Hi, I'm kind of a newbie myself. I've had similar problems and it can be very frustrating. I did get them all resolved so I'll share some of what I did in hopes that it will fix your issue.
To get some of my phones to work (Grandstream BT100) I had to add a line "nat = yes" in my sip.conf under each phone config. I also had to set the bindaddr, externip, and localnet (which is a network address not a host address) in the general section of sip.conf. I'll put a plug in for Grandstream here. Their phones aren't nearly so expensive as some others and they work very well. Check to make sure they have the features you need, if they do, I definitely recommend them. I believe the issues you are having are NAT related. SIP uses one set of rules for routing and RTP uses a different set of rules. You may see other things - like one way audio - in addition once you are getting the config close. If you have the equipment, one way to isolate NAT issues with non-routeable addresses (192.168.x.x or 10.x.x.x) is to create a VPN tunnel between the network where your server is located and the network where your clients are located. If the system works until you remove the tunnel, you are definitely having NAT problems (the tunnel masks the problem because it will actually send the non-routeable packets to the other side). My personal preference is IPSec, but PPTP or L2TP should work fine for testing. I keep reading everything I can. The wiki is very helpful, even though you have to search for a while to find some answers. I also have pretty good luck by searching in Google for examples of other people's files (which an amazing number of people are kind enough to post). I have posted several issues to this bulletin board and I have gotten very good answers that way too. Don't give up. Once you get your configuration correct, Asterisk works amazingly well. I prefer it over every commercial product I've seen. Regards, John _______________________________________________ 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
