VoIP Mailing List Archives

[gec at davisfloyd.com]

Well it would seem that Cisco chose not to make their SEP_MAC.cnf.xml format
standard across all of their phone models. I have had similar issues getting
various models to work. It has made it a challenge, for no obvious good
reason IMO. I have pasted a SEP_MAC.cnf.xml that I use for a 7941G-GE. Give
it a try if you like, or just use it for comparison....

Glenn

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<device xsi:type="axl:XIPPhone" ctiid="1566023366">
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>root</sshUserId>
<sshPassword>root</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/Ya</dateTemplate>
<timeZone>Eastern Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>10.10.30.10</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>10.10.30.10</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>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>true</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled></natEnabled>
<natAddress></natAddress>
<phoneLabel>Ext 3105</phoneLabel>
<stutterMsgWaiting>0</stutterMsgWaiting>
<callStats>false</callStats>

<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBurs
ts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>3105</featureLabel>
<proxy>10.10.30.10</proxy>
<port>5060</port>
<name>3105</name>
<displayName>3105</displayName>
<autoAnswer>
<autoAnswerEnabled>1</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>3105</authName>
<authPassword>3105</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>3105</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID>9</featureID>
<featureLabel>3105</featureLabel>
<proxy>10.10.30.10</proxy>
<port>5060</port>
<name>3105</name>
<displayName>3105</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>3105</authName>
<authPassword>3105</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>3105</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</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>3105</phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<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>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>

<versionStamp>1136931633-57191cee-5ffc-4342-b286-4246b4991890</versionStamp>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en</langCode>
<version>8.3(3)</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<version>8.3(3)</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>4</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
</device>
_____

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gregory Wong
Sent: Friday, January 25, 2008 6:36 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Unprovisioned 7961


Hi Everyone,

I am having some issues getting my 7961 working with Trixbox. I have loaded
the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an
unprovisioned state. A status message shows up and says "Error Verifying
Config Info".

I have read quite a bit on this topic (getting 7961's to work with Asterisk
and TB) and only came across a few postings where other people encountered
this issue but no solution was given. I have checked the SEP.cnf.xml file
for the phone and everything seems to be right. I even tried to remove some
parts of the code as people suggested but no luck. I already have a 7960 on
TB so I know that TFTP is working correctly.

Any ideas on how I can get this to work would be much appreciated.

Thank.

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[cosmond at messagelabs...]

Try this configuration file...

http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configu
ration+files+for+SIP#Cisco7961with833SR2ConfigurationExamples
Chad
________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gregory
Wong
Sent: Friday, January 25, 2008 6:36 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Unprovisioned 7961


Hi Everyone,

I am having some issues getting my 7961 working with Trixbox. I have
loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes
into an unprovisioned state. A status message shows up and says "Error
Verifying Config Info".

I have read quite a bit on this topic (getting 7961's to work with
Asterisk and TB) and only came across a few postings where other people
encountered this issue but no solution was given. I have checked the
SEP.cnf.xml file for the phone and everything seems to be right. I even
tried to remove some parts of the code as people suggested but no luck.
I already have a 7960 on TB so I know that TFTP is working correctly.

Any ideas on how I can get this to work would be much appreciated.

Thank.
______________________________________________________________________
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[gwong at wong-consulti...]

Thanks Chad. This config seemed to have worked a bit. I don't get the
"Unprovisioned" or "Error Verifying Config Info" messages anymore. However,
the phone sits at "Registering" and will never register.

I took a look at the sip debug and I see the below messages. Do I need to
enable NAT in the SEP.cnf.xml file since I am behind NAT? I know my 7960
config file has natEnabled = 1.

Scheduling destruction of call
'0018195a-a6770002-4ba9e20e-e89879bd at 192.168.15.100' in 15000 ms

<-- SIP read from <MY HOME IP ADDRESS>:49157:
REGISTER sip:<TB IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <MY HOME IP ADDRESS>:1140;branch=z9hG4bK48e89c16
From: <sip:86003@<TB IP ADDRESS>>;tag=0018195aa6770003efaf5095-54a486b0
To: <sip:86003@<TB IP ADDRESS>>
Call-ID: 0018195a-a6770003-b6544a3f-d9fc3a32 at 192.168.15.100
Max-Forwards: 70
Date: Mon, 08 Oct 2007 23:42:08 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7961G/8.3.0
Contact: <sip:86003@<HOME IP
ADDRESS>:1140;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-000
0-0018195aa677>";+u.sip!model.ccm.cisco.com="30018"
Supported: (null),X-cisco-xsi-6.0.2
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:25 Name=SEP0018195AA677
Load=SIP41.8-3-3SR2S Last=initialized"
Expires: 3600
--- (14 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to <HOME IP ADDRESS> : 1140 (non-NAT)
Transmitting (no NAT) to <HOME IP ADDRESS>:1140:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP <HOME IP
ADDRESS>:1140;branch=z9hG4bK48e89c16;received=<HOME IP ADDRESS>
From: <sip:86003@<TB IP ADDRESS>>;tag=0018195aa6770003efaf5095-54a486b0
To: <sip:86003@<TB IP ADDRESS>>;tag=as1886ecd1
Call-ID: 0018195a-a6770003-b6544a3f-d9fc3a32 at 192.168.15.100
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


On 1/25/08 10:29 AM, "Chad Osmond" <cosmond at messagelabs.com> wrote: