[asterisk-users] SIP to IAX?

[Kristian Kielhofner]

On Tue, Sep 9, 2008 at 3:34 PM, Darren Sessions <dmsessions at gmail.com> wrote:
> I would suggest using OpenSIPS with Asterisk and bypass IAX all together for
> this particular application.
> An OpenSIPS solution will take care of your traveler's NAT issues (and could
> handle the registrations) while you used Asterisk for voicemail and whatever
> else.
> I've personally used this type of general setup in the past with a great
> deal of success for remote offices and soft-phones on laptops.
> _____________________________
> Darren Sessions
> dmsessions at gmail.com
> http://www.darrensessions.com
> _____________________________
>

OpenSIPS/Kamailio will only help if the OP doesn't want to wait for
Asterisk 1.6 to mature and would like to traverse these firewalls
using SIP TLS over a non-standard port (which still may not work) to
proxy back to a standard pre-1.6 SIP TLS Asterisk system.

Their best bet is to use some type of VPN to traverse these
firewalls/NATs.  IPSEC, OpenVPN, etc.

-- 
Kristian Kielhofner
http://blog.krisk.org