Press the menu softkey
Scroll over to network Config
Press the * key
Press the # key
Press the # key
Press the green talk button
Press the select softkey you should now see more to the menu
To Change DHCP TFTP.
Go back to the menu root
Scroll over to profiles
Select Network profile
Select a profile that is nit auto (IE Profile1, 2, or 3)
Edit those profile settings and apply
You’ve Got Mail! Alot…
Click on Play.
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Memo
Cisco 7971 Color LCD Integration with Asterisk PBX
When you have the correct SIP firmware installed on your phone you will be ready to configure your phone to
run on an Asterisk PBX. You can either do this via a TFTP configuration (recommended) file or manually
using the menu buttons on the phone itself.
TFTP File Method
Create a file called XMLDefault.cnf.xml which lives in the root folder of your TFTP server. There are many
settings you can define in the XMLDefault.cnf.xml file but most you can leave set as the default. Typically you
put all of your Global settings in the XMLDefault.cnf.xml file and then any phone specific settings in another
file in the format SEP0019E84FDD17.cnf.xml (0019E84FDD17 is the MAC address located on the back of the
phone).
You can also put the phone specific configuration files in a sub-directory such as sip_phone under the root
folder of your TFTP server. The following is an example of what you might set in the XMLDefault.cnf.xml file
for your Asterisk PBX. The configuration assumes that you are running firmware version SIP70.8-5-4S.
Be careful with the syntax of the .xml files as they will return an error if not correct that may take a bit of time
to find.
XMLDefault.cnf.xml
<Default>
<callManagerGroup>
<members>
<member priority=”0″>
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName></processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation8 model=”IP Phone 7940″>P003-08-9-00</loadInformation8>
<loadInformation7 model=”IP Phone 7960″>P003-08-9-00</loadInformation7>
<loadInformation6 model=”IP Phone 7970″>SIP70.8-5-4S</loadInformation6> **identifies
the filename to LOAD (SIP70.8-5-4S.loads)
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>
SEP0019E84FDD17.cnf.xml
<device xsi:type=”axl:XIPPhone” ctiid=”203849429″ uuid=”{96f8508b-10ef-f98c-d20d-
0471777ec725}”>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>user</sshUserId>
<sshPassword>pass</sshPassword>
<devicePool uuid=”{a755aa55-089c-2b47-9603-c7d51b9ca4b5}”>
<name>Dallas 5.0 Beta</name>
<dateTimeSetting uuid=”{9ec4850a-7748-11d3-bdf0-00108302ead1}”>
<name>CMLocal</name>
<dateTemplate>D/M/Ya</dateTemplate>
<timeZone>New Zealand Standard/Daylight Time</timeZone>
</dateTimeSetting>
<callManagerGroup>
<name>5.0 Beta</name>
<tftpDefault>true</tftpDefault>
<members>
<member priority=”0″>
<callManager>
<name>192.168.99.15</name>
<description>Your PBX</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>ccm-beta-5-1</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo uuid=”{cd241e11-4a58-4d3d-9661-f06c912a18a3}”>
<name>Disable</name>
<srstOption>Disable</srstOption>
<userModifiable>false</userModifiable>
<ipAddr1>192.168.99.15</ipAddr1>
<port1>2000</port1>
<ipAddr2></ipAddr2>
<port2>2000</port2>
<ipAddr3></ipAddr3>
<port3>2000</port3>
<sipIpAddr1>192.168.99.15</sipIpAddr1>
<sipPort1>5060</sipPort1>
<sipIpAddr2></sipIpAddr2>
<sipPort2>5060</sipPort2>
<sipIpAddr3></sipIpAddr3>
<sipPort3>5060</sipPort3>
<isSecure>false</isSecure>
</srstInfo>
<mlppDomainId>-1</mlppDomainId>
<mlppIndicationStatus>Default</mlppIndicationStatus>
<preemption>Default</preemption>
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<sipProfile>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort>5060</emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort>5060</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>true</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel>your name</phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button=”1″>
<featureID>9</featureID>
<featureLabel>220</featureLabel>
<proxy>192.168.99.15</proxy>
<port>5060</port>
<name>yourextn</name>
<displayName>Your name</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>yourextn</authName>
<authPassword>yourextnpassword</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>yourextn</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button=”2″>
<featureID>9</featureID>
<featureLabel>yourextn</featureLabel>
<proxy>192.168.99.15</proxy>
<port>5060</port>
<name>yourextn</name>
<displayName>Your name</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>yourextn</authName>
<authPassword>yourextnpassword</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>yourextn</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button=”4″>
<featureID>2</featureID>
<featureLabel>Conf</featureLabel>
<speedDialNumber>850</speedDialNumber>
</line>
<line button=”7″>
<featureID>2</featureID>
<featureLabel>Bob DDI</featureLabel>
<speedDialNumber>2123456</speedDialNumber>
</line>
<line button=”8″>
<featureID>2</featureID>
<featureLabel>Pickup</featureLabel>
<speedDialNumber>*8</speedDialNumber>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile>SK50719900-3bee-4594-bc3f-6400e1a33bf0.xml</softKeyFile>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP70.8-5-4S</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>1</webAccess>
<daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>10:30</displayOnDuration>
<displayIdleTimeout>00:10</displayIdleTimeout>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<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>New_Zealand</networkLocale>
<networkLocaleInfo>
<name>New_Zealand</name>
<version>5.0(2)</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL></authenticationURL>
<directoryURL>http://192.168.99.15/xmlservices/directory.xml</directoryURL>
<idleURL></idleURL>
<informationURL>http://192.168.99.15/ccmcip/GetTelecasterHelpText.jsp</informationURL>
<messagesURL></messagesURL>
<proxyServerURL>192.168.99.15</proxyServerURL>
<servicesURL>http://192.168.99.15/xmlservices/service.xml</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>
</device>
Manual method
The manual method basically means programming your settings via the buttons on your phone. This can be
quite time consuming so be prepared to spend some time setting your phone up. Before you can edit your
settings you need to unlock the configuration. Press the settings button then **# to unlock. You can’t
configure everything manually so your best bet really is to get the TFTP file method mentioned previously,
working correctly on your LAN.
Files required in the TFTP directory
Don’t forget the syntax is very important when creating .xml files.
You will need a file called dialplan.xml in the TFTP root directory. This tells the phone what dialing
instructions to follow when making calls. An example below basically means, * allow anything dialed and wait
for 3 seconds before dialing.
<DIALTEMPLATE>
<TEMPLATE MATCH=”*” Timeout=”3″/> <!– Anything else –>
</DIALTEMPLATE>
Customising your phone ringtones
If you want to use different ring tones then the defaults, set up a file called distinctiveringlist.xml in the TFTP
root directory. In this is the list of ringtones that you want available to the phone. As well as the file you will
also need the ringtones in .pcm or .raw formats.
<CiscoIPPhoneRingList>
<Ring>
<DisplayName>Ring 1</DisplayName>
<FileName>ring1.pcm</FileName>
</Ring>
<Ring>
<DisplayName>Ring 2</DisplayName>
<FileName>ring2.raw</FileName>
</Ring>
</CiscoIPPhoneRingList>
Customising your phone menu buttons
If you want to make use of the directory and services buttons, set up a folder in your web directory of your
Asterisk PBX – /var/www/html. Call it xmlservices. In this folder you can set up the appropriate files for the
directory and services buttons. A directory example in xml format as below:
<?xml version=”1.0″ encoding=”ISO-8859-1″ ?>
<CiscoIPPhoneMenu>
<Title>Directory Services</Title>
<Prompt>Choose one of the above</Prompt>
<MenuItem>
<Name>Company Directory</Name>
<URL>http://yourPBXIP/xmlservices/companydirectory.xml</URL>
</MenuItem>
<MenuItem>
<Name>Skype Directory</Name>
<URL>http://yourPBXIP/xmlservices/skypedirectory.xml</URL>
</MenuItem>
<MenuItem>
<Name>Suppliers Directory</Name>
<URL>http://yourPBXIP/xmlservices/suppliersdirectory.xml</URL>
</MenuItem>
</CiscoIPPhoneMenu>
In the same directory set up 3 files like the companydirectory.xml, the skypedirectory.xml and the
suppliersdirectory.xml files (change to suit your needs) A company directory example in xml format as below:
<?xml version=”1.0″ encoding=”ISO-8859-1″ ?>
<CiscoIPPhoneDirectory>
<Title>Company Directory</Title>
<Prompt>Call this person:</Prompt>
<DirectoryEntry>
<Name>Bob – Home</Name>
<Telephone>034567890</Telephone>
</DirectoryEntry>
<DirectoryEntry>
<Name>Bob – Cell</Name>
<Telephone>0272345678</Telephone>
</DirectoryEntry>
<DirectoryEntry>
<Name>Bob – Work DDI</Name>
<Telephone>0398765432</Telephone>
</DirectoryEntry>
</CiscoIPPhoneDirectory>
You can adapt the above files as services eg have a list of local taxi companies or a list of pizza delivery
numbers by using the same 2 tier file system. Adjust the SEP0019E84FDD17.cnf.xml file accordingly.
Customizing your phone screen/logo
You can display your company logo or a picture on your phones screen. Use an image manipulation program
of your choice to create two Portable Network Graphics (PNG) files for each image:
• Full size image: 320 pixels (width) x 212 pixels (height) X 16 Depth
• Thumbnail image:80 pixels (width) x 53 pixels (height) X 16 Depth
The size ratio of these two images is 4 to 1.
Set up a folder called Desktops in the root folder of your TFTP server. In this folder create another folder
called 320x212x12. Your images in .png format go in here as well as a file you need to create called List.xml.
The List.xml file can include up to 50 background images. The images are in the order that they appear in the
Background Images menu on the phone. For each image, the List.xml file contains one element type, called
ImageItem. The ImageItem element includes these two attributes:
• Image_The uniform resource identifier (URI) that specifies where the phone obtains the thumbnail
image that appears on the Background Images menu on a phone.
• URL_The URI that specifies where the phone obtains the full size image.
In the List.xml file should be something like the following:
<CiscoIPPhoneImageList>
<ImageItem Image=”TN-logo1.png”
URL=”logo1.png”/>
<ImageItem Image=”TN-logo2.png”
URL=”logo2.png”/>
<ImageItem Image=”TN-logo3.png”
URL=”logo3.png”/>
</CiscoIPPhoneImageList>
For my images to work I had to put them in my root TFTP folder along with the List.xml in my tftp root. I think List.xml needs to be a CAPITAL L
Hopefully, the above should provide enough of a guide to get you set up and running on your Asterisk PBX
with your Cisco 7971 phone. Settings may differ for your own setup, but the main thing to remember is that
NAT must be off or nat=no in your Asterisk PBX extension settings to get the phone to register.
Trixbox gets a little tricky if you want to modify outbound call groups. Here is a trick to get an extension to dial out on a specific trunk, this will only work if you want that extension to dial out on that trunk and only that trunk so if that trunk is busy the extension will not dial out. Go into trixbox admin via the web panel and click on trunks on the right side you will see your trunks listed click on the trunk you want to dedicate to that extension, in the bottom of your browser you should see a url with something along the lines of extdisplay=OUT_2 that means trunk 2. Below is my macro pre dial hook notice I have ext 2004 and 2006 dialing out on trunk 3
extensions_custom.conf
[macro-dialout-trunk-predial-hook]
exten => s,1,goto(${AMPUSER},1)
exten => 2004,1,Set(DIAL_TRUNK=3)
exten => 2006,1,Set(DIAL_TRUNK=3)
If you are like me and have 1 line for international and 1 line for local I set a dial pattern to pick a trunk based on what is dialed, I also set it to grab line 2 first then line 1, in order to make this work you need to look in sip_additional and get the names of your trunks mine is [Line1] and [line2] you also need to change the context in your webadmin extension field for example I changed mine “from-internal” to custom-trunk-selector for each ext you want to use this rule.
extensions_custom.conf
[custom-trunk-selector]
exten => 911,1,dial(SIP/${EXTEN}@Line2,20) ; grab line 2 first for emergency
exten => 911,n,dial(SIP/${EXTEN}@Line1,20) ; dial out over 1 if line 2 is busy
exten => 611,1,dial(SIP/${EXTEN}@Line2,20) ; dial out on line 2
exten => 611,n,dial(SIP/${EXTEN}@Line1,20) ; dial out on line 1
exten => _1|NXXNXXXXXX,1,dial(SIP/${EXTEN}@Line2,20) ; dial out on line 2 for local calls
exten => _1|NXXNXXXXXX,n,dial(SIP/${EXTEN}@Line1,20) ; if 2 is busy dial out on line 1
exten => _NXXNXXXXXX,1,dial(SIP/${EXTEN}@Line2,20) ; same for local
exten => _NXXNXXXXXX,n,dial(SIP/${EXTEN}@Line1,20) ; same for local if 2 is busy
exten => _011.,1,dial(SIP/${EXTEN}@Line1,20) ; Dial international only on line 1 do not grab line 2 if line 1 is busy.
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HTC Animated Weather Theme for iPhone 4
HTC animated Weather Theme for iPhone 4 (208)
PBX Phone system Ring tone
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Download Modern Phone-Ringtone
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Here is my tutorial integrating a Toshiba-CTX 100 PBX with asterisk Tritxbox 2.8. Full VM integration including VM Message lighting and transferring from AA using an openvox 8 port analog card connected to an RSTU2 8 port toshiba analog station card. The only thing buggy is after logging into vm from a toshiba phone you have to wait for time-out the toshiba disconnect signal sent from the pbx does not hang up vm.
Install asterisk from the .iso image mine had a bug call recording (cdr) and the voicemail web interface would kick you out after you entered your username and pw. This is caused from memcached not starting on boot so in terminal run
Service memcached start
so memcahced starts on boot run: chkconfig memcached on. in terminal
Setup your normal VM settings in emanager setting your hunt group and analog vm station ports, If you don’t know how to do this then this is way to advance for you and do not continue. In SYSTEM VOICE MAIL DATA PROGRAM 579 I took out all the 91 and 92′s except 13: retrieve messages set this to 92. My VM hunt group is 850 with ext’s 401,402,403, and 404. I have only 4 ports out of the 8 in my VM hunt group I need the other 4 for transfering calls from asterisk to toshiba
=========================================================
Now lets move to the Trixbox Hacks to get Toshiba talking to the VM
We will not create any extensions in the trixbox web GUI
Under Trunks create your g0 for the ZAP channels this should be there by default.
Under outbound routes modify 0 9_outside add your zap trunk sequences and add 2. to the dial patterns so anytime we dial a 2XX toshiba extension from a SIP phone or IVR it is going to send the call out over the Openvox card to the rstu.
Now we need to tell asterisk to goto VM of the toshiba ext when rna forwards the call to the openvox card open up extensions_custom.conf and add these lines. Where _2xx is is my toshiba ext’s so if you use 4 digits like 2000 change it to _2XXX of if you use 4000 change to _4XXX
———————-Extensions_custom.conf————————————
[vm_from_toshiba]
; For VM integration with toshiba CTX 8 port FXS/FXO Card no T1 integration VM only, Extensions on Toshiba system are 200 + modify 2xx to your extensions for example extension 7000 would be 7xxx
;exten => s,1,Goto(vm_toshiba_set_mwi,_2XX,1) ; uncomment to text MWI I am no longer using this I created a new MWI.sh
exten => s,1,NoCDR()
exten => s,n,wait(.5)
exten => s,n,answer
exten => s,n,SET(TIMEOUT(absolute)=1800)
exten => s,n,SET(TIMEOUT(digit)=1.5)
exten => s,n,SET(TIMEOUT(response)=7)
exten => s,n,waitexten(3)
exten => s,n,VoicemailMain()
exten => s,n,hangup
exten => _X.,1,noop(Dialed extension is ${EXTEN})
;exten => _X.,n,Gotoif($["${EXTEN:-1}" = "D"]?terminated) ;;Checkout Toshiba seems to be sending a terminator ‘D’ There may have been a hangup before the whole dtmf string was transmitted.
;exten => _X.,n,System(/root/toshiba_error_finder ${EXTEN} &) ;;; This calls a little debugging script – comment it out
;exten => _X.,n,Voicemail(9999,u) ;;; There is a mistake on the toshiba forwarding string, let the caller tell me who they were trying to call – comment it out.
;exten => _X.,n,wait(1.5)
;exten => _X,n,Goto(test) ; Uncomment to Test MWI
exten => _X.,n,Voicemail(${EXTEN}) ; Goto Mailbox of toshiba ext RNA
;exten => _92X.,n,Voicemailmain(${EXTEN})
exten => _X.,n(terminated),NoCDR()
;exten => _X.,n,Voicemail(9999,u)
exten => _X.,n,hangup
;exten => _912XX,1,SET(TIMEOUT(absolute)=300) ;;; max msg 5 min – adjust as needed
;exten => _912XX,n,Goto(hs-vm-with-dtmf-detect,${EXTEN:2},1)
;exten => _912XX,n,Hangup()
;exten => _912XX,1,System(touch /tmp/vm/old_ext/${EXTEN:2}) ;;;; This is for debuging too
;exten => _912XX,n,Hangup()
exten => _8XX2XX,1,VoicemailMain(${EXTEN}) ; Goto Voice mail main when 850 is dialed from toshiba so you can check messages, emanager circular hunt group program 209 field 12
exten => _852XX,n,Hangup()
exten => _852XX,1,System(touch /tmp/vm/92/${EXTEN})
exten => _852XX,n,Hangup()
exten => _922XX,1,VoicemailMain(${EXTEN:2}|s)) ; Goto VoiceMail Main when message button is pressed 92 is in field of emanager program 579 field 12 puts 92 in fron of ext.
exten => _922XX,n,Hangup()
exten => _922XX,1,System(touch /tmp/vm/92/${EXTEN})
exten => _922XX,n,Hangup()
;exten => T,1,Hangup()
[Hz-vm-with-dtmf-detect]
exten => _2XX,1,System(asterisk -rx “originate Local/${CHANNEL:4:2}@hs-vm-meetme application voicemail \\”${EXTEN}|su\\”")
exten => _2XX,n,MeetMe(vmd${CHANNEL:4:2},AdFpqXx) ;Meetme mail boxes defined for zap channels 49-56
exten => _2XX,n,Hangup()
exten => _[ABD#],1,Hangup()
exten => h,1,MeetMeAdmin(vmd${CHANNEL:4:2},K) ;Kick all users out of conference
[hs-vm-meetme]
exten => _X.,1,NoCDR()
exten => _X.,n,Wait(.3) ; give hs-vm-with-dtmf-detect time to create meetme
exten => _X.,n,MeetMe(vmd${EXTEN},q)
exten => _X.,n,Hangup()
—————–END extensions_custom.conf————————————
Now we need to tell Trixbox to use the vm_from_toshiba open up chan_dahdi.conf and uncomment the line include chan_dahdi_additional.conf
open up chan_dahdi_additional change context= to vm_from_toshiba
context=vm_from_toshiba
restart asterisk
now run asterisk -r and put in dahdi show channels all contexts should read vm_from_toshiba.
Now if you dial your vm port from a toshiba phone it should ring the openvox card and you should see it in CLI.
Now we need to create the mailbox since we can not create the same toshiba ext in the trixbox gui because if we did that we would not be able to dial toshiba phones from the sip side. open up voicemail.conf and add the lines to create mailbox’s
Voicemail.conf (add to end)
200 => 200,Receptionsist,xxx@foo.com,,attach=yes|saycid=yes|envelope=yes|delete=no
205 => 205,Shipping,xxxfoo.com,,attach=yes|saycid=yes|envelope=yes|delete=no
210 => 210,Billing,xxx@foo.com,,,attach=no|saycid=no|envelope=no|delete=no
=======================================================
No we got VM answering let’s get the message light working goto /usr/bin and create a file mwi.sh
—————–MWI.SH———————
#!/bin/bash
VM_EXTEN=$2
MESSAGES=$3
if [ "${MESSAGES}" -gt 50 ]; then
echo “Not an Avaya phone on extension ${VM_EXTEN} – exiting” >> /var/log/asterisk/mwi
exit
fi
if [ "$MESSAGES" -gt 0 ]; then
echo “Turning on mwi on ${VM_EXTEN}” >> /var/log/asterisk/mwi
echo “Channel: DAHDI/1-3/#63${VM_EXTEN}” > $MAILBOX.call
else
echo “Turning off mwi on ${VM_EXTEN}” >> /var/log/asterisk/mwi
echo “Channel: DAHDI/1-3/#64${VM_EXTEN}” > $MAILBOX.call
fi
echo “MaxRetries: 5″ >> $MAILBOX.call
echo “RetryTime: 15″ >> $MAILBOX.call
echo “WaitTime: 30″ >> $MAILBOX.call
echo “Context: custom-vmnotify” >> $MAILBOX.call
echo “Extension: s” >> $MAILBOX.call
echo “Priority: 1″ >> $MAILBOX.call
echo “Archive: yes” >> $MAILBOX.call
chown asterisk.asterisk $MAILBOX.call
mv -f $MAILBOX.call /var/spool/asterisk/outgoing/
———————-END MWI.SH————————-
Now make sure asterisk can execute the file from terminal run
chown asterisk:asterisk /usr/bin/msi.sh
Now we need to call the script in asterisk open up vm_general.conf and add
externnotify=/usr/bin/mwi.sh
Now dial a toshiba ext from a toshiba phone it should ring goto VM light should light and you should be able to press the message button it should say enter your mailbox number.
I don’t know why this works but if I dial 200 from a sip phone I get dial tone form the toshiba then I can dial a toshiba ext and intercom/transfer works from sip to toshiba.
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