Bruce,
I think this is related to your firewall. You may want to take a look a posting I did a few weeks ago.
http://lists.digium.com/pipermail/asterisk-users/2004-May/046511.html
Something on this topic probably belongs in the wiki.
-brian
Bruce Komito wrote:
I have a problem that is almost certainly nat-related, but I can't figure out what's happening.
Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call terminates immediately because I am watching the CDRs come out. The * server is on a public address with no firewall between it and the outside world.
sip.conf: (both extensions have identical settings) ; Bruce [5815] type=friend username=5815 secret=wpti5815 host=dynamic [EMAIL PROTECTED] context=vpbx-wpti qualify=3000 dtmfmode=inband disallow=all allow=ulaw allow=alaw nat=yes
I'm thinking this has something to do with a setting in the Sipura, but I don't know where to start. I have nat keep-alive turned on, but I had to turn stun off because it was causing a long, inexplicable delay after dialing before the call would complete.
I'm realizing NAT with VoIP is a real problem. Anyone have a silver bullet they wish to share?
Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115
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