How can i be a IP authentification provider

Hi dears ,

I always used to be a sip client , and connect to a sip provider to root my calls abroad , now i wanna do the contrary , i wanna be a sip provider and give my clients an IP authentification .

Whould you please helpe me ?? what should i do in my sip.conf ans extensions.conf ti have to enable IP authentification to my clients ?

Thanks a looot :smile:

The same as you do to your local SIP phones.

Thanks a lot for ur replay ,

But my local sip phone have a login and a password ,
Now i want that the registring will be done only by IP Address , My addresse ans the sip client ip address .
The SIP phones should know only my IP Address :frowning:

That’s easier, but most customers of such services don’t have the static IP addresses you need for this.

You need to understand and the sample configuration files much better before you even think of running a service like that.

Specifically, if you want authentication to work inbound, don’t disable it with insecure, and do provide a password.

Thanks for replaying ,
I still can’t make my service ok :frowning:
this is what i did
Server 1 : ip address :

exten :
exten => s , 1 , Answer()
exten => s , 2 , Dial…

Server2 : Ip Adresse
Same sip config
friend connected to the server

exten :
exten => 111,1,Dial( SIP/111)
exten => 113,1,Dial( SIP/

when i want to call 113 logs in server 2give me inable to connect to the , congestion
in sever 1 i don’t hace nothing :frowning:

Whould you please give me a sample configuration ?

insecure is wrong (insecure=invite disables incoming authentication, and makes no sense without a secret).
secret is missing
Not just in this case, peer is better than friend.

I don’t understand what you mean by server IP address? (This may or may not explain why you might need friend.)

You appear to have two entries with the same host IP address. Asterisk cannot distinguish between incoming calls if they have the same address if it has to match on address.

Why are you calling Answer?

I don’t give sample configurations for relatively normal cases.

I guess 2give means “to give” (text speak) and “i don’t hace nothing” means “I get nothing” (double negative, capitalisation and typo).

You will need to turn up debugging to find out why is rejecting INVITEs to sip: Note you are connecting to, not to

Thanks for replaying
Just befor i wanted to explain my network architectur :
To make my tests , i v built a virtual environement where i have two asterisk servers . The first have the address and the second .

i v modified my configuration . but still can’t have my servers communicate :frowning:
would u please verify if i’ did the correct config in my sip :

insecure =port

thaaanks a lot for ur colaboration

canreinvite= was renamed to directmedia= in Asterisk 1.6.2 to more accurately describe what this setting does. See also the closely related setting directrtpsetup.

If what you are trying to do, is to connect 2 Asterisk boxes. This book have an example called Connecting Two Asterisk Boxes Together via SIP … -book.html