccpphoneconfig>96</dscpforsccpphoneconfig> <dscpforsccpphoneservices>0</dscpforsccpphoneservices> <dscpforcm2dvce>96</dscpforcm2dvce> <transportlayerprotocol>2</transportlayerprotocol> <capfauthmode>0</capfauthmode> <capflist> <capf> <phoneport>3804</phoneport> </capf> </capflist> <certhash></certhash> <encrconfig>false</encrconfig> <sipprofile> <sipproxies> <backupproxy>192.168.1.11</backupproxy> <backupproxyport>35560</backupproxyport> <emergencyproxy>192.168.1.11</emergencyproxy> <emergencyproxyport>35560</emergencyproxyport> <outboundproxy>192.168.1.11</outboundproxy> <outboundproxyport>35560</outboundproxyport> <registerwithproxy>true</registerwithproxy> </sipproxies> <sipcallfeatures> <cnfjoinenabled>true</cnfjoinenabled> <callforwarduri>x--serviceuri-cfwdall</callforwarduri> <callpickupuri>x-cisco-serviceuri-pickup</callpickupuri> <callpickuplisturi>x-cisco-serviceuri-opickup</callpickuplisturi> <callpickupgroupuri>x-cisco-serviceuri-gpickup</callpickupgroupuri> <meetmeserviceuri>x-cisco-serviceuri-meetme</meetmeserviceuri> <abbreviateddialuri>x-cisco-serviceuri-abbrdial</abbreviateddialuri> <rfc2543hold>false</rfc2543hold> <callholdringback>2</callholdringback> <localcfwdenable>true</localcfwdenable> <semiattendedtransfer>true</semiattendedtransfer> <anonymouscallblock>2</anonymouscallblock> <calleridblocking>2</calleridblocking> <dndcontrol>0</dndcontrol> <remoteccenable>true</remoteccenable> </sipcallfeatures> <sipstack> <sipinviteretx>6</sipinviteretx> <sipretx>10</sipretx> <timerinviteexpires>180</timerinviteexpires> <timerregisterexpires>3600</timerregisterexpires> <timerregisterdelta>5</timerregisterdelta> <timerkeepaliveexpires>120</timerkeepaliveexpires> <timersubscribeexpires>120</timersubscribeexpires> <timersubscribedelta>5</timersubscribedelta> <timert1>500</timert1> <timert2>4000</timert2> <maxredirects>70</maxredirects> <remotepartyid>false</remotepartyid> <userinfo>None</userinfo> </sipstack> <autoanswertimer>1</autoanswertimer> <autoansweraltbehavior>false</autoansweraltbehavior> <autoansweroverride>true</autoansweroverride> <transferonhookenabled>false</transferonhookenabled> <enablevad>false</enablevad> <preferredcodec>g711alaw</preferredcodec> <dtmfavtpayload>101</dtmfavtpayload> <dtmfdblevel>3</dtmfdblevel> <dtmfoutofband>avt</dtmfoutofband> <alwaysuseprimeline>false</alwaysuseprimeline> <alwaysuseprimelinevoicemail>false</alwaysuseprimelinevoicemail> <kpml>3</kpml> <stuttermsgwaiting>1</stuttermsgwaiting> <callstats>true</callstats> <silentperiodbetweencallwaitingbursts>10</silentperiodbetweencallwaitingbursts> <disablelocalspeeddialconfig>true</disablelocalspeeddialconfig> <startmediaport>16384</startmediaport> <stopmediaport>32768</stopmediaport> <voipcontrolport>35560</voipcontrolport> <dscpforaudio>184</dscpforaudio> <ringsettingbusystationpolicy>0</ringsettingbusystationpolicy> <dialtemplate> <template match="*" timeout="3"/> </dialtemplate> <phonelabel>Cisco</phonelabel> <natreceivedprocessing>false</natreceivedprocessing> <natenabled>false</natenabled> <nataddress></nataddress> <siplines> <line button="1"> <featureid>9</featureid> <featurelabel>222</featurelabel> <proxy>192.168.1.11</proxy> <port>35560</port> <name>222</name> <displayname>222</displayname> <autoanswer> <autoanswerenabled>2</autoanswerenabled> </autoanswer> <callwaiting>3</callwaiting> <authname>222</authname> <authpassword>123456789</authpassword> <sharedline>false</sharedline> <messagewaitinglamppolicy>3</messagewaitinglamppolicy> <messagesnumber></messagesnumber> <ringsettingidle>4</ringsettingidle> <ringsettingactive>5</ringsettingactive> <contact>222</contact> <forwardcallinfodisplay> <callername>true</callername> <callernumber>false</callernumber> <redirectednumber>false</redirectednumber> <dialednumber>true</dialednumber> </forwardcallinfodisplay> </line> <line button="2"> <featureid></featureid> <featurelabel></featurelabel> <speeddialnumber></speeddialnumber> </line> </siplines> </sipprofile> </device>
Дебаг Asterisk
Registered SIP '222' at 192.168.1.5:35560
Saved useragent "Cisco-CP7941G/8.5.3" for peer 222 -- Registered SIP '222' at 192.168.1.220:35560
6 | No.6 Revision редактировать |
Здравствуйте, картина такая Hyper-V 2012 R2> Ubuntu Server 14.02 > Asterisk 13.13 Cisco 7941G прошивка SIP41.8-5-4S.
Тел пишет регистрация, астериск говорит мол сип 222 зарегистрирован, на телефон можно звонить, но в трубке тишина и с телефона звонки не идут. Полагаю что это кодеки, но почему тел пишет Status-Line: SIP/2.0 401 Unauthorized
Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits)
Ethernet II, Src: Microsof_01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc_53:10:9e (00:21:60:53:10:9e)
Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222
User Datagram Protocol, Src Port: 35560, Dst Port: 35560
Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
Message Header
Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits)
Ethernet II, Src: Microsof_01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc_53:10:9e (00:21:60:53:10:9e)
Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222
User Datagram Protocol, Src Port: 35560, Dst Port: 35560
Source Port: 35560
Destination Port: 35560
Length: 560
Checksum: 0x867b [unverified]
[Checksum Status: Unverified]
[Stream index: 2]
Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.1.222:35560;branch=z9hG4bK326b363a;received=192.168.1.222
From: <sip:222@192.168.1.11>;tag=00215553089c0002d90d93b8-afe29f02
To: <sip:222@192.168.1.11>;tag=as11c1032b
Call-ID: 00215553-089c0002-13ee4830-b47113aa@192.168.1.222
CSeq: 101 REGISTER
Server: FPBX-13.0.190.9(13.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79801381"
Content-Length: 0
конфиг телефона
Дебаг Asterisk
Registered SIP '222' at 192.168.1.5:35560
Saved useragent "Cisco-CP7941G/8.5.3" for peer 222 -- Registered SIP '222' at 192.168.1.220:35560
7 | No.7 Revision редактировать |
Здравствуйте, картина такая Hyper-V 2012 R2> Ubuntu Server 14.02 > Asterisk 13.13 Cisco 7941G прошивка SIP41.8-5-4S.
Тел пишет регистрация, астериск говорит мол сип 222 зарегистрирован, на телефон можно звонить, но в трубке тишина и с телефона звонки не идут. Полагаю что это кодеки, но почему тел пишет Status-Line: SIP/2.0 401 Unauthorized
Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits)
Ethernet II, Src: Microsof_01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc_53:10:9e (00:21:60:53:10:9e)
Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222
User Datagram Protocol, Src Port: 35560, Dst Port: 35560
Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
Message Header
Frame 24: 594 bytes on wire (4752 bits), 594 bytes captured (4752 bits)
Ethernet II, Src: Microsof_01:13:60 (00:15:5d:01:18:70), Dst: CiscoInc_53:10:9e (00:21:60:53:10:9e)
Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.222
User Datagram Protocol, Src Port: 35560, Dst Port: 35560
Source Port: 35560
Destination Port: 35560
Length: 560
Checksum: 0x867b [unverified]
[Checksum Status: Unverified]
[Stream index: 2]
Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.1.222:35560;branch=z9hG4bK326b363a;received=192.168.1.222
From: <sip:222@192.168.1.11>;tag=00215553089c0002d90d93b8-afe29f02
To: <sip:222@192.168.1.11>;tag=as11c1032b
Call-ID: 00215553-089c0002-13ee4830-b47113aa@192.168.1.222
CSeq: 101 REGISTER
Server: FPBX-13.0.190.9(13.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79801381"
Content-Length: 0
конфиг телефона
<deviceprotocol>SIP</deviceprotocol> <devicepool><device>
<fullconfig>true</fullconfig>
Дебаг Asterisk
Registered SIP '222' at 192.168.1.5:35560
Saved useragent "Cisco-CP7941G/8.5.3" for peer 222 -- Registered SIP '222' at 192.168.1.220:35560
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.