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

[David J.]

  It depends on your configuration.

You can place it before or after.

Because you dont want to authenticate inbound calls, you can have a 
simple if statement that checks if the user is not local and alias 
exists, then relay to that alias.

Not real code:

if(not_from_local){
     if(alias()){
         relay;
     }
}

On 9/14/10 3:32 AM, Brett Woollum wrote:
> Hi David,
>
> As far as I can tell, the alias module is independent of how the call 
> is authenticated. My understanding is that it will look for a 
> replacement URI based on the current one, and replace if a new one is 
> found. It appears as though this "function" would go into the config 
> file somewhere after the section I'm working on now.
>
> Is my understanding correct?
>
> I'll need some way to determine if this is an inbound call (i.e.; not 
> originating from a subscriber's phone) prior to mapping it to the 
> alias module. Also, I'd like to determine if the incoming call is from 
> my PSTN gateway and give different aliases than if the call was a SIP 
> URI call.
>
> Brett Woollum
> Brett at Woollum.com
>
>
> ----- Original Message -----
> From: "David J." <david at styleflare.com>
> To: "OpenSIPS users mailling list" <users at lists.opensips.org>
> Sent: Tuesday, September 14, 2010 12:20:23 AM GMT -08:00 US/Canada Pacific
> Subject: Re: [OpenSIPS-Users] Help with Inbound PSTN, and Inbound SIP 
> URI Authentication Sub-Routine
>
> 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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