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
[email protected]
----- Original Message -----
From: "David J." <[email protected]>
To: "OpenSIPS users mailling list" <[email protected]>
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 [email protected] 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
[email protected]
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