This guide will demonstrate the process of configuring a Linksys PAP2 ATA with Asterisk to make calls between two regular phones. Operating system used in this tutorial is Debian GNU / Linux (etch), kernel version 2.6.18-5-486 was used.
Environment used
- ATA Linksys PAP2;
- 2 analog phones;
- A hub;
Requirements
- Asterisk installed and configured correctly
Installing DHCP
First you must take care not to put two DHCP servers on the same network, it can be used for a hub to link the local DHCP and ATA. To install the dhcp server will use the APT package manager, as illustrated in the following command.
# Apt-get install dhcp3-server
DHCP Configuration
To configure the DHCP service is needed to edit / etc/dhcp3/dhcpd.conf.
option domain-name "voffice.com.br";
option domain-name-servers 192.168.1.1;
option subnet-mask 255.255.255.0;
default-lease-time 600;
max-lease-time 7200;
server-name "VOffice";
subnet 192.168.1.0 netmask 255.255.255.0 (
range 192.168.1.100 192.168.1.150;
ATA host (
hardware ethernet 00:12:17: FC: 36: AB;
fixed-address 192.168.1.135;
)
)
It can be observed in the above configuration, the IP address of the Linksys ATA has been configured according to their MAC address.
Configuring the Linksys PAP2 ATA
To configure the two lines of the Linksys ATA is necessary to access it using any browser with an IP address set to the same.
http://IP-DO-ATA/admin/advanced
LINE 1
- Access the menu option "Line 1"
- Line Enable: yes
- Proxy: 192.168.1.1
- Register: yes
- Display name: 1000
- Password: 1000
- User ID: 1000
LINE 2
- Access the menu option "Line 2"
- Line Enable: yes
- Proxy: 192.168.1.1
- Register: yes
- Display name: 1001
- Password: 1001
- User ID: 1001
The above options may vary according to the needs of each. Finished the setting, click Save Settings.
Adding the extensions on Asterisk
To add the extensions in Asterisk you must edit the file / etc / asterisk / sip.conf and add the configuration shown below.
# Vi / etc / asterisk / sip.conf
[1000]
type = friend
username = 1000
callerid = 1000
secret = 1000
host = dynamic
context = test
[1001]
type = friend
username = 1001
callerid = 1001
secret = 1001
host = dynamic
context = test
Adding a context in Asterisk
Now you must create the context test, the phones can make calls, so you must edit / etc / asterisk / extenions.conf.
# Vi / etc / asterisk / extensions.conf
[Test]
exten => _XXXX, 1, Dial (SIP / $ (Extensible))
exten => _XXXX, Hangup
Final Thoughts
After carrying out all the settings correctly you must start the asterisk, as illustrated below.
# Asterisk-r
VOffice * CLI> sip reload
VOffice * CLI> extensions reload
VOffice * CLI> sip show peers
Name / username Host Dyn Nat ACL Port Status
1001/1001 192.168.1.135 D 5060 Unmonitored
1000/1000 192.168.1.135 D 5060 Unmonitored
Now test calls can be performed from one analogue phone to other one.
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