Tag Archives: asterisk

Cisco 7961G with Asterisk

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We got a couple of refurbished Cisco 7961G phones that I wanted to use as SIP phones with an Asterisk server. If I knew it would be so hard, I would have thrown them away and bought some decent SIP phones. However, I did not know it would be hard, and I hate to give up 🙂

Just as a reference I wanted to share the config I used, and the problems I ran into.

First, the Cisco 7961G phones do not support SIP by default. They only speak SCCP, which we do not use with Asterisk. I downloaded the SIP version of the firmware with Cisco (cmterm-7941_7961-sip.8-0-4SR1.zip). I think you can also find this version somewhere on the internet. Be careful not to upgrade to the highest version in one go. You can only upgrade one version at a time. So don’t upgrade to the latest version. Take a loot at the following URL, as not all firmware versions are free of bugs…
Voip-info on 7961G

Second, when I got the firmware loaded, after I bricked the phone a couple of times, I could start the config. I have some 7960’s and they work with simple text config files. The 7961G use XML config. If there is a config error the config will not load an the phone is unprovisioned. Also I did not find a way to change the config on the phone menu itself. I’ve pasted the XML config below. Be sure to change it for your environment.

Thirdly, I had to configure the sip user in Asterisk. The phone does not support NAT and qualify. I’ve pasted the config below as well.

Last but not least… the phone does not support NAT. Where our old 7960’s work fine over NAT, this phone only accepts SIP traffic back on port 5060 which by default makes it impossible (or unmanagable) to have multiple phones behind NAT. I have configured a VPN tunnel between our Asterisk server and the location where the phones are.

I hope this helps someone save some time on configuring the 7961G for SIP with Asterisk 🙂

XML config file (SEPXXXXXXXXXXXX.cnf.xml):

<device>

<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/Ya</dateTemplate>
<timeZone>Eastern Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>10.10.10.10</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>10.10.10.11</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>FirstnameUser</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>line1</featureLabel>
<proxy>10.10.10.11</proxy>
<port>5060</port>
<name>100</name>
<displayName>Firstname</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>100</authName>
<authPassword>password</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>100</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button=”2″>
<featureID>9</featureID>
<featureLabel>line2</featureLabel>
<proxy>10.10.10.11</proxy>
<port>5060</port>
<name>101</name>
<displayName>Firstname2</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>101</authName>
<authPassword>password</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>101</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>SIP41.8-0-4SR1S</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>1</webAccess>
<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>proxy:3128</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>

asterisk config:

[100]
username = 100
transfer = yes
mailbox = 100
call-limit = 100
fullname = Username
registersip = no
host=dynamic
callgroup = 1
qualify=no
context = employees
cid_number = 100
hasvoicemail = no
vmsecret =
email =
canreinvite=no
threewaycalling = yes
hasdirectory = no
callwaiting = yes
hasmanager = no
hasagent = yes
hassip = yes
hasiax = yes
secret = password
nat=no
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
autoprov = no
label =
macaddress =
linenumber = 1
LINEKEYS = 1
disallow = all
allow = g729,alaw,ulaw,gsm

 

SIP with Speedtouch ADSL modem

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We have a small VOIP network with 10 phone, a dedicated DSL line from Orange/Online and an external Asterisk server in a datacenter. The DSL line is terminated on a Speedtouch modem. With the default settings of the modem we experienced two problems:

  • Incoming calls did not get through
  • The sound of outgoing calls disappeared while the call was not dropped

In the Asterisk logs we could see the following message:

[Dec 30 13:40:51] WARNING[1911] chan_sip.c: Maximum retries exceeded on transmis
sion 0016c7ea-28120012-73ca27ca-35d5391a@10.0.0.45 for seqno 102 (Critical Respo
nse) -- See doc/sip-retransmit.txt.
[Dec 30 13:40:51] WARNING[1911] chan_sip.c: Hanging up call 0016c7ea-28120012-73
ca27ca-35d5391a@10.0.0.45 - no reply to our critical packet (see doc/sip-retrans
mit.txt).

To fix this, you have to disable the SIP helper on the Speedtouch modem. Connect to the modem with telnet (default ip: 10.0.0.138, default user: Administrator, default password: ) and enter the following commands:

_{Administrator}=>connection
{Administrator}[connection]=>appconfig application=SIP SIP_ALG=disabled
{Administrator}[connection]=>exit