Asterisk: Authenticated SIP trunk

asteriskauthenticationsip

Let's say I have an Asterisk system with a bunch of connections: there are phones (who register itself with *) and providers (who wish to establish SIP trunks to put a lot of calls over, with different Caller IDs).

Here is my vision about how calls should be placed over an authenticated SIP trunk: remote end of SIP trunk should send INVITE's with From field set to it's identity (username for authentication) and Contact field set to what should be Caller ID for this call.

Is that true?

Why I believe this should be true: now, I can not specify username/secret and host=<IP> — for remote end to register I need to say host=dynamic. So, I can setup "pseudo-trunk" as a bunch of extensions, or I need to setup trunk with host and no authentication. No authentication is bad.

So, is the above true? And will asterisk match peers by looking at From field and use Caller ID from Contact field?

If that is not true — how an authenticated SIP trunk should work?

Thanks!

Best Answer

Read "asterisk the future of telephony" o'rellys book

it have all sip protocol nice described.

your qestion can be solved as

host=dynamic
defaultip=ip_of_other_side
deny=0.0.0.0/0.0.0.0
permit=ip_of_other_side/255.255.255.255

or

host=

on both side and no register string

http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf

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