Cisco 7965 sip + asterisk пример xml

у меня работает cisco 7911 (sip8.5.3)

<device> <deviceProtocol>SIP</deviceProtocol> <sshUserId>root</sshUserId> <sshPassword>root</sshPassword> <devicePool> <dateTimeSetting> <dateTemplate>M.D.Ya</dateTemplate> <timeZone>Ekaterinburg Standard Time</timeZone> <ntps> <ntp> <name>192.168.1.70</name> <ntpMode>Unicast</ntpMode> </ntp> </ntps> </dateTimeSetting> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> </ports> <processNodeName>192.168.1.70</processNodeName> </callManager> </member> </members> </callManagerGroup> </devicePool> <commonProfile> <phonePassword></phonePassword> <backgroundImageAccess>true</backgroundImageAccess> <callLogBlfEnabled>2</callLogBlfEnabled> </commonProfile> <loadInformation>SIP11.8-5-3S</loadInformation> <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> <sshAccess>0</sshAccess> <sshPort>22</sshPort> <webAccess>0</webAccess> <spanToPCPort>1</spanToPCPort> <loggingDisplay>1</loggingDisplay> <loadServer></loadServer> </vendorConfig> <networkLocale>Russian_Federation</networkLocale> <networkLocaleInfo> <name>Russian_Federation</name> <uid>64</uid> <version>1.0.0.0-1</version> </networkLocaleInfo> <deviceSecurityMode>1</deviceSecurityMode> <authenticationURL></authenticationURL> <directoryURL></directoryURL> <idleURL></idleURL> <informationURL></informationURL> <messagesURL></messagesURL> <proxyServerURL></proxyServerURL> <servicesURL></servicesURL> <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></backupProxy> <backupProxyPort></backupProxyPort> <emergencyProxy></emergencyProxy> <emergencyProxyPort></emergencyProxyPort> <outboundProxy></outboundProxy> <outboundProxyPort></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>none</preferredCodec> <dtmfAvtPayload>101</dtmfAvtPayload> <dtmfDbLevel>3</dtmfDbLevel> <dtmfOutofBand>avt</dtmfOutofBand> <alwaysUsePrimeLine>false</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <natEnabled>false</natEnabled> <natAddress></natAddress> <stutterMsgWaiting>0</stutterMsgWaiting> <callStats>false</callStats> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> <startMediaPort>16384</startMediaPort> <stopMediaPort>32766</stopMediaPort> <voipControlPort>5060</voipControlPort> <dscpForAudio>184</dscpForAudio> <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> <dialTemplate>dialplan.xml</dialTemplate> <phoneLabel>cis</phoneLabel> <sipLines> <line button="1"> <featureID>9</featureID> <featureLabel>cis</featureLabel> <name>cis</name> <displayName>cis</displayName> <proxy>192.168.1.70</proxy> <port>5060</port> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>3</callWaiting> <authName>cis</authName> <authPassword>1000</authPassword> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>1</messageWaitingLampPolicy> <messagesNumber>*97</messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>false</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> </line> </sipLines> </sipProfile> </device> 

там граблей то много может быть, у меня с этим конфигом на прошивках 9-2-1/3 не регистрируется

на воип инфо есть ссылка на рабочие конфиги. гдето тута http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx