82 | Section 5
Turn some services off for now
At the shell prompt, type ‘disable-iptables’ and then <enter>, then type ‘disable-fail2ban’ and
hit <enter>. These are security tools that could get in your way until you’re fully set up. If you
run your machine behind a router or firewall, you won’t need these anyway.
3 Congure Asterisk
Asterisk needs to be configured for your application. There is plenty of information on this
topic on the Asterisk site. In brief, the steps are:
1. Set up trunks.
2. Set up outbound routes.
We’ll work through a simple example in the following pages.
We prefer to use SIP extensions rather than SIP trunks to connect Asterisk lines to the
VX because they are more flexible. For example, station SIP signaling conveys caller ID,
which the VX can display and use. SIP trunks, on the other hand, do not pass CID. The
VX supports the required SIP registration when you enter the authentication password
on the show configuration pages.
By default, Asterisk uses SIP Reinvites. This causes the VX or VoIP phone to make a direct
connection to the Telco, bypassing Asterisk’s processing when compatible codecs are present
at both ends. This is generally a good idea, but it can be troublesome because it forces the
VX to change the IP address and other parameters for the line source. While Reinvites are
supported in VX since April 2011, we have discovered an issue with Asterisk which can cause
audio problems in some cases. If Asterisk is used as a PSTN gateway or connected to a telco,
it should be fine, but if it is accessible from the public internet (ie. anyone with a softphone
can call it), it is better to disable Reinvites. If one is not sure, better to be on the safe side. Until
re-invites are supported, the Asterisk ‘canreinvite’ parameter should be set to ‘no’ (in later
versions of Asterisk, those are two options: ‘directmedia’ and ‘directrtpsetup’)
(In case anybody is wondering - Asterisk doesn’t re-negotiate the codecs. For example: VX
supports g.711 and g.722. Let’s say someone is calling from a softphone supporting g.711
and Speex. Asterisk supports all four, and will advertise it this way, transcoding if necessary.
However, when it makes a direct connection, it doesn’t change the codecs to those actually
supported by each party. Thus, VX will end up thinking that the other end supports g.722, and
the softphone will think that VX supports Speex - resulting in silence in both ways.)
When Asterisk and VX are on the same subnet, Asterisk’s NAT (Network Address Translation)
support should be set to ‘no’.
It remains to be seen if, and how well, Asterisk will support wideband codecs. Until this is
clarified, and when the VX supports wideband codecs, it might be required to keep Asterisk out
of a wideband call path.
Let’s get started…
Log in to the web GUI
Point your browser to your Asterisk server address, being sure that your computer is on
the same subnet. For example, if your Asterisk box is on 192.168.0.248 with a netmask of
255.255.255.0 and your browser computer is on 192.168.0.11 or any IP with the same first three
bytes 192.168.0 and the last byte between 2 and 254, accompanied by the same netmask as the
Asterisk box’s, you are good. Supply the login credentials and you should see the screen below:
To Next Page
Loading...