System information

Typing the following command returns a listing of all the peers that Asterisk knows
about (regardless of their state):
*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
0000FFFF0001/0000FFFF0001 192.168.1.100 D N 5060 Unmonitored
0000FFFF0002/0000FFFF0002 192.168.1.101 D N 5060 Unmonitored
You may notice that the Name/username field does not always show the
full name of the device. This is because this field is limited to 25
characters.
Note that the Status in our example is set to Unmonitored. This is because we are not
using the qualify=yes option in our sip.conf file.
Analog Phones
There are two popular methods for connecting analog phones to Asterisk. The first is
by using an ATA that most commonly connects to Asterisk using the SIP protocol. The
Asterisk configuration for an ATA is the same as it would be for any other SIP-based
handset. The other method is to directly connect the phones to the Asterisk server using
telephony hardware from a vendor such as Digium. Digium sells telephony cards that
can be added to your server to provide FXS ports for connecting analog phones (or fax
machines). For the purposes of demonstrating the configuration, we’re going to show
the configuration required if you had a Digium AEX440E card, which is an AEX410
half-length PCI-Express with four FXS modules and hardware-based echo cancellation.
Regardless of which hardware you are using, consult your vendor’s
documentation for any hardware-specific configuration requirements.
First, ensure that both Asterisk and DAHDI are installed (refer back to “How to Install
It” on page 48 for instructions). Note that DAHDI must be installed before you install
Asterisk. When you install DAHDI, be sure to install the init script as well. This will
ensure that your hardware is properly initialized when the system boots up. The init
script is installed from the DAHDI-tools package.
100 | Chapter 5: User Device Configuration