[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
Mon May 1 06:46:10 CDT 2017


    [ https://issues.asterisk.org/jira/browse/ASTERISK-13145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236809#comment-236809 ] 

Andrey Zharkov commented on ASTERISK-13145:
-------------------------------------------

SEPB8BEBF236B20.cnf.xml
{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>10.10.10.254</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>
    <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_udp.xml</softKeyFile>
	<dialTemplate>dialplan_3.xml</dialTemplate>
	<phoneLabel>123456789012</phoneLabel>

	<sipLines>
	    <line button="2" lineIndex="2">
		<featureID>9</featureID>
		<featureLabel>101</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>
	    <line button="3" lineIndex="3">
		<featureID>9</featureID>
		<featureLabel>103</featureLabel>
		<name>103</name>
		<displayName>103</displayName>
		<contact>103</contact>
		<proxy>USECALLMANAGER</proxy>
		<port>5060</port>
		<autoAnswer>
		    <autoAnswerEnabled>2</autoAnswerEnabled>
		</autoAnswer>
		<callWaiting>1</callWaiting>
 
		<authName>103</authName>
		<authPassword>103</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="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="4">
		<featureID>21</featureID>
    		<featureLabel>199</featureLabel>
                <speedDialNumber>199</speedDialNumber>
                <featureOptionMask>1</featureOptionMask>
            </line>
	</sipLines>
    </sipProfile>
</device>
{noformat}

sip.conf
{noformat}
[general]
t38pt_udptl=yes
callerid=Unknown
limitonpeers=yes
videosupport=yes
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=10.10.10.34
transport=tcp
allowguest=no
subscribecontext=extensions
callcounter=yes
nat=no
progressinband=yes
prematuremedia=no
alwaysauthreject=yes
notifyhold=yes
notifycid=yes


[extension](!)
type=friend
host=dynamic
context=extensions
nat=no
trustrpid=no
sendrpid=rpid
rpid_update=yes
rpid_immediate=yes
allowsubscribe=yes
notifyhold=yes
callcounter=yes
videosupport=no
disallow=all
allow=ulaw
allow=alaw
allow=g729

[cisco-usecallmanager](!,extension)
; Only use TCP as the transport protocol
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=yes

[cisco-79xx](!,cisco-usecallmanager)

[100](cisco-79xx)
secret=100
callerid="100" <100>
description=100
callgroup=1
pickupgroup=1
mailbox=100 at all
subscribe=101
subscribe=103
subscribe=199
register=101
register=103
qualify=yes

[101](cisco-79xx)
secret=101
callerid="101" <101>
description=101
callgroup=1
pickupgroup=1
mailbox=101 at all

[103](cisco-79xx)
secret=103
callerid="103" <103>
description=100
callgroup=1
pickupgroup=1
mailbox=103 at all

[199](extension)
secret=199
callerid="199" <199>
description=199
callgroup=1
pickupgroup=1
mailbox=199 at all
{noformat}

Thank you.

> [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: David McNett
>            Assignee: Gareth Palmer
>         Attachments: 00_READ_ME_FIRST.txt, cisco-usecallmanager-11.25.1.patch, cisco-usecallmanager-13.15.0.patch, dialtemplate.xml, featurepolicy.xml, SEP000000000000.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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