Page 131 - Asterisk™: The Future of Telephony

type=friend
secret=welcome
context=osaka_incoming
host=dynamic
disallow=all
allow=ulaw
And the configuration for the Osaka box:
[general]
register => osaka:welcome@192.168.2.202/toronto

[toronto]
type=friend
secret=welcome
context=toronto_incoming
host=dynamic
disallow=all
allow=ulaw
Many of the previous options may be familiar to you by now, but let’s take a look at
them further just in case they are not.

The second line of the file tells our Asterisk box to register to the other box, with the
purpose of telling the remote Asterisk box where to send calls when it wishes to send
a call to our local Asterisk box. Remember how we mentioned a little oddity in the
configuration? Notice that at the end of the registration line we tag on a forward slash
and the username of the remote Asterisk box? What this does is tell the remote Asterisk
box what digest name to use when it wants to set up a call. If you forget to add this,
then when the far end tries to send you a call, you’ll see the following at your Asterisk
CLI:
[Apr 22 18:52:32] WARNING[23631]: chan_sip.c:8117 check_auth: username mismatch,
have <toronto>, digest has <s>
So by adding the forward slash and username, we tell the other end what to place in
the Digest username of the Proxy Authorization field in the SIP INVITE message.
The rest of the file is the authorization block we use to control the incoming and out-
going calls from the other Asterisk box. On the Toronto box, we have the [osaka]
authorization block, and on the Osaka box, we have the [toronto] block. We define
the type as a friend, which allows us to both receive and place calls from the other
Asterisk box. The secret is the password the other system should use when authenti-
cating. The context is where incoming calls are processed in the dialplan
(extensions.conf). We set the host parameter to dynamic, which tells our Asterisk box
that the other endpoint will register to us, thereby telling us what IP address to set up
calls when we want to send a call to the other end. Finally, the disallow and allow
parameters control the codecs we wish to use with the other end.
If you save the file and reload the SIP channel on both Asterisk boxes (sip reload from
the Asterisk console), you should see something like the following, which will tell you
the remote box successfully registered:

Connecting Two Asterisk Boxes Together via SIP | 103