//Debuging
*CLI> sip show peers
*CLI> sip show users
sip show subscriptions
Install TFTP Server
- Make the Linux server into a TFTP server Download here http://minded.ca/wp-content/uploads/2010/06/tftpmanager-1.1.0.0.tgz
http://minded.ca/wp-content/uploads/2010/06/tftpmanager-1.1.0.0.tgz
- Go to Admin Module Administration Upload Module Upload the file (in tgz format) Then you will have to click on it and then click install Then click process Then click apply at the top in red It’s installed.
- Install the module Module Administration System Administration -> TFTP Manager
Convert 7941/7960G to SIP + Register extension
note: SIP phones expect Option 66 for the TFTP server. The 7960 phone also requires option 150
Method 1)
1.) Install the tftp server role to the Astrix (Instructions above)
2. Download the COP file from cisco (7960)
cisco (7941)
->Extract in /tftpboot on asterisk server (Wincp scp Connection)
3. Download firmware to boot the phone
-> Extract in /tftpboot on asterisk server (Wincp scp Connection)
4. Download config to boot the phone
-> Extract in /tftpboot on asterisk server (Wincp scp Connection)
5. Create the extension, username and password on Asterisk web page
6. Upload the username and password for the phone configuration file to /tftpboot
7. Upload Adding a new line for the model phone in the Default XML File
8. Add the extension in AsteriskNOW
—————————————————
Adding a phone + Extension.
1. Applications -> Extensions -> Add new -> CHAN_SIP
NOTE: Make sure the secret for the phone is blank!!!
—Restart the Asterisk service
service asterisk restart
–Adding a 794X Cisco IP phone
a) Change the port for SIP to 5060. NOTE: DO NOT SET JPSIP
Settings -> Asterisk SIP settings ->Chan SIP settings -> 5060
///FBX -> Settings -> Asterisk SIP Settings -> Chan SIP Settings -> Port
1. Applications -> Extensions -> Add new -> CHAN_SIP ! NOTE NOT PJSIP
2. Enable anonymous Phones
Restart the machine.
uses XML configuration files. NOTE “USECALLMANAGER” IS NOT TO BE REPLACED with IP under the LINE
Add the followng SIP firmware to /tftpboot via wincp
—-Debugging a SIP connection
cd /etc/asterisk
then asterisk -vvvvvvvrd
When you’re in the CLI> menu type:
sip set debug
(note: on newer versions of Asterisk, it is sip set debug {on|off|ip|peer} )
Or on the GUI
Reports-> Asterisk Log files
—Accessing Voicemail
/var/spool/asterisk/voicemail/context/boxnumber/INBOX/
Call *97
Enter PIN
Direct dial can be used to check VM
*989999 !check VM for extension 9999
—Shared extension voicemail
//Definitely worked for each extension
- Update the Extension -> Advanced -> mailbox to be “9999@default”
- Update the Extension -> Followme -> Enable
- Update the Extension -> Followme -> Followme list “9999”
//Dunno if this shit worked
Enable TCP on Asterisk, make default
Reboot
Enable root hints
Use the attached xml configuration file
Update the check time to 1 seconds
Extension 9999 -> Transport TCP primary
—Single dial login/logout of dynamic queue.
*45 login/logoff
checking the status
/etc/asterisk -rd
queue show 9000
Update the xml file for the second line to be speed dial to *45
<line button="2">
<featureID>21</featureID>
<featureLabel>Employee A</featureLabel>
<speedDialNumber>761</speedDialNumber>
</line>
—-Queue
Error: Dynamic agent phone only rings once
Applications Queues -> Queue -> Timing and Agent options -> Agent timeout 16 seconds. Note 15 is bugged.
————————-IVR.
1. Create your IVR message.
Admin -> System recordings -> New
Link to feature code -> Yes
2. Assign the IVR to the queue.
Applications -> Queues -> 9000 -> Join Announcement -> IVR
3. Dial the feature code found in step 1 to set the IVR
*297 For me
————————-Call Overflow
1. Create 2 queues.
-> Applications -> Queues -> Add “Servicedesk”
-> Applications -> Queues -> Add “Servicedesk-CallOverflow”
2. Configure the first queue
Applications -> Queues -> Failover destination -> Queue “Servicedesk-CallOverflow”
-> Add your dynamic overflow phones
3. Configure Voicemail final destination for Servicedesk-CallOverflow
Applications -> Queues -> Failover destination -> VM -> 9999
4.If the queue is empty send directly to voicemail
Applications -> Queues -> Capacity Options -> Join Empty = “No”
—Update DNS address
/etc/resolv.
conf
nameserver 8.8.8.8
—–change IP address
1. login as a root
2. cd /etc/sysconfig/network-scripts
3. Open the file ifcgf-eth0 (or ifcgf-eth1 if you use the second card etc) using your favorite editor. For example vi, or nano.
4. Modify the file , replacing BOOTPROTO=dhcp with BOOTPROTO=none, and adding :
NETMASK=255.255.255.0
IPADDR=192.168.0.10
GATEWAY=192.168.0.1
obviously, replace the ip numbers with yours.
5. save the changes
6. restart network service: service network restart
— Asternic Call Stats
1. Mysql password is “” (Blank)
2. Only version 2.1.0 is installing
3. winscp File to server
4. tar zxvf asternic-stats-pro-2.1.0.tgz
5. cd asternic-stats-pro-2.1.0
6. Make install (Note: an active Internet connection is required to download reqs)
7. set the password for admin
nano /etc/asterisk/manager.conf
[admin]
secret=admin
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read=all
write=all
8. nano /var/www/html/stats/config.phpfile:
// Manager details (for realtime tab)
$MANAGER_HOST = “127.0.0.1”;
$MANAGER_USER = “admin”;
$MANAGER_SECRET = “admin”;
9. reboot
login and license
192.168.0.120/stats
admin/admin
—Asternic Reporting
Change the SLA time
Setup -> Preferences -> Page 2 -> SLA Interval
—Asternic don’t count IVR
Report Designer – > Wait time (x5ish) = TRANSFERRED_WAIT_TIME
—Asternic Realtime stats
/etc/asterisk/manager.conf
Add the following lines
[admin2]
secret=amp111
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read=all
write=all
/var/www/html/stats/config.php
$MANAGER_HOST = “127.0.0.1”;
$MANAGER_USER = “admin2”;
$MANAGER_SECRET = “amp111”;
Restart both asternic + asterisk server
—-VOIPLINE POC
You can use the below example to configure your SIP trunk on a Asterisk, FreePBX, Elastix and other similar systems:
1. Drag the SIP trunk object into your mPBX call flow.
2. Select mode 3 (Create SIP registration).
3. Click save and apply the configuration.
Sip server: check SIP Server address shown in the properties of SIP Trunk object (don’t forget about port number too)
User: Use SIP username shown in the SIP Trunk object properties
Pass: use SIP Password generated and shown in the SIP Trunk object properties
Configuration details:
Trunk name: SIP Username
PEER Details:
username=[SIP Username]
type=peer
secret=[YOURPASS]
qualify=yes
nat=yes
host=sipm.voipline.net.au
port=7060
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=g729
allow=ulaw
insecure=port,invite
Register String:
register => sipusername:sippassword@sipm.voipline.net.au:7060
—-Phone only rings, when you answer no audio is heard.
1. Convert everything to alaw.
2. On both sip trunk (Asterisk) + VOIPLINE
only allow codec alaw.
3. Convert phones to alaw
—Ubuntu build
//Static IP
auto eth0
iface eth0 inet static
address 192.168.1.88
netmask 255.255.255.0
network 192.168.1.0
broadcast 192.168.1.255
gateway 192.168.1.1
dns-nameserver 8.8.8.8
///Update librarys
sudo apt-get update
sudo apt-get upgrade
Download Asterisk dependencies
sudo apt-get install build-essential wget libssl-dev libncurses5-dev libnewt-dev libxml2-dev linux-headers-$(uname -r) libsqlite3-dev uuid-dev git subversion
Download Asterisk
sudo wget downloads.asterisk.org/pub/telephony/asterisk/asterisk-15-current.tar.gz
sudo tar zxvf asterisk-15-current.tar.gz
cd /tmp/asterisk-15
sudo sudocontrib/scripts/install_prereq install
sudo ./configure
sudo make menuselect
sudo make
sudo make install
sudo make samples
sudo make config
sudo service asterisk start
//////////////////////install the webgui
cd /usr/src
sudo svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui
cd /usr/src/asterisk-gui
sudo
./configure
sudo make
sudo make install
nano /etc/asterisk/manager.conf
[general]
enabled = yes
webenabled = yes
read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,d$
write = system,call,agent,user,config,command,reporting,originate,mes$
nano /etc/asterisk/http.conf
enabled=yes
bindaddr=0.0.0.0
bindport=80
enablestatic=yes
cd /usr/src/asterisk-gui
sudo
make checkconfig
Add the following line
sudo nano /etc/asterisk/guipreferences.conf
config_upgraded = yes
Set the permissions
sudo chown -R calemblake:calemblake /etc/asterisk/ /var/lib/asterisk /usr/share/asterisk# if asterisk runs as the user "asterisk"
sudo chmod 644 /etc/asterisk/*
sudo service asterisk start
//Website
http://192.168.1.96/static/config/index.html
http://192.168.1.96/static/config/cfgbasic.html
—–Australian PhoneCompany SIP Trunk