[Asterisk-Users] NAT + iConnectHere Broken in 1.0RC1

[Olle E. Johansson]

There has been a number of changes in how SIP handles outbound registrations.
That is registration with Asterisk as a SIP client to another SIP proxy,
propably with a service provider.

To be able to document a new "how to" I would like those of you that have problems
with this mail me a SIP DEBUG *off list*, as well as relevant parts of your SIP.conf.
Please replace passwords if you are sending me service provider account information.

Please document if you have problems with outgoing or incoming calls, or both.
If you have a NAT between your Asterisk and the service provider, send me
that information as well.

If I get too many mails, I can't promise to respond to all of you, but I'll try to
find common mistakes and sort his out once and for all in the documentation.

I'll be away over the weekend, so I'll start this process on monday.

Some notes as advice
* We are matching [peer] entries now for register= lines
* We are resolving hosts with SRV records by default
* If you want incoming calls, make sure the extension you use when you register
   is reachable in the context you have in the [general] section of sip.conf
   ...otherwise we'll try to authenticate the incoming call
* When we are calling out, a Fromuser and Fromdomain entry will change how
   we identify ourselves to the sip proxy. This is *very* useful for most
   service providers.
* READ THE SAMPLE SIP.CONF in your source code directory

The howto will be submitted to Asteriskdocs.org - the documentation project.

Have a nice SIP weekend :-)
/Olle