Goautodial CISCO IP Phone

[edmund.uba]

Greetings!

I am new to VOIP and BPO. I recently installed goautodial for almost 8 months now, and its working. We are using the version 3.0. I want to install or use cisco IP phone, the problem is, I dont know how. I already have a cisco unified call manager installed. Can anybody help?

[williamconley]

Vicidial (the software underneath your goautodial server) is a management system that turns asterisk into a predictive autodial system. Asterisk handles all the actual voip communications.

That being said, you don't want to connect "Cisco phones" to "GoAutodial". In fact you do not want to connect them to "Vicidial" either. You want to connect them to "Asterisk". The method of connection would be to create a sip account for each phone and then program the phones to connect to those sip accounts.

Vicidial will allow you to create the accounts under Admin->Phones. Then you will use the same method to connect those phones to your asterisk server as you have with your existing phones, except of course you already know how to program the existing phones and you have to "figure out" how to program your new cisco phones. That, unfortunately, is a bit outside the scope of this board. However, there are numerous boards for asterisk, there may even be history on this board, and there should certainly be support available from whomever sold you the phones.

You will need to use the "Extension" (in admin->phones) as the "user/account/auth id" (or whatever cisco wants to call it today) and the "Registration password" for the "secret/password" (or whatever cisco wants to call it today) plus the IP of the server.

Some phones can be programmed right from the handset. Some have their own "mini-web-server" and some require a local TFTP service to program them. Each phone model is different and each manufacturer even differs between models.

We've programmed many for clients both remotely and here in the office (some are complex and require "in-person" to make it happen). The only challenge is that if the phones are "locked" to a previous server ... well, if you've ever had a locked cell phone you know what that means.

[edmund.uba]

Thanks for the info. I did a research and studied the things you have said. I already configured a TFTP server for the CISCO phone. Sadly, i hard reset the cisco phone since I have to convert them to use SIP instead of SCCP. The problem is the SEP<MAC>.xml file. I copied some configuration over the internet and did some editing. I hope my configuration is correct.

<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>admin</sshPassword>
<devicePool>
<name>RTI-ANS</name>
<dateTimeSetting>
<name>CMLocal</name>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>China Standard/Daylight Time</timeZone>
</dateTimeSetting>
<callManagerGroup>
<name>5.0 Beta</name>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<name>RTI_ANS_DAVAO</name>
<description>Call Manager RTI-ANS Davao</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>RTI_ANS_DAVAO</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-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>false</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</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>
<phoneLabel>9001</phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>false</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>

<sipLines> #I think this is where the goautodial phone settings and registration is entered here. Still unsure if I did it right
<line button="1">
<featureID>9</featureID>
<featureLabel>9001</featureLabel>
<proxy>192.168.4.254</proxy> #This is the goautodial address, Im unsure if it goes here.
<port>5060</port>
<name>9001</name>
<displayName>9001</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>9001</authName>
<authPassword>contact690</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>Freeswitch_Voicemail_Number</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>Freeswitch_Extension</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>softkeyDefault.xml</softKeyFile>
</sipProfile>

<commonProfile>
<phonePassword>cisco</phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>1</callLogBlfEnabled>
</commonProfile>

<loadInformation>SIP45.8-5-2S</loadInformation>

<vendorConfig>
<g722CodecSupport>2</g722CodecSupport>
<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>1</webAccess>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>10:30</displayOnDuration>
<displayIdleTimeout>01:00</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
</vendorConfig>

<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>Philippines</networkLocale>
<networkLocaleInfo>
<name>Philippines</name>
<uid>64</uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>

<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<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>
<advertiseG722Codec>1</advertiseG722Codec>
</device>

[edmund.uba]

i simplified the configuration because its a headache...

<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>admin</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>China Standard/Daylight Time</timeZone>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<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>0</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>g711ulaw</preferredCodec>
<natEnabled>true</natEnabled>
<phoneLabel>9001</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>9001</featureLabel>
<proxy>192.168.4.254</proxy>
<port>5060</port>
<name>9001</name>
<displayName>9001</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>9001</authName>
<authPassword>contact690</authPassword>
<sharedLine>false</sharedLine>
<contact>9001</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<loadInformation>SIP41.8-2-2SR3S</loadInformation>
</device>


im still having a problem, the phone is always registering and it cannot connect to the goautodial....

[satpal kumar]

Hi all,the same problem i also facing with cisco 7940g ip phone please if anybody knows help...