System information

or from the Asterisk CLI:
*CLI> dialplan reload
You should now be able to dial between your two new extensions. Open up the CLI
in order to see the call progression. You should see something like this (and the set you
are calling should ring):
-- Executing [100@LocalSets:1] Dial("SIP/0000FFFF0001-0000000c",
"SIP/0000FFFF0001") in new stack
-- Called 0000FFFF0001
-- SIP/0000FFFF0001-0000000d is ringing
If this does not happen, you are going to need to review your configuration and ensure
you have not made any typos.
Under the Hood: Your First Call
In order to get you thinking about what is happening under the hood, we’re going to
briefly cover some of what is actually happening with the SIP protocol when two sets
on the same Asterisk system call each other.
Asterisk as a B2BUA
Bear in mind that there are actually two calls going on here: one from
the originating set to Asterisk, and another from Asterisk to the desti-
nation set. SIP is a peer-to-peer protocol, and from the perspective of
the protocol there are two calls happening. The SIP protocol is not aware
that Asterisk is bridging the calls; each set understands its connection
to Asterisk, with no real knowledge of the set on the other side. It is for
this reason that Asterisk is often referred to as a B2BUA (Back to Back
User Agent). This is also why it is so easy to bridge different protocols
together using Asterisk.
For the call you just made, the dialogs shown in Figure 5-2 will have taken place.
For more details on how SIP messaging works, please refer to Appendix B and the SIP
RFC at http://www.ietf.org/rfc/rfc3261.txt.
104 | Chapter 5: User Device Configuration