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 <br> 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="*" <br=""> 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
Полагаю, что у вас комбинация nat на астериске и телефоне не совпадает.
Это не сложно, всегото 4 комбинации, просто проверить
Задан: 2017-01-23 11:30:40 +0400
Просмотрен: 389 раз
Обновлен: Jan 24 '17
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.
"Дебаг Asterisk" - если это для вас дебаг , то обратитесь к системному админстратору , чтобы он снял этот самый дебаг.
zzuz ( 2017-01-23 17:19:10 +0400 )редактировать