If you are getting a vtiger timeout make sure you are using mod_php instead of mod_fcgid

I spend a few days trying to install vtiger 5.30 and the install kept timing out with internal server error 500 while creating the mysql database.

The error I was getting was:

(104)Connection reset by peer: mod_fcgid: error reading data from FastCGI server, referer:

Premature end of script headers: install.php, referer: install.php

Change server setting to use m0d_php instead to mod_fcid

Peer Details:

username=??????
secret=??????
type=friend
qualify=no
port=5070
nat=yes
insecure=very
host=4.55.44.16
allow=all
context=from-trunk
dtmfmode=auto
canreinvite=yes

User Details:

type=peer
dtmfmode=auto
context=from-pstn
host=4.55.44.16

SSH into you box

1.

First install sox

[root@elastix ~]# yum install sox

2.

You need to install mpg123. You can get the wget command and download it from this site.

http://pkgs.repoforge.org/mpg123/

I’m running a 64bit version so I downloaded x86_64

[root@elastix ~]#wget http://pkgs.repoforge.org/mpg123/mpg123-1.13.3-1.el5.rf.x86_64.rpm

[root@elastix ~]#rpm -i mpg123-1.13.3-1.el5.rf.x86_64.rpm

3.

Here is the trick, once you have uploaded your files and the conversion is successful you need to go into the mohmp3 directory and delete all files that have *.gsm file extension.

Login into you phone system with SSH or secure ftp. I prefer secure ftp because its easier to create, delete directories and files.

delete every file with a .gsm file extension in /var/lib/asterisk/mohmp3/

Its pretty stupid but your newly converted files will not play if those files are in the directory.

1.

First backup your Trixbox

Trixbox 2.6.x left out the freepbx backup and restore module which I thought was better then the Trixbox backup. You will have to go to the module admin and upload it

First down load the module from freepbx

http://www.freepbx.org/trac/browser/modules/release/2.5/backup-2.5.0.2.tgz

Go to module admin / upload module/ browse for the module you just downloaded / manage modules locally / scroll down to the backup & restore module / click on install / process / apply changes

Now go to tools / backup & restore  and do a complete backup of your phone system.

The backup file is store in /var/lib/asterisk/backups.

You will need a ftp utility to get the file off the server. I use coretfp, its free and easy to use.

You should also backup the tftpboot folder is you want to save your phone config files.

2.

Now that you have a backup of your Trixbox system login into your newly installed Elastix system. We first need to copy over the backup file to Elastix

Since Elastix has not backed anything up yet a backups folder does not exist on the system. You will need to create this folder in SSH or some sFTP client

SSH

[root@elastix]# cd /var/lib/asterisk

[root@elastix]# mkdir backups

Now copy your Trixbox backup to

/var/lib/asterisk/backups

3.

Log into Elastix and click on PBX. Click on unembedded freePBX. This should open another windows. Login and now you should be in unembedded freePBX. Click on tools / backup & restore. On the right hand side click on restore backup.

You should now see the trixbox backup file. Click on it and do a Restore Full Backup Set. Is a few seconds you should see that ugly green Trixbox version of freePBX. Do not click the orange apply bar yet

Clicking the orange apply bar at this point will not do anything. The problem here is that Elastix and Trixbox use two different Asterisk Manager passwords. Elastix uses elastix456 and trixbox used amp111.

SSH into your box to go to

[root@elastix]# cd /etc/asterisk

[root@elastix]# nano manager.conf

Config should look like this.

;
; AMI – Asterisk Manager interface
;
; FreePBX needs this to be enabled. Note that if you enable it on a different IP, you need
; to assure that this can’t be reached from un-authorized hosts with the ACL settings (permit/deny).
; Also, remember to configure non-default port or IP-addresses in amportal.conf.
;
; The AMI connection is used both by the portal and the operator’s panel in FreePBX.
;
; FreePBX assumes an AMI connection to localhost:5038 by default.
;
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0

[admin]
secret = elastix456
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
permit=192.168.1.99/255.255.255.0
read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate

#include manager_additional.conf
#include manager_custom.conf

Know that we have the Asterisk manager password fixed we need t update freePBX to a version that is compatible with Elastix. The problem with Trixbox is when you go into freepbx module admin and click check online for updates the system try’s to call update.trixbox.com instead of the real freePBX update servers. You need to either update all the links in the file /var/www/html/admin/functions.inc.php or just down a copy of mine that is already fixed.

http://www.fintechcommunications.com/trixbox-to-elastix-fix.zip

The only major issue and key to this upgrade is the file above

Unzip and overwrite the file that is aleady in /var/www/html/admin

When you do updates now it will call freePBX update servers instead of trixbox.

The next major step is to update freepbx to version 2.6. On the module admin click on check for updates. Download and intall 2.6 upgrade tool.

Now click on upgrade tool (upper left corner) click on the 2.6 upgrade tool.

Do as the 2.6 upgrade instruction says. Click upgrade.

Go back to module admin and click check for updates.

Download freepbx framework and upgrade and install.

Yeah, Your back to freepbx colors again

From here its just a matter of updating modules and version until you end up with the freePBX version you want.

<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>M/D/Ya</dateTemplate>
<timeZone>Pacific Standard/Daylight Time</timeZone>
</dateTimeSetting>
<callManagerGroup>
<name>5.0 Beta</name>
<tftpDefault>true</tftpDefault>
<members>
<member priority=”0″>
<callManager>
<name>192.168.1.4</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>192.168.1.4</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo uuid=”{cd241e11-4a58-4d3d-9661-f06c912a18a3}”>
<name>Disable</name>
<srstOption>Disable</srstOption>
<userModifiable>false</userModifiable>
<ipAddr1>192.168.1.4</ipAddr1>
<port1>2000</port1>
<ipAddr2></ipAddr2>
<port2>2000</port2>
<ipAddr3></ipAddr3>
<port3>2000</port3>
<sipIpAddr1>192.168.1.4</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>true</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>Justin</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>3015</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>3015</name>
<displayName>3015</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>3015</authName>
<authPassword>53124AA53124</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>3015</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button=”2″>
<featureID>9</featureID>
<featureLabel>Fintech Voicemail</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>3050</name>
<displayName>3050</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>3050</authName>
<authPassword>53124AA53124</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>3050</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button=”3″>
<featureID>9</featureID>
<featureLabel>E-Recycler Voicemail</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>3060</name>
<displayName>3060</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>3060</authName>
<authPassword>53124AA53124</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>3060</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button=”6″>
<featureID>2</featureID>
<featureLabel>Conf</featureLabel>
<speedDialNumber>825</speedDialNumber>
</line>
<line button=”7″>
<featureID>2</featureID>
<featureLabel>Page</featureLabel>
<speedDialNumber>850</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>sip8961.9-1-1SR1</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>6,7</daysDisplayNotActive>
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>18:00</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>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
<version>5.0(2)</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL></authenticationURL>
<directoryURL>http://192.168.1.4/xmlservices/directory.xml</directoryURL>
<idleURL></idleURL>
<informationURL>http://192.168.1.4/ccmcip/GetTelecasterHelpText.jsp</informationURL>
<messagesURL></messagesURL>
<proxyServerURL>192.168.1.4</proxyServerURL>
<servicesURL>http://192.168.1.4/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>192.168.1.4</processNodeName>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
</device>

ssh-keygen -t rsa

# hit return three times

This copies the newly generated key to the backup server

ssh-copy-id -i ~/.ssh/id_rsa.pub root@192.168.3.253

# enter your password for username on remote_host

This will be the cron commands that we run each week. This will backup freepbx and the phone config files

rsync -ave ssh /var/lib/asterisk/backups/ 192.168.3.253:/var/lib/asterisk/backups/

rsync -ave ssh /tftpboot/ 192.168.3.253:/tftpboot/

for Elastix run

rsync -ave ssh /var/www/backup/ 192.168.3.253:/var/www/backup/

The first step is to create a SIP trunk with TCP support. My favorite distro is Elastix. Add the following to your sip_general_custom.conf file

elastix*CLI> nano /etc/asterisk/sip_general_custom.conf

Add the following:

tcpenable=yes
tcpbindaddr=0.0.0.0

Save your work cntrl-shift-O

Go back into elastix or freepbx

create a sip trunk with the following:

host=192.168.1.7 (IP address of Lync)
type=friend
insecure=very
transport=tcp
port=5060
context=from-internal


create and outbound route to lync. In my case 30XX for extension 3015

This is all you have to do on the Asterisk Side

On the Lync side go into your topology editor and add the IP address of your asterisk server to PSTN gateway. Do forget to add the PSTN gateway to you mediation server.

192.168.1.8 = Vicidial

192.168.1.15 = Elastix

68.68.124.55 = Sip trunk provider

You not have to create a dial plan in Lync

Under the default dial plan

I'm dialing extension 205 so add the following normalization rull ^(2\d{2})$ this should allow me to dial 2XX

Next under voice policy

Next under route

Under PSTN usuage

That should be it. You should be able to dial between systems.