Connecting Gizmo to Asterisk

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As I explained briefly here, I've set myself up with the SIP URI 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 DNS server that points to the server that Asterisk is running on. The record looks like this:
_sip._udp       IN      SRV     1       0       5060
  • Tested the SRV record:
# host -v -t srv
Trying ""
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 55718
;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 2, ADDITIONAL: 3

;                IN      SRV

;; ANSWER SECTION: 3600    IN      SRV     1 0 5060

;; AUTHORITY SECTION:           3600    IN      NS           3600    IN      NS

;; ADDITIONAL SECTION:       3600    IN      A            3600    IN      A     3600    IN      A
  • In the Asterisk sip.conf I have set the default context to default:
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):
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 get routed directly through to the extension on my desk.