[Asterisk-Users] Serious NAT problems: can't call between lines on sipura
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Serious NAT problems: can't call between lines on sipura
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Serious NAT problems: can't call between lines on sipura
Please ignore my previous post (below), as it's not really relevant to your problem. I was in some kind of mindless auto-email processing mode and responded without fully reading your message. Too much spam, too little sleep. Geesh. -brian Brian Cuthie wrote: 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
