FreePBX FOP1 to FOP2

How to change FOP1 to FOP2 on Elastix

1.  Download and install FOP2

cd /usr/src
wget http://www.fop2.com/file.php?file=20
tar -zxvf fop2-2.22-centos5-i386.tgz 
cd fop2
make && make install

2. Add fop2 user to Asterisk Manager.

Edit /etc/asterisk/manager.conf

[fop2]
secret = psfop2
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
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

3. Edit /usr/localfop2/fop2.cfg and change the following lines.

manager_user=fop2
manager_secret=psfop2

4.Edit /etc/amportal.conf and edit the uncomment the line starting FOPRUN and change it as following.

FOPRUN=false

5. Edit /var/www/html/panel/op_server.cfg and change listen_port.

listen_port=4444

6. Edit /etc/httpd/conf/httpd.conf and add the following line under  Aliases:

alias /panel /var/www/html/fop2

7. Add FreePBX index file and change the fop2 ownership asterisk.

cp /var/www/html/fop2/index.html /var/www/html/fop2/index_amp.php
chown -R asterisk.asterisk /var/www/html/fop2

8. Restart httpd,amportal and fop2 services.

/etc/init.d/httpd restart
chkconfig fop2 on
service fop2 start
/usr/sbin/amportal restart

FreePBX Resolve Errors

The below error occurs when you define a FAX extension and forget define a voice extension in inbound routes.

PHP Fatal error:  Call to a member function output() on a non-object in /var/www/html/admin/extensions.class.php on line 303
1

Cisco SIP Configuration

Cisco 7911G/7945/7962 Phone with Asterisk

Download the firmware (7911 , 7945 , 7962) and extract it.

Download and install/extract the tftp server software.

Open the tftp server software and make the SIP firmware  extracted directory as the root directory of the tftp server.

Goto command prompt(Start>Run>CMD and press enter) and enter the following command.

C:\Users\user>tftp <tftp-server-ip-address> get dialplan.xml

You should get the message starting “Transfer Successful”.(If your OS is Win7/Vista you have to install tftp client from the Add/Remove Programs)

C:\Users\shyju>tftp 192.168.20.124 get dialplan.xml
Transfer successful: 258 bytes in 1 second(s), 258 bytes/s

Open your dhcp server configuration and add  TFTP server IP address as the boot server in DHCP scope Options. Refer this article to configure DHCP Options.

Rename the  with SEP<MAC-ADDRESS-OF-YOUR-PHONE>.cnf.xml. Then open that file and change the following lines to match with your IP PBX details.

<processNodeName>
<featureLabel>
<proxy>
<port>
<name>
<displayName>
<authName>
<authPassword>

Edit your Asterisk SIP configuration and add nat = no below the user context.

This step is important otherwise the phones will not register and on the phone’s display you can see the message Registering..

If you are using FreePBX the file will be /etc/asterisk/sip_additional.conf, In the case of Asterisk-GUI file is /etc/asterisk/users.conf

[610]
deny=0.0.0.0/0.0.0.0
type=friend
secret=jbsdf7h4ks
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=no
mailbox=610@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/610
context=from-internal
canreinvite=no
callgroup=
callerid=device <610>
allow=all
accountcode=
call-limit=50