Connecting Gizmo to Asterisk

As I explained briefly here, I've set myself up with the SIP URI sip:peter@rukavina.net. As a result, users of SIP clients like Gizmo Project can call me by entering that as my "phone number." Here's what I needed to do to set that up:

• Asterisk set up and operationg my server.
• Set up an SIP SRV record on the rukavina.net DNS server that points to the server that Asterisk is running on. The record looks like this:

_sip._udp       IN      SRV     1       0       5060    sip.rukavina.net.

• Tested the SRV record:

# host -v -t srv _sip._udp.rukavina.net
Trying "_sip._udp.rukavina.net"
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 55718
;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 2, ADDITIONAL: 3

;; QUESTION SECTION:
;_sip._udp.rukavina.net.                IN      SRV

;; ANSWER SECTION:
_sip._udp.rukavina.net. 3600    IN      SRV     1 0 5060 sip.rukavina.net.

;; AUTHORITY SECTION:
rukavina.net.           3600    IN      NS      dns.reinvented.net.
rukavina.net.           3600    IN      NS      dns.ypi.com.

;; ADDITIONAL SECTION:
sip.rukavina.net.       3600    IN      A       198.167.161.49
dns.ypi.com.            3600    IN      A       69.28.218.52
dns.reinvented.net.     3600    IN      A       198.167.161.50

• In the Asterisk sip.conf I have set the default context to default:

[general]
context = default

• In the Asterisk extensions.conf I have an extension, in the default context, named peter that rings my phone (the local SIP extension 33):

[default]
exten => peter,1,Dial(SIP/33)

And that's it: after reloading the Asterisk configuration and restarting my DNS server, SIP clients dialing the URI sip:peter@rukavina.net get routed directly through to the extension on my desk.