[OpenSIPS-Users] Help with Inbound PSTN, and Inbound SIP URI Authentication Sub-Routine
Kennard_White at logitech.com
Kennard_White at logitech.com
Tue Sep 14 17:44:14 CEST 2010
Hi Brett,
For what it is worth, I do it the other way around: I check the source IP,
and if from a PSTN provider process the telephone number as appropriate for
them; otherwise I do user auth.
A question: if you're allowing "outside" users to call in, why authenticate
any INVITE traffic? (Ok, you have to authenticate traffic going to PSTN
from your subscribers, but other than that...)?
Regards,
Kennard
From: Brett Woollum <brett at woollum.com>
To: OpenSIPS users mailling list <users at lists.opensips.org>
Date: 09/14/2010 02:26 AM
Subject: Re: [OpenSIPS-Users] Help with Inbound PSTN, and Inbound SIP
URI Authentication Sub-Routine
Sent by: users-bounces at lists.opensips.org
David,
The "is_from_local" function is just what I needed. It will allow me to
decipher whether or not the user appears local or not, and authenticate
them if so (ie: a subscriber), or check their IP if not (ie: from my gw).
Thanks!
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 1:08:38 AM GMT -08:00 US/Canada Pacific
Subject: Re: [OpenSIPS-Users] Help with Inbound PSTN, and Inbound SIP URI
Authentication Sub-Routine
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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