[OpenSIPS-Users] Help with Inbound PSTN, and Inbound SIP URI Authentication Sub-Routine

[David J.]

  Hi Brett,

The common practice is to use the alias module for inbound routing.

You can look at the docs for its usage, but essentially you can map 
DID's to local users.



On 9/14/10 3:18 AM, Brett Woollum wrote:
> Hello!
>
> I have an OpenSIPS 1.6.3 installation that is working well. I have 
> subscribers registering to OpenSIPS, and they can dial between each 
> other and outside of my domain (to my media servers and to the PSTN). 
> All is well.
>
> I am now beginning to write the configuration that will process 
> inbound calls - meaning calls from non-subscribers. This will include 
> calls from the PSTN gateway, as well as direct SIP URI calls to the 
> OpenSIPS subscribers. For example, a person can call 515-555-1212 from 
> a regular phone, and the call will come to OpenSIPS as an 
> un-authenticated call from my PSTN gateway. Also, I'd like to accept 
> SIP URI's for incoming calls. For example, calling 
> mycompany at mysipdomain.com from a soft phone might route the call to 
> subscriber A's phone.
>
> The code I have that applies to this is: (This is currently configured 
> to authenticate all outbound calls from subscribers only.)
>     # authenticate if from local subscriber
>     if (!(method=="REGISTER")) {
>             if (!proxy_authorize("", "subscriber")) {
>                 proxy_challenge("", "0");
>                 exit;
>             }
>             if (!db_check_from()) {
>                 send_reply("403","Forbidden auth ID");
>                 exit;
>             }
>
>             consume_credentials();
>             # caller authenticated
>     }
>
> I am looking for direction on how to expand this to determine if the 
> call is A) from a subscriber calling outbound, B) inbound from the 
> PSTN, or C) inbound from any other user calling my SIP URI's. Once I 
> am able to determine this information, I'll be able to route the call 
> appropriately within the rest of my scripts.
>
> My problem is that my SIP phones usually attempt to place calls 
> without including authorization in the header (because they are 
> registered already), then OpenSIPS replies requiring proxy 
> authentication. The SIP phones will then try the call again including 
> the credentials in the header, which works. How can I re-write this 
> section of code to allow inbound SIP URI calls and calls from my PSTN 
> gateway, while still asking my subscribers to authenticate? Or, is 
> there a method that might work better?
>
> Notes:
> - Each of my PSTN gateway's has a static IP.
> - It's safe to assume a single-domain setup (mysipdomain.com).
>
> Thanks in advance!
>
> Brett Woollum
> Brett at Woollum.com
>
>
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