I have an asterisk server at home. I'd like to test one just installed elsewhere.
Both servers are behind firewalls. I can see the session start in CLI, my congratulations is apparently playing and RTP is being sent. Hearing no audio. Can send key presses and see audio playing changed. "Peer audio RTP is at port 198.145.28.177:10180", but that never shows at the client side, behind a linksys wrt54g, ver 1. w/ latest firmware update. My belief is this should be possible, as the SIP phone is registered to my asterisk box "inside" my home network, asterisk should "stay in the middle" and forward the RTP packets to my laptop... am I totally off base? If so, what are some key elements to make that happen? I'll stop now, before I get ignored for being too verbose. '-) Cheers, -- |\ /| | | ~ ~ | \/ | |---| `|` ? | |ichael | |iggins \^ / michael.higgins[at]evolone[dot]org _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
