I think I'm able to connect with Ekiga, at least it reports "registered". Curiously, when I exit Ekiga and switch to SFLphone, it isn't able to connect with the exact same parameters; it just says "trying" and never resolves.
I'm not able to test outside connectivity because of too many hops: thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:[email protected] -m "hi" No SRV record: _sip._tcp.ekiga.net No SRV record: _sip._udp.ekiga.net using A record: ekiga.net Max-Forwards set to 0 message received: SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 192.168.1.3:44370;branch=z9hG4bK.6f1c2f33;rport=44370;alias;received=96.48.128.162 From: sip:[email protected]:44370;tag=981cae4 To: sip:[email protected];tag=c64e1f832a41ec1c1f4e5673ac5b80f6.bf6c Call-ID: [email protected] CSeq: 1 OPTIONS Server: Kamailio (1.5.3-notls (i386/linux)) Content-Length: 0 ** reply received after 161.445 ms ** SIP/2.0 483 Too Many Hops final received thufir@doge:~$ but that's ok. How can I test, I mean make a voice call, given that I only have two computers to work with at the moment? The server runs Asterisk on tleilax, and doge is the client. Both connect to the same router. the ip address for tleilax is 192.168.1.2 and the ip address for doge is 192.168.1.3 (generally; doge uses DHCP). When I get more pc's, I can maybe have doge call another pc on the network, but, for right now, what can I do to test this out? thanks, Thufir -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
