Connecting Gizmo to Asterisk
As I explained briefly here, I've set myself up with the SIP URI sip:firstname.lastname@example.org. 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 220.127.116.11 dns.ypi.com. 3600 IN A 18.104.22.168 dns.reinvented.net. 3600 IN A 22.214.171.124
- 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:email@example.com get routed directly through to the extension on my desk.