[asterisk-users] configuring a Cisco 7961 so that different line appearances register to different SIP proxy addresses
Ken Alker
ken at impulse.net
Sun Jan 15 02:35:18 CST 2012
Hi,
I have been using Cisco 7960's with Asterisk for years. I am trying get a
7961 working and have a problem. In my configuration, not all of my line
appearances register to the same Asterisk SIP server. I have an Asterisk
server at home and another at work. My Line 1 button registers to the home
server and my Line 2 button registers to the work server. This has worked
for years with a 7960, but I can't get it to work with a 7961.
A copy of my SEP*.cnf.xml file can be found below. What I have found is
that when Line 1's "proxy" variable is defined as "USECALLMANAGER", Line 1
registers successfully to the IP address defined with the "processNodeName"
variable but it tries to register Line 2 to that same IP address instead of
the one defined via Line 2's "proxy" variable. The Proxy Address field for
both lines (on the phone itself) are blank. When I set Line 1's "proxy"
variable to the IP address of my home Asterisk server and Line 2's "proxy"
variable to the IP address of my work Asterisk server those IP addresses
are now visible in the Proxy Address fields on the phones, but the phone
simply will not transmit any SIP packages at all. I'm using WireShark to
sniff LAN traffic and I see the phone successfully negotiate DHPC and TFTP
and then it just sits there with the "Registering" message at the bottom
left (and the spinning wheel) but never throws out any SIP packets (just
lots of NTP).
Either a) I have something misconfigured in my SEP*.cnf.xml file, b) I need
another *cnf.xml file with some more parameters in it that I'm unaware of
(I just created the one config file), c) there is a bug in the firmware, or
d) Cisco no longer supports the assignment of different proxy servers to
different line appearances [I hope not!]
I have tried both 9.2(1) and 9.3(1) firmware.
Is there anyone out there who has different line appearances successfully
registering to different proxy servers with a 7961 (or some other phone
that uses SEP*cnf.xml style configuration files)? If so, would you let me
know phone model you are using, what version of firmware, and shoot me a
copy of your SEP*cnf.xml file?
Thank you!
Ken Alker
Impulse Internet Services
http://www.impulse.net
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/YYYYa</dateTemplate>
<timeZone>Pacific Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>10.0.0.9</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>10.0.0.9</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<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>1</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>true</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g711ulaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<phoneLabel>Alker Study</phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>true</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>Home Line</featureLabel>
<proxy>10.0.0.9</proxy>
<port>5060</port>
<name>study_line1</name>
<displayName>Alker Study</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>study_line1</authName>
<authPassword>youguessit</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>400</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>study_line1</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID>9</featureID>
<featureLabel>Dad's Line</featureLabel>
<proxy>x.y.z.m</proxy> <---- actually a public IP that I
don't want to publish
<port>5060</port>
<name>study_line2</name>
<displayName>Alker Study</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>study_line2</authName>
<authPassword>youguessit2</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>400</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>study_line2</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>1</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP41.9-2-1S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>1</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>00:00</displayOnTime>
<displayOnDuration>00:00</displayOnDuration>
<displayIdleTimeout>00:00</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp>
<networkLocale>US</networkLocale>
<networkLocaleInfo>
<name>US</name>
<version>5.0(2)</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>
</device>
More information about the asterisk-users
mailing list