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sil at infiltrated.net Guest
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mattgibson.ca at gmail... Guest
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Posted: Sat Mar 29, 2008 4:25 am Post subject: [asterisk-users] Cisco 7971 |
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Make sure you are using md5secret for your password, and turn off the
regular secret. Here's my file working on a 7970 with SIP 8.3.3
-----
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>root</sshUserId>
<sshPassword>supersecretone</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/Ya</dateTemplate>
<timeZone>Eastern Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>136.159.2.2</name>
<ntpMode>Unicast</ntpMode>
</ntp>
<ntp>
<name>192.43.244.18</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<name>YOUR.PBX.IP.HERE</name>
<description>AsterPBX</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>YOUR.PBX.IP.HERE</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<mlppDomainId>-1</mlppDomainId>
<mlppIndicationStatus>Default</mlppIndicationStatus>
<preemption>Default</preemption>
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<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>true</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>g711u</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel>Flewid Inc</phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>x123 - Line 1</featureLabel>
<proxy>YOUR.PBX.IP.HERE</proxy>
<name>123</name>
<displayName>Your Name</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>123</authName>
<authPassword>321</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*98</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>123</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="6">
<featureID>9</featureID>
<featureLabel>Intercom</featureLabel>
<proxy>YOUR.PBX.IP.HERE</proxy>
<name>124</name>
<displayName>Intercom</displayName>
<autoAnswer>
<autoAnswerEnabled>3</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>124</authName>
<authPassword>421</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*98</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>124</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>
<softKeyFile>softkey.xml</softKeyFile>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP70.8-3-3S</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>
<webAccess>0</webAccess>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>8:00</displayOnTime>
<displayOnDuration>10:30</displayOnDuration>
<displayIdleTimeout>00:10</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
</vendorConfig>
<versionStamp>1136931633-57191cee-5ffc-4342-b286-4246b4991890</versionStamp>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en_US</langCode>
<version>1.0.0.0-1</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<idleURL></idleURL>
<authenticationURL>http://YOUR.PBX.IP.HERE/cisco/authenticate.php</authenticationURL>
<directoryURL>http://YOUR.PBX.IP.HERE/cisco/directory.php</directoryURL>
<informationURL>http://YOUR.PBX.IP.HERE/cisco/help.php</informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://YOUR.PBX.IP.HERE/cisco/services.php</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
<processNodeName>ccm-beta-5-1</processNodeName>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<natReceivedProcessing>false</natReceivedProcessing>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
</device>
Thanks,
Matt
On Fri, Mar 28, 2008 at 2:58 PM, J. Oquendo <sil at infiltrated.net> wrote:
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asterisk-list at puzzl... Guest
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Posted: Sat Mar 29, 2008 12:13 pm Post subject: [asterisk-users] Cisco 7971 |
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On Sat, 2008-03-29 at 05:25 -0400, Matthew Gibson wrote:
Quote: | Make sure you are using md5secret for your password, and turn off the
regular secret. Here's my file working on a 7970 with SIP 8.3.3
| [snip big cisco config file]
Maybe it has a different name but I don't see any option containing
"md5" in the config you pasted. What is the md5 option called? I would
like to setup md5 authentication between my 7961 on SIP 8.3.3 with
Asterisk 1.4.18.
Thanks,
Patrick |
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sil at infiltrated.net Guest
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Posted: Mon Mar 31, 2008 6:33 am Post subject: [asterisk-users] Cisco 7971 |
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Matthew Gibson wrote:
Quote: | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret
then in your sip.conf
[ext]
...
;secret=123
md5secret=MD5SECRET
|
Hey Martin, thanks for your response... Still no dice:
Quick questions... Where are the following coming from? Is this
something you placed, something generated, if so by what, CCM, the phone
itself.
<authenticationURL>http://YOUR.PBX.IP.HERE/cisco/authenticate.php</authenticationURL>
<directoryURL>http://YOUR.PBX.IP.HERE/cisco/directory.php</directoryURL>
<informationURL>http://YOUR.PBX.IP.HERE/cisco/help.php</informationURL>
<servicesURL>http://YOUR.PBX.IP.HERE/cisco/services.php</servicesURL>
Second...
<loadInformation>SIP70.8-3-3S</loadInformation>
I don't have SIP70.8-3-3s I have term71.default.loads which includes all
images listed inside the file:
# cat term71.default.loads
# This file contains a list of archive image files that will be
requested by the
# RELEASE load version 8-3-3ES2
#
jar70sip.8-3-3ES2.sbn
cnu70.8-3-3ES2.sbn
apps70.8-3-3ES2.sbn
dsp70.8-3-3ES2.sbn
cvm70sip.8-3-3ES2.sbn
I tried posting both term71.default and cvm70sip.8-3-3ES2
<loadInformation>term71.default</loadInformation>
<loadInformation>cvm70sip.8-3-3ES2</loadInformation>
For NAT, when I have it set to true on SEPxxxx.xml, phone registers and
this is what happens in the course of 5 seconds:
<natReceivedProcessing>true</natReceivedProcessing>
<natEnabled>true</natEnabled>
-- Registered SIP '99999' at 64.xxx.xxx.xx port 49344 expires 3600
-- Saved useragent "Cisco-CP7971G-GE/8.3.0" for peer 99999
[Mar 31 07:17:02] NOTICE[2743]: chan_sip.c:15322 sip_poke_noanswer: Peer
'99999' is now UNREACHABLE! Last qualify: 0
On sip show peer: (truncated)
ToHost : 64.xxx.xxx.xx
Addr->IP : 64.xxx.xxx.xx Port 49344
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 123
SIP Options : (none)
Codecs : 0x104 (ulaw|g729)
Codec Order : (g729:20,ulaw:20)
Auto-Framing: No
Status : UNREACHABLE
Useragent : Cisco-CP7971G-GE/8.3.0
Reg. Contact : sip:123 at 192.168.1.145:5060;transport=udp
So I set contact to match:
astterm*CLI>
-- Registered SIP '99999' at 192.168.1.145 port 5060 expires 3600
-- Saved useragent "Cisco-CP7971G-GE/8.3.0" for peer 99999
[Mar 31 07:28:12] NOTICE[2743]: chan_sip.c:15322 sip_poke_noanswer: Peer
'99999' is now UNREACHABLE! Last qualify: 0
Now it matches but the same disconnect occurs:
sip show peer truncated
ToHost : 64.xxx.xxx.xx
Addr->IP : 192.168.1.145 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 99999
SIP Options : (none)
Codecs : 0x104 (ulaw|g729)
Codec Order : (g729:20,ulaw:20)
Auto-Framing: No
Status : UNREACHABLE
Useragent : Cisco-CP7971G-GE/8.3.0
Reg. Contact : sip:99999 at 192.168.1.145:5060;transport=udp
About to kick this 7971
Nope, no firewall, clean connection, and no NAT is being used period.
Most appreciated response if any. I'm definitely scratching my head on
this one. 7970's I have working fine, never had a problem getting those
to work. I'm wondering if its the sip firmware version I'm using at this
point.
====================================================
J. Oquendo
SGFA #579 (FW+VPN v4.1)
SGFE #574 (FW+VPN v4.1)
wget -qO - www.infiltrated.net/sig|perl
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x3AC173DB
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