I experimented a little bit and Asterisk behind NAT with SIP works. I created an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
Great! I copied your information for other users to the Wiki.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER
Now, I have to check why it doesn't work on my Asterisk. Propably newbie behind console keyboard, but anyway...
There has been a fair amount of discussion on the list as to whether nat
works with various/different configurations of sip phones with *. The
"exact" configuration required is highly dependent on a number of technical
factors that must be well understood before anyone can make a generic
statement relative to whether it works or doesn't work. Without that
understanding, practically every statement made on the list has been based on opinion and/or some trial & error methodology that has resulted
in a working example. (Nothing wrong with that, but the majority of the
postings leave out critical info that causes the next person to attempt
the same implementation but fails, and additional questions are generated.)
Rich, Thank you for your additional information on the NAT/VoIP issue. Is it ok with you if I add it to the Wiki?
As you say, we need to collect information and compose a data base of what works and what's not working in certain circumstances.
Jan got * -> SER working, I can't. We have different NAT:s. To try to solve my problem I made sure his solution was documented so far.
There's no silver bullet here. With NATs, we've built a network without end-to-end connectivity and we need to patch it up to get VoIP working on an IPv4 network with NATs in every corner.
I just hope that IPv6 will make life easier for the next generation of VoIP users. Right now, we need to understand all variables.
/O
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