[asterisk-bugs] [JIRA] (ASTERISK-13145) [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML
Andrey Zharkov (JIRA)
noreply at issues.asterisk.org
Fri Feb 1 05:29:55 CST 2019
[ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=246122#comment-246122 ]
Andrey Zharkov commented on ASTERISK-13145:
-------------------------------------------
Hi Gareth.
I cannot make call from the second line from 78XX and 88XX series phones. Calls goes like from first line.
>From second line of 79XX phones calls goes correctly.
This my configuration of test system:
1. CISCO-7965 (fw sip45.9-4-2SR2-2S) ext: 1 line - 100, 2 line - 101
2. CISCO-8845 (fw sip8845_65.12-1-1-12) ext: 1 line - 106
3. CISCO-8865 (fw sip8845_65.12-1-1-12) ext: 1 line - 107, 2 line - 108
sip.conf
{noformat}
[general]
context=public
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=tcp,udp
srvlookup=yes
[extension](!)
type=friend
context=extensions
host=dynamic
nat=no
trustrpid=no
sendrpid=rpid
rpid_update=yes
rpid_immediate=yes
parkinglot=default
allowsubscribe=yes
notifyhold=no
callcounter=yes
videosupport=no
disallow=all
allow=g722,ulaw,alaw,g729
[cisco-usecallmanager](!,extension)
transport=tcp
cisco_usecallmanager=yes
cisco_pickupnotify_alert=from,to
cisco_pickupnotify_timer=5
cisco_keep_conference=no
cisco_multiadmin_conference=yes
dndbusy=yes
huntgroup_default=no
[insecure-mode](!)
transport=tcp
[cisco-88XX](!,cisco-usecallmanager)
busylevel=4
call-limit=5
videosupport=yes
allow=h264
[cisco-79XX](!,cisco-usecallmanager)
busylevel=4
call-limit=5
[100](cisco-79XX,insecure-mode)
secret=100
callerid="NUMBER 100" <100>
description=NUMBER 100
callgroup=1
pickupgroup=1
mailbox=100 at default
register=101
[101](cisco-79XX,insecure-mode)
secret=101
callerid="NUMBER 101" <101>
description=NUMBER 101
[106](cisco-88XX,insecure-mode)
secret=106
callerid="NUMBER 106" <106>
description=NUMBER 106
callgroup=1
pickupgroup=1
mailbox=106 at default
[107](cisco-88XX,insecure-mode)
secret=107
callerid="NUMBER 107" <107>
description=NUMBER 107
callgroup=1
pickupgroup=1
mailbox=107 at default
register=108
[108](cisco-88XX,insecure-mode)
secret=108
callerid="NUMBER 108" <108>
description=NUMBER 108
;[150](extension)
;secret=150
;callerid="NUMBER 150" <150>
;description=NUMBER 150
;callgroup=1
;pickupgroup=1
;mailbox=150 at default
;transport=udp
;videosupport=yes
;; Allow the video codec
;allow=h264
{noformat}
extensions.conf
{noformat}
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp
[call-extension]
exten => _X.,1,Set(PEERNAME=${EXTEN})
; By manually checking for call-forwarding, the call can be forwarded even if the phone is unregistered
same => next,Set(CALLFORWARD=${SIPPEER(${PEERNAME},callforward)})
same => next,GotoIf($[${LEN(${CALLFORWARD})} != 0]?callforward,1)
same => next,Set(MAILBOX=${SIPPEER(${PEERNAME},mailbox)})
; Phones can be made to play a different ring using the Alert-Info header, see Ring Tones for examples
same => next,ExecIf($[${LEN(${ALERT_INFO})} != 0]?SIPAddHeader(Alert-Info: <${ALERT_INFO}>))
same => next,Dial(SIP/${PEERNAME},30)
same => next,Goto(${TOLOWER(${DIALSTATUS})},1)
; Cisco SIP phones support displaying diversion information
exten => callforward,1,SIPAddHeader(Diversion: "${SIPPEER(${PEERNAME},callerid_name)}" <sip:${SIPPEER(${PEERNAME},callerid_num)}@localhost>\;screen=yes\;privacy=off)
same => next,Goto(extensions,${CALLFORWARD},1)
; If ${MAILBOX} is empty send congestion, otherwise go to voicemail extension below
exten => congestion,1,ExecIf($[${LEN(${MAILBOX})} = 0]?Congestion(10))
same => next,Set(GREETING=u)
same => next,Goto(voicemail,1)
; If ${MAILBOX} is empty send busy, otherwise go to voicemail extension below
exten => busy,1,ExecIf($[${LEN(${MAILBOX})} = 0]?Busy(10))
same => next,Set(GREETING=b)
same => next,Goto(voicemail,1)
exten => noanswer,1,Goto(congestion,1)
exten => chanunavail,1,Goto(congestion,1)
exten => voicemail,1,Answer()
same => next,Wait(0.5)
same => next,VoiceMail(${MAILBOX},${GREETING})
same => next,Hangup(normal_clearing)
[extensions]
exten => 100,1,Goto(call-extension,${EXTEN},1)
same => hint,SIP/100,SIP/100
exten => 101,1,Goto(call-extension,${EXTEN},1)
same => hint,SIP/101,SIP/101
exten => 106,1,Goto(call-extension,${EXTEN},1)
same => hint,SIP/106,SIP/106
exten => 107,1,Goto(call-extension,${EXTEN},1)
same => hint,SIP/107,SIP/107
exten => 108,1,Goto(call-extension,${EXTEN},1)
same => hint,SIP/108,SIP/108
exten => 8000,1,Set(CONNECTEDLINE(all)="Voice Mail" <${EXTEN}>)
same => next,Answer()
same => next,Wait(0.5)
same => next,Set(MAILBOX=${SIPPEER(${CHANNEL(peername)},mailbox)})
same => next,VoiceMailMain(${MAILBOX},s)
same => next,Hangup(normal_clearing)
{noformat}
7965.cnf.xml - 100,101
{noformat}
<?xml version="1.0" encoding="UTF-8"?>
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>pbxadmin</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<name>Dallas 5.0 Beta</name>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>Arabian Standard Time</timeZone>
<ntps>
<ntp>
<name>10.10.11.254</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>10.10.10.34</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>3</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP45.9-4-2SR2-2S</loadInformation>
<featurePolicyFile>DefaultFP.xml</featurePolicyFile>
<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>
<webAccess>0</webAccess>
<daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>10:30</displayOnDuration>
<displayIdleTimeout>01:00</displayIdleTimeout>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<spanToPCPort>1</spanToPCPort>
</vendorConfig>
<addOnModules>
<addOnModule idx="1">
<loadInformation>B016-1-0-4-2</loadInformation>
</addOnModule>
<addOnModule idx="2">
<loadInformation>B016-1-0-4-2</loadInformation>
</addOnModule>
</addOnModules>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>8.5.0.0(1)</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL>http://10.10.10.34/auth</authenticationURL>
<servicesURL>http://10.10.10.34/cgi/cisco/services?name=SEPB8BEBF236B20</servicesURL>
<directoryURL>http://10.10.10.34/cgi/cisco/services?name=SEPB8BEBF236B20</directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesNumber>8000</messagesNumber>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>1</transportLayerProtocol>
<dndCallAlert>5</dndCallAlert>
<phonePersonalization>1</phonePersonalization>
<rollover>0</rollover>
<singleButtonBarge>0</singleButtonBarge>
<joinAcrossLines>1</joinAcrossLines>
<autoCallPickupEnable>false</autoCallPickupEnable>
<blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
<blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<sipProfile>
<sipProxies>
<backupProxy>USECALLMANAGER</backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy>USECALLMANAGER</emergencyProxy>
<emergencyProxyPort>5060</emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI>
<callPickupURI>*8</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>
<retainForwardInformation>true</retainForwardInformation>
</sipCallFeatures>
<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>180</timerRegisterExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>true</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>0</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>true</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>10000</startMediaPort>
<stopMediaPort>20000</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<dscpVideo>136</dscpVideo>
<dscpForTelepresence>128</dscpForTelepresence>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<softKeyFile>softkey_7965_tcp.xml</softKeyFile>
<dialTemplate>dialplan_3.xml</dialTemplate>
<phoneLabel>123456789012</phoneLabel>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>100</featureLabel>
<name>100</name>
<displayName>100</displayName>
<contact>100</contact>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>100</authName>
<authPassword>100</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>1</messageWaitingAMWI>
<messagesNumber>8000</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>6</maxNumCalls>
<busyTrigger>4</busyTrigger>
</line>
<line button="2" lineIndex="2">
<featureID>9</featureID>
<featureLabel>qwe101</featureLabel>
<name>101</name>
<displayName>101</displayName>
<contact>101</contact>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>101</authName>
<authPassword>101</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>1</messageWaitingAMWI>
<messagesNumber>8000</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>6</maxNumCalls>
<busyTrigger>4</busyTrigger>
</line>
</sipLines>
</sipProfile>
{noformat}
8845.cnf.xml - 106
{noformat}
<?xml version="1.0" encoding="UTF-8"?>
<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>Arabian Standard Time</timeZone>
<ntps>
<ntp>
<name>10.10.11.254</name>
<ntpMode>unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>10.10.10.34</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<!-- <vpnGroup>
<mtu>1290</mtu>
<failConnectTime>30</failConnectTime>
<authMethod>0</authMethod>
<pswdPersistent>1</pswdPersistent>
<autoNetDetect>1</autoNetDetect>
<enableHostIDCheck>0</enableHostIDCheck>
<addresses>
<url1></url1>
</addresses>
<credentials>
<hashAlg>0</hashAlg>
<certHash1></certHash1>
</credentials>
</vpnGroup> -->
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-cisco-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>1</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>3</anonymousCallBlock>
<callerIdBlocking>3</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
<retainForwardInformation>false</retainForwardInformation>
<uriDialingDisplayPreference>1</uriDialingDisplayPreference>
</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>true</remotePartyID>
<userInfo>Phone</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>true</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>0</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>true</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<natEnabled>false</natEnabled>
<natReceivedProcessing>false</natReceivedProcessing>
<natAddress></natAddress>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>106</featureLabel>
<name>106</name>
<displayName>106</displayName>
<contact>106</contact>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>106</authName>
<authPassword>106</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>1</messageWaitingAMWI>
<messagesNumber>8000</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>6</maxNumCalls>
<busyTrigger>4</busyTrigger>
</line>
</sipLines>
<externalNumberMask></externalNumberMask>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate></dialTemplate>
<softKeyFile></softKeyFile>
</sipProfile>
<MissedCallLoggingOption>1</MissedCallLoggingOption>
<featurePolicyFile></featurePolicyFile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>3</callLogBlfEnabled>
</commonProfile>
<vendorConfig>
<defaultWallpaperFile></defaultWallpaperFile>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<enableMuteFeature>false</enableMuteFeature>
<enableGroupListen>true</enableGroupListen>
<holdResumeKey>1</holdResumeKey>
<recentsSoftKey>1</recentsSoftKey>
<dfBit>1</dfBit>
<pcPort>0</pcPort>
<spanToPCPort>1</spanToPCPort>
<garp>0</garp>
<rtcp>1</rtcp>
<videoRtcp>1</videoRtcp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>1</videoCapability>
<hideVideoByDefault>0</hideVideoByDefault>
<separateMute>0</separateMute>
<ciscoCamera>1</ciscoCamera>
<usb1>1</usb1>
<usb2>1</usb2>
<usbClasses>0,1,2</usbClasses>
<sdio>1</sdio>
<wifi>1</wifi>
<bluetooth>1</bluetooth>
<bluetoothProfile>0,1</bluetoothProfile>
<btpbap>0</btpbap>
<bthfu>0</bthfu>
<ehookEnable>0</ehookEnable>
<autoSelectLineEnable>1</autoSelectLineEnable>
<autoCallSelect>1</autoCallSelect>
<incomingCallToastTimer>10</incomingCallToastTimer>
<dialToneFromReleaseKey>0</dialToneFromReleaseKey>
<joinAndDirectTransferPolicy>0</joinAndDirectTransferPolicy>
<minimumRingVolume></minimumRingVolume>
<simplifiedNewCall>0</simplifiedNewCall>
<actionableAlert>0</actionableAlert>
<showCallHistoryForSelectedLine>0</showCallHistoryForSelectedLine>
<kemOneColumn>0</kemOneColumn>
<lineMode>0</lineMode>
<allCallsOnPrimary>0</allCallsOnPrimary>
<softKeyControl>0</softKeyControl>
<settingsAccess>1</settingsAccess>
<webProtocol>0</webProtocol>
<webAccess>0</webAccess>
<webAdmin>1</webAdmin>
<adminPassword></adminPassword>
<sshAccess>0</sshAccess>
<detectCMConnectionFailure>0</detectCMConnectionFailure>
<g722CodecSupport>1</g722CodecSupport>
<handsetWidebandEnable>2</handsetWidebandEnable>
<headsetWidebandEnable>2</headsetWidebandEnable>
<headsetWidebandUIControl>1</headsetWidebandUIControl>
<handsetWidebandUIControl>1</handsetWidebandUIControl>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>10:00</displayOnDuration>
<displayIdleTimeout>00:10</displayIdleTimeout>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<displayRefreshRate>0</displayRefreshRate>
<daysBacklightNotActive>1,7</daysBacklightNotActive>
<backlightOnTime>08:00</backlightOnTime>
<backlightOnDuration>10:00</backlightOnDuration>
<backlightIdleTimeout>00:10</backlightIdleTimeout>
<backlightOnWhenIncomingCall>1</backlightOnWhenIncomingCall>
<recordingTone>0</recordingTone>
<recordingToneLocalVolume>100</recordingToneLocalVolume>
<recordingToneRemoteVolume>50</recordingToneRemoteVolume>
<recordingToneDuration></recordingToneDuration>
<moreKeyReversionTimer>5</moreKeyReversionTimer>
<peerFirmwareSharing>0</peerFirmwareSharing>
<loadServer></loadServer>
<problemReportUploadURL></problemReportUploadURL>
<enableCdpSwPort>1</enableCdpSwPort>
<enableCdpPcPort>0</enableCdpPcPort>
<enableLldpSwPort>1</enableLldpSwPort>
<enableLldpPcPort>0</enableLldpPcPort>
<cdpEnable>true</cdpEnable>
<outOfRangeAlert>0</outOfRangeAlert>
<scanningMode>2</scanningMode>
<applicationURL></applicationURL>
<appButtonTimer>0</appButtonTimer>
<appButtonPriority>0</appButtonPriority>
<specialNumbers></specialNumbers>
<sendKeyAction>0</sendKeyAction>
<powerOffWhenCharging>0</powerOffWhenCharging>
<homeScreen>0</homeScreen>
<accessContacts>1</accessContacts>
<accessFavorites>1</accessFavorites>
<accessVoicemail>1</accessVoicemail>
<accessApps>1</accessApps>
</vendorConfig>
<versionStamp>d902ed5a-c1e5-4233-b1d6-a960d53d1c3a</versionStamp>
<!--loadInformation>sip8845_65.12-1-1-12</loadInformation-->
<!-- <addOnModules>
<addOnModule idx="1">
<deviceType></deviceType>
<deviceLine></deviceLine>
<loadInformation></loadInformation>
</addOnModule>
</addOnModules> -->
<phoneServices useHTTPS="false">
<provisioning>2</provisioning>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="2" category="0">
<name>Voicemail</name>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
<userLocale>
<name></name>
<uid>1</uid>
<langCode></langCode>
<version></version>
<winCharSet>utf-8</winCharSet>
</userLocale>
<networkLocale></networkLocale>
<networkLocaleInfo>
<name></name>
<version></version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL>http://10.10.10.34/auth</authenticationURL>
<servicesURL>http://10.10.10.34/cgi/cisco/services?name=SEPB8BEBF236B20</servicesURL>
<directoryURL>http://10.10.10.34/cgi/cisco/services?name=SEPB8BEBF236B20</directoryURL>
<messagesURL></messagesURL>
<idleURL></idleURL>
<informationURL></informationURL>
<proxyServerURL></proxyServerURL>
<secureAuthenticationURL></secureAuthenticationURL>
<secureMessagesURL></secureMessagesURL>
<secureServicesURL></secureServicesURL>
<secureDirectoryURL></secureDirectoryURL>
<secureInformationURL></secureInformationURL>
<secureIdleURL></secureIdleURL>
<transportLayerProtocol>1</transportLayerProtocol>
<TLSResumptionTimer>3600</TLSResumptionTimer>
<phonePersonalization>1</phonePersonalization>
<autoCallPickupEnable>true</autoCallPickupEnable>
<blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
<blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
<dndCallAlert>1</dndCallAlert>
<dndReminderTimer>5</dndReminderTimer>
<advertiseG722Codec>1</advertiseG722Codec>
<rollover>0</rollover>
<joinAcrossLines>0</joinAcrossLines>
<capfAuthMode>0</capfAuthMode>
<capfList></capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<sshUserId></sshUserId>
<sshPassword></sshPassword>
</device>
{noformat}
8865.cnf.xml - 107,108
{noformat}
<?xml version="1.0" encoding="UTF-8"?>
<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>Arabian Standard Time</timeZone>
<ntps>
<ntp>
<name>10.10.11.254</name>
<ntpMode>unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>10.10.10.34</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<!-- <vpnGroup>
<mtu>1290</mtu>
<failConnectTime>30</failConnectTime>
<authMethod>0</authMethod>
<pswdPersistent>1</pswdPersistent>
<autoNetDetect>1</autoNetDetect>
<enableHostIDCheck>0</enableHostIDCheck>
<addresses>
<url1></url1>
</addresses>
<credentials>
<hashAlg>0</hashAlg>
<certHash1></certHash1>
</credentials>
</vpnGroup> -->
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-cisco-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>1</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>3</anonymousCallBlock>
<callerIdBlocking>3</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
<retainForwardInformation>false</retainForwardInformation>
<uriDialingDisplayPreference>1</uriDialingDisplayPreference>
</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>true</remotePartyID>
<userInfo>Phone</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>true</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>0</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>true</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<natEnabled>false</natEnabled>
<natReceivedProcessing>false</natReceivedProcessing>
<natAddress></natAddress>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel>107</featureLabel>
<name>107</name>
<displayName>107</displayName>
<contact>107</contact>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>107</authName>
<authPassword>107</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>1</messageWaitingAMWI>
<messagesNumber>8000</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>6</maxNumCalls>
<busyTrigger>4</busyTrigger>
</line>
<line button="2" lineIndex="2">
<featureID>9</featureID>
<featureLabel>108</featureLabel>
<name>108</name>
<displayName>108</displayName>
<contact>108</contact>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<!--authName>108</authName>
<authPassword>108</authPassword-->
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messageWaitingAMWI>1</messageWaitingAMWI>
<messagesNumber>8000</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
<maxNumCalls>6</maxNumCalls>
<busyTrigger>4</busyTrigger>
</line>
</sipLines>
<externalNumberMask></externalNumberMask>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate></dialTemplate>
<softKeyFile></softKeyFile>
</sipProfile>
<MissedCallLoggingOption>1</MissedCallLoggingOption>
<featurePolicyFile></featurePolicyFile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>3</callLogBlfEnabled>
</commonProfile>
<vendorConfig>
<defaultWallpaperFile></defaultWallpaperFile>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<enableMuteFeature>false</enableMuteFeature>
<enableGroupListen>true</enableGroupListen>
<holdResumeKey>1</holdResumeKey>
<recentsSoftKey>1</recentsSoftKey>
<dfBit>1</dfBit>
<pcPort>0</pcPort>
<spanToPCPort>1</spanToPCPort>
<garp>0</garp>
<rtcp>1</rtcp>
<videoRtcp>1</videoRtcp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>1</videoCapability>
<hideVideoByDefault>0</hideVideoByDefault>
<separateMute>0</separateMute>
<ciscoCamera>1</ciscoCamera>
<usb1>1</usb1>
<usb2>1</usb2>
<usbClasses>0,1,2</usbClasses>
<sdio>1</sdio>
<wifi>1</wifi>
<bluetooth>1</bluetooth>
<bluetoothProfile>0,1</bluetoothProfile>
<btpbap>0</btpbap>
<bthfu>0</bthfu>
<ehookEnable>0</ehookEnable>
<autoSelectLineEnable>1</autoSelectLineEnable>
<autoCallSelect>1</autoCallSelect>
<incomingCallToastTimer>10</incomingCallToastTimer>
<dialToneFromReleaseKey>0</dialToneFromReleaseKey>
<joinAndDirectTransferPolicy>0</joinAndDirectTransferPolicy>
<minimumRingVolume></minimumRingVolume>
<simplifiedNewCall>0</simplifiedNewCall>
<actionableAlert>0</actionableAlert>
<showCallHistoryForSelectedLine>0</showCallHistoryForSelectedLine>
<kemOneColumn>0</kemOneColumn>
<lineMode>0</lineMode>
<allCallsOnPrimary>0</allCallsOnPrimary>
<softKeyControl>0</softKeyControl>
<settingsAccess>1</settingsAccess>
<webProtocol>0</webProtocol>
<webAccess>0</webAccess>
<webAdmin>1</webAdmin>
<adminPassword></adminPassword>
<sshAccess>0</sshAccess>
<detectCMConnectionFailure>0</detectCMConnectionFailure>
<g722CodecSupport>1</g722CodecSupport>
<handsetWidebandEnable>2</handsetWidebandEnable>
<headsetWidebandEnable>2</headsetWidebandEnable>
<headsetWidebandUIControl>1</headsetWidebandUIControl>
<handsetWidebandUIControl>1</handsetWidebandUIControl>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>10:00</displayOnDuration>
<displayIdleTimeout>00:10</displayIdleTimeout>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<displayRefreshRate>0</displayRefreshRate>
<daysBacklightNotActive>1,7</daysBacklightNotActive>
<backlightOnTime>08:00</backlightOnTime>
<backlightOnDuration>10:00</backlightOnDuration>
<backlightIdleTimeout>00:10</backlightIdleTimeout>
<backlightOnWhenIncomingCall>1</backlightOnWhenIncomingCall>
<recordingTone>0</recordingTone>
<recordingToneLocalVolume>100</recordingToneLocalVolume>
<recordingToneRemoteVolume>50</recordingToneRemoteVolume>
<recordingToneDuration></recordingToneDuration>
<moreKeyReversionTimer>5</moreKeyReversionTimer>
<peerFirmwareSharing>0</peerFirmwareSharing>
<loadServer></loadServer>
<problemReportUploadURL></problemReportUploadURL>
<enableCdpSwPort>1</enableCdpSwPort>
<enableCdpPcPort>0</enableCdpPcPort>
<enableLldpSwPort>1</enableLldpSwPort>
<enableLldpPcPort>0</enableLldpPcPort>
<cdpEnable>true</cdpEnable>
<outOfRangeAlert>0</outOfRangeAlert>
<scanningMode>2</scanningMode>
<applicationURL></applicationURL>
<appButtonTimer>0</appButtonTimer>
<appButtonPriority>0</appButtonPriority>
<specialNumbers></specialNumbers>
<sendKeyAction>0</sendKeyAction>
<powerOffWhenCharging>0</powerOffWhenCharging>
<homeScreen>0</homeScreen>
<accessContacts>1</accessContacts>
<accessFavorites>1</accessFavorites>
<accessVoicemail>1</accessVoicemail>
<accessApps>1</accessApps>
</vendorConfig>
<versionStamp>d902ed5a-c1e5-4233-b1d6-a960d53d1c3a</versionStamp>
<!--loadInformation>sip8845_65.12-1-1-12</loadInformation-->
<!-- <addOnModules>
<addOnModule idx="1">
<deviceType></deviceType>
<deviceLine></deviceLine>
<loadInformation></loadInformation>
</addOnModule>
</addOnModules> -->
<phoneServices useHTTPS="false">
<provisioning>2</provisioning>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="2" category="0">
<name>Voicemail</name>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
<userLocale>
<name></name>
<uid>1</uid>
<langCode></langCode>
<version></version>
<winCharSet>utf-8</winCharSet>
</userLocale>
<networkLocale></networkLocale>
<networkLocaleInfo>
<name></name>
<version></version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL>http://10.10.10.34/auth</authenticationURL>
<servicesURL>http://10.10.10.34/cgi/cisco/services?name=SEPB8BEBF236B20</servicesURL>
<directoryURL>http://10.10.10.34/cgi/cisco/services?name=SEPB8BEBF236B20</directoryURL>
<messagesURL></messagesURL>
<idleURL></idleURL>
<informationURL></informationURL>
<proxyServerURL></proxyServerURL>
<secureAuthenticationURL></secureAuthenticationURL>
<secureMessagesURL></secureMessagesURL>
<secureServicesURL></secureServicesURL>
<secureDirectoryURL></secureDirectoryURL>
<secureInformationURL></secureInformationURL>
<secureIdleURL></secureIdleURL>
<transportLayerProtocol>1</transportLayerProtocol>
<TLSResumptionTimer>3600</TLSResumptionTimer>
<phonePersonalization>1</phonePersonalization>
<autoCallPickupEnable>true</autoCallPickupEnable>
<blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
<blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
<dndCallAlert>1</dndCallAlert>
<dndReminderTimer>5</dndReminderTimer>
<advertiseG722Codec>1</advertiseG722Codec>
<rollover>0</rollover>
<joinAcrossLines>0</joinAcrossLines>
<capfAuthMode>0</capfAuthMode>
<capfList></capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<sshUserId></sshUserId>
<sshPassword></sshPassword>
</device>
{noformat}
If i place call from ext. 108 (second line on cisco-8865) to ext 106 (cisco-8845) i have call from 107 -> 106 and this debug messages:
DEBUG: 108 (second line) -> 106
{noformat}
<--- SIP read from TCP:10.10.11.117:50168 --->
NOTIFY sip:108 at 10.10.10.34 SIP/2.0
Via: SIP/2.0/TCP 10.10.11.117:50168;branch=z9hG4bK43b09531
To: "108" <sip:108 at 10.10.10.34>
From: "108" <sip:108 at 10.10.10.34>;tag=00b1e3bb7fbe0027129238de-1e5a92b5
Call-ID: 57dc5a2e-47f811d4 at 10.10.11.117
Session-ID: 6b204c1600105000a00000b1e3bb7fbe;remote=00000000000000000000000000000000
Date: Fri, 01 Feb 2019 09:16:38 GMT
CSeq: 7 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:108 at 10.10.11.117:50168;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00B1E3BB7FBE"
Authorization: Digest username="107",realm="asterisk",uri="",response="cf45c8a27f7091477f9e798252245a14",nonce="6722db96",algorithm=MD5
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 358
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="2" state="partial" entity="sip:108 at 10.10.11.117">
<dialog id="5" call-id="00b1e3bb-7fbe0007-06d62756-293fb6ac at 10.10.11.117" local-tag="00b1e3bb7fbe002677f573d5-69d3d234"><state>trying</state></dialog>
</dialog-info>
<------------->
--- (17 headers 4 lines) ---
Sending to 10.10.11.117:50168 (no NAT)
<--- Transmitting (no NAT) to 10.10.11.117:50168 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.11.117:50168;branch=z9hG4bK43b09531;received=10.10.11.117
From: "108" <sip:108 at 10.10.10.34>;tag=00b1e3bb7fbe0027129238de-1e5a92b5
To: "108" <sip:108 at 10.10.10.34>;tag=as4b98d765
Call-ID: 57dc5a2e-47f811d4 at 10.10.11.117
CSeq: 7 NOTIFY
Server: Asterisk PBX 13.24.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '57dc5a2e-47f811d4 at 10.10.11.117' in 32000 ms (Method: NOTIFY)
<--- SIP read from TCP:10.10.11.117:50168 --->
INVITE sip:106 at 10.10.10.34;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.11.117:50168;branch=z9hG4bK6bacd0b0
From: "Anonymous" <sip:Anonymous at 10.10.10.34>;tag=00b1e3bb7fbe002677f573d5-69d3d234
To: <sip:106 at 10.10.10.34>
Call-ID: 00b1e3bb-7fbe0007-06d62756-293fb6ac at 10.10.11.117
Max-Forwards: 70
Session-ID: 14f1b3ef00105000a00000b1e3bb7fbe;remote=00000000000000000000000000000000
Date: Fri, 01 Feb 2019 09:16:38 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP8865/12.1.1
Contact: <sip:108 at 10.10.11.117:50168;user=phone;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00B1E3BB7FBE";video
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "108" <sip:108 at 10.10.10.34>;party=calling;id-type=subscriber;privacy=full;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="107",realm="asterisk",uri="sip:106 at 10.10.10.34;user=phone",response="c1b7aed4b5571d25ac2063eb133a97e8",nonce="6722db96",algorithm=MD5
Content-Length: 1183
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 21191 0 IN IP4 10.10.11.117
s=SIP Call
b=AS:4064
t=0 0
m=audio 27626 RTP/AVP 0 8 116 18 101
c=IN IP4 10.10.11.117
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 29196 RTP/AVP 100 126 97
c=IN IP4 10.10.11.117
b=TIAS:4000000
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=640C16;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428016;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
<------------->
--- (23 headers 33 lines) ---
Sending to 10.10.11.117:50168 (no NAT)
Sending to 10.10.11.117:50168 (no NAT)
Using INVITE request as basis request - 00b1e3bb-7fbe0007-06d62756-293fb6ac at 10.10.11.117
Found peer '107' for 'Anonymous' from 10.10.11.117:50168
<--- Reliably Transmitting (no NAT) to 10.10.11.117:50168 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 10.10.11.117:50168;branch=z9hG4bK6bacd0b0;received=10.10.11.117
From: "Anonymous" <sip:Anonymous at 10.10.10.34>;tag=00b1e3bb7fbe002677f573d5-69d3d234
To: <sip:106 at 10.10.10.34>;tag=as09c74932
Call-ID: 00b1e3bb-7fbe0007-06d62756-293fb6ac at 10.10.11.117
CSeq: 101 INVITE
Server: Asterisk PBX 13.24.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f2f90c9"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '00b1e3bb-7fbe0007-06d62756-293fb6ac at 10.10.11.117' in 32000 ms (Method: INVITE)
<--- SIP read from TCP:10.10.11.117:50168 --->
ACK sip:106 at 10.10.10.34;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.11.117:50168;branch=z9hG4bK6bacd0b0
From: "Anonymous" <sip:Anonymous at 10.10.10.34>;tag=00b1e3bb7fbe002677f573d5-69d3d234
To: <sip:106 at 10.10.10.34>;tag=as09c74932
Call-ID: 00b1e3bb-7fbe0007-06d62756-293fb6ac at 10.10.11.117
Session-ID: 14f1b3ef00105000a00000b1e3bb7fbe;remote=00000000000000000000000000000000
Max-Forwards: 70
Date: Fri, 01 Feb 2019 09:16:38 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from TCP:10.10.11.117:50168 --->
INVITE sip:106 at 10.10.10.34;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.11.117:50168;branch=z9hG4bK13db0942
From: "Anonymous" <sip:Anonymous at 10.10.10.34>;tag=00b1e3bb7fbe002677f573d5-69d3d234
To: <sip:106 at 10.10.10.34>
Call-ID: 00b1e3bb-7fbe0007-06d62756-293fb6ac at 10.10.11.117
Max-Forwards: 70
Session-ID: 14f1b3ef00105000a00000b1e3bb7fbe;remote=00000000000000000000000000000000
Date: Fri, 01 Feb 2019 09:16:38 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP8865/12.1.1
Contact: <sip:108 at 10.10.11.117:50168;user=phone;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00B1E3BB7FBE";video
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "108" <sip:108 at 10.10.10.34>;party=calling;id-type=subscriber;privacy=full;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="107",realm="asterisk",uri="sip:106 at 10.10.10.34;user=phone",response="b972c8477db25755cdbc80d1df8e3a95",nonce="1f2f90c9",algorithm=MD5
Content-Length: 1183
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 21191 0 IN IP4 10.10.11.117
s=SIP Call
b=AS:4064
t=0 0
m=audio 27626 RTP/AVP 0 8 116 18 101
c=IN IP4 10.10.11.117
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 29196 RTP/AVP 100 126 97
c=IN IP4 10.10.11.117
b=TIAS:4000000
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=640C16;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428016;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428016;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=108000;max-fs=3600;max-rcmd-nalu-size=256000
a=imageattr:* recv [x=800,y=480,q=0.60] [x=1280,y=720,q=0.50]
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=sendrecv
<------------->
--- (23 headers 33 lines) ---
Sending to 10.10.11.117:50168 (no NAT)
Using INVITE request as basis request - 00b1e3bb-7fbe0007-06d62756-293fb6ac at 10.10.11.117
Found peer '107' for 'Anonymous' from 10.10.11.117:50168
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 116
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 116
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Found RTP video format 100
Found RTP video format 126
Found RTP video format 97
Found video description format H264 for ID 100
Found video description format H264 for ID 126
Found video description format H264 for ID 97
Capabilities: us - (g722|ulaw|alaw|g729|h264), peer - audio=(ulaw|alaw|g729|ilbc)/video=(h264)/text=(nothing), combined - (ulaw|alaw|g729|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0xb7406798 -- Strict RTP learning after remote address set to: 10.10.11.117:27626
Peer audio RTP is at port 10.10.11.117:27626
> 0xb7410678 -- Strict RTP learning after remote address set to: 10.10.11.117:29196
Peer video RTP is at port 10.10.11.117:29196
Looking for 106 in extensions (domain 10.10.10.34)
sip_route_dump: route/path hop: <sip:108 at 10.10.11.117:50168;user=phone;transport=tcp>
<--- Transmitting (no NAT) to 10.10.11.117:50168 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.11.117:50168;branch=z9hG4bK13db0942;received=10.10.11.117
From: "Anonymous" <sip:Anonymous at 10.10.10.34>;tag=00b1e3bb7fbe002677f573d5-69d3d234
To: <sip:106 at 10.10.10.34>
Call-ID: 00b1e3bb-7fbe0007-06d62756-293fb6ac at 10.10.11.117
CSeq: 102 INVITE
Server: Asterisk PBX 13.24.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:106 at 10.10.10.34:5060;transport=tcp>
Call-Info: <urn:x-cisco-remotecc:callinfo>; security=NotAuthenticated
Content-Length: 0
<------------>
-- Executing [106 at extensions:1] Goto("SIP/107-00000009", "call-extension,106,1") in new stack
-- Goto (call-extension,106,1)
-- Executing [106 at call-extension:1] Set("SIP/107-00000009", "PEERNAME=106") in new stack
{noformat}
If i place call from ext. 101 (second line on cisco-7965) to ext 106 (cisco-8845) i have call from 101 -> 106 and this debug messages:
DEBUG: 101 (second line) -> 106
{noformat}
<--- SIP read from TCP:10.10.10.211:49816 --->
INVITE sip:106 at 10.10.10.34;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.211:49816;branch=z9hG4bK1bdd65a7
From: "101" <sip:101 at 10.10.10.34>;tag=b8bebf236b200048db68ed18-a3b806ef
To: <sip:106 at 10.10.10.34>
Call-ID: b8bebf23-6b200008-1cef99da-278a8df9 at 10.10.10.211
Max-Forwards: 70
Date: Fri, 01 Feb 2019 09:36:30 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7965G/9.4.2
Contact: <sip:101 at 10.10.10.211:49816;transport=tcp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "101" <sip:101 at 10.10.10.34>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 358
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 26880 0 IN IP4 10.10.10.211
s=SIP Call
t=0 0
m=audio 17444 RTP/AVP 0 8 18 102 116 101
c=IN IP4 10.10.10.211
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (19 headers 16 lines) ---
Sending to 10.10.10.211:49816 (no NAT)
Sending to 10.10.10.211:49816 (no NAT)
Using INVITE request as basis request - b8bebf23-6b200008-1cef99da-278a8df9 at 10.10.10.211
Found peer '101' for '101' from 10.10.10.211:49816
<--- Reliably Transmitting (no NAT) to 10.10.10.211:49816 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 10.10.10.211:49816;branch=z9hG4bK1bdd65a7;received=10.10.10.211
From: "101" <sip:101 at 10.10.10.34>;tag=b8bebf236b200048db68ed18-a3b806ef
To: <sip:106 at 10.10.10.34>;tag=as4a58344e
Call-ID: b8bebf23-6b200008-1cef99da-278a8df9 at 10.10.10.211
CSeq: 101 INVITE
Server: Asterisk PBX 13.24.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="692fab6e"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'b8bebf23-6b200008-1cef99da-278a8df9 at 10.10.10.211' in 32000 ms (Method: INVITE)
<--- SIP read from TCP:10.10.10.211:49816 --->
ACK sip:106 at 10.10.10.34;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.211:49816;branch=z9hG4bK1bdd65a7
From: "101" <sip:101 at 10.10.10.34>;tag=b8bebf236b200048db68ed18-a3b806ef
To: <sip:106 at 10.10.10.34>;tag=as4a58344e
Call-ID: b8bebf23-6b200008-1cef99da-278a8df9 at 10.10.10.211
Max-Forwards: 70
Date: Fri, 01 Feb 2019 09:36:30 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from TCP:10.10.10.211:49816 --->
INVITE sip:106 at 10.10.10.34;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.211:49816;branch=z9hG4bK505ea3b0
From: "101" <sip:101 at 10.10.10.34>;tag=b8bebf236b200048db68ed18-a3b806ef
To: <sip:106 at 10.10.10.34>
Call-ID: b8bebf23-6b200008-1cef99da-278a8df9 at 10.10.10.211
Max-Forwards: 70
Date: Fri, 01 Feb 2019 09:36:30 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7965G/9.4.2
Contact: <sip:101 at 10.10.10.211:49816;transport=tcp>
Authorization: Digest username="101",realm="asterisk",uri="sip:106 at 10.10.10.34;user=phone",response="5523f47db3e9de5b3e59a2d493405c46",nonce="692fab6e",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "101" <sip:101 at 10.10.10.34>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 358
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 26880 0 IN IP4 10.10.10.211
s=SIP Call
t=0 0
m=audio 17444 RTP/AVP 0 8 18 102 116 101
c=IN IP4 10.10.10.211
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (20 headers 16 lines) ---
Sending to 10.10.10.211:49816 (no NAT)
Using INVITE request as basis request - b8bebf23-6b200008-1cef99da-278a8df9 at 10.10.10.211
Found peer '101' for '101' from 10.10.10.211:49816
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 102
Found RTP audio format 116
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format L16 for ID 102
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Capabilities: us - (g722|ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729|slin16|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0xb6b0da80 -- Strict RTP learning after remote address set to: 10.10.10.211:17444
Peer audio RTP is at port 10.10.10.211:17444
Looking for 106 in extensions (domain 10.10.10.34)
sip_route_dump: route/path hop: <sip:101 at 10.10.10.211:49816;transport=tcp>
<--- Transmitting (no NAT) to 10.10.10.211:49816 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.211:49816;branch=z9hG4bK505ea3b0;received=10.10.10.211
From: "101" <sip:101 at 10.10.10.34>;tag=b8bebf236b200048db68ed18-a3b806ef
To: <sip:106 at 10.10.10.34>
Call-ID: b8bebf23-6b200008-1cef99da-278a8df9 at 10.10.10.211
CSeq: 102 INVITE
Server: Asterisk PBX 13.24.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:106 at 10.10.10.34:5060;transport=tcp>
Call-Info: <urn:x-cisco-remotecc:callinfo>; security=NotAuthenticated
Content-Length: 0
<------------>
-- Executing [106 at extensions:1] Goto("SIP/101-0000000d", "call-extension,106,1") in new stack
-- Goto (call-extension,106,1)
-- Executing [106 at call-extension:1] Set("SIP/101-0000000d", "PEERNAME=106") in new stack
{noformat}
What am I doing wrong?
> [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML
> -------------------------------------------------------------------------------
>
> Key: ASTERISK-13145
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-13145
> Project: Asterisk
> Issue Type: New Feature
> Components: Channels/chan_sip/NewFeature
> Reporter: Gareth Palmer
> Assignee: Gareth Palmer
> Labels: patch, pjsip
> Attachments: 00_READ_ME_FIRST.txt, AppDialRules.xml, cisco-usecallmanager-13.24.1.patch, cisco-usecallmanager-16.1.1.patch, DialTemplate.xml, FeaturePolicy.xml, SEPMAC.cnf.xml, SoftKeys.xml
>
>
> This patch provides support for Cisco 6900, 7900, 8800 and 9900 series phones using the SIP firmware.
> Available features are: Busy Lamp Field, Off Hook Notification, Call Forward, Do Not Disturb, Huntgroup Login, Call Park (Notify and Monitor), Server-Side Ad-Hoc Conference, Conference List, Kick and Mute/Unmute, Multi-Admin Conference, Multiple Lines via Bulk Register, Immediate Divert, Call Recording, Restart or Reset via CLI, Call Pickup Notification, Call Back, Join Calls, Mallicious Call ID, Quality Reporting Tool and Fail-over/Fail-back.
> Also included is Application Server Events used by non-USECALLMANAGER phones (Call Forward and Do Not Disturb only).
> *Important:* Read the documentation at [http://usecallmanager.nz] to see the additional configuration options required for the phones to operate correctly.
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