The solution can be found if someone wants to take up these requests:
- http://bugs.digium.com/bug_view_page.php?bug_id=0000359 - http://bugs.digium.com/bug_view_page.php?bug_id=0000104
JT
Chris Hariga wrote:
This is bull... I can't believe that...Read more on NAT and Voip here: http://www.voip-info.org/wiki-NAT+and+VOIP
It's not simple. The server must be reached in order to allow registration for clients. A server inside a NAT can't be reached unless you use port forwarding. A client inside can reach a server outside and there are ways to keep a connection open in the NAT. Asterisk does not support this very well.
The other side of the coin is the contstruction of SIP and SDP. That's a long story, but it ends in something like: Asterisk help clients on the inside of a NAT with NAT=yes in SIP.CONF, but can't work the other way as of today.
Must be a solution...Do some research, add som code and we will all be happy! IPv6 is a solution, if NAT is avoided.
/O
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: Monday, October 13, 2003 9:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] No sound with SIP Phones on the Internet
Chris Hariga wrote:
Yes, my Asterisk is behind a NAT but I forward all ports (100-56000) to my Linux box.
There is your problem.. Asterisk does not like playing behind NAT.. The UA's can be made to work behind NAT but the server must have a public IP address..
-- *** Olle E. Johansson, [EMAIL PROTECTED]
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