[OpenSIPS-Users] PBX sending calls to Opensips
Ali Pey
alipey at gmail.com
Wed Jul 25 22:58:40 CEST 2012
This would work too. Here though you do a db query for each call and can
slow down the performance if that's important to you. Examining usernames'
patterns can be faster.
You can also use the registered function instead of a db query:
if (registered("location","$fu")) {
xlog("caller is registered\n");
}
http://www.opensips.org/html/docs/modules/1.8.x/registrar.html#id293162
Regards,
Ali Pey
On Wed, Jul 25, 2012 at 4:23 PM, Schneur Rosenberg <rosenberg11219 at gmail.com
> wrote:
> I already did something similar look at snippet bellow so any call
> coming from a IP thats registered to our server will always do
> proxy_authorize(), other calls will assume that its a unauthenticated
> DID call or a call going to a local call
>
> if (!(method=="REGISTER"))
> {
> avp_db_query("select username from location where
> contact regexp '$si' or received like
> 'sip:$si%'","$avp(is_registered)");
> }
> if (!(method=="REGISTER") && avp_check("$avp(is_registered)",
> "gt/1/g"))
> {
> if(!is_from_gw())
> {
> if (!proxy_authorize("sosglobal", "subscriber"))
> {
> append_hf("P-hint: Proxy auth failed\r\n");
> proxy_challenge("sosglobal", "0");
> exit;
> }
>
>
> }
>
>
> On Wed, Jul 25, 2012 at 8:48 PM, Ali Pey <alipey at gmail.com> wrote:
> > Schneur,
> >
> > You can examine the src_ip first to see if the call if from your pbx or
> not.
> > Then you can also examine to request-uri to distinguish the call between
> a
> > pstn call or a sip client - assuming your sip clients have a different
> sip
> > address/pattern than pstn numbers. Things like this:
> >
> > if ( src_ip == pbx1_ip || src_ip == pbx2_ip ){
> > # From PBXs
> > }
> >
> > if ($rU=~"^\+?[0-9]{3,18}") {
> > # request-uri is for a PSTN number, send the message to whatever
> > route(1)
> > }
> >
> > Basically you need to find a difference between the call attributes and
> > examine that, it can be the src_ip, ruri pattern, etc.
> >
> > Regards,
> > Ali Pey
> >
> > On Wed, Jul 25, 2012 at 9:41 AM, Schneur Rosenberg
> > <rosenberg11219 at gmail.com> wrote:
> >>
> >> check_source_address won't work for me, my clients are behind Dynamic
> >> ip's, there is no way for me to know in advance their ip address
> >>
> >> On Mon, Jul 23, 2012 at 8:55 PM, Brett Nemeroff <brett at nemeroff.com>
> >> wrote:
> >> > Scot,
> >> > the function "is_from_local" uses the From URI and as such, will not
> >> > work if
> >> > the originator mangles the from uri (as in the case of your example
> >> > below).
> >> >
> >> > A more secure way to do this that may suit your needs is to use the
> >> > permissions module and actually check the source IP of the request:
> >> >
> >> >
> >> >
> http://www.opensips.org/html/docs/modules/1.8.x/permissions.html#id293503
> >> >
> >> > Look at the "check_source_address" and or "get_source_group". Either
> of
> >> > these can compare the source IP of the originator to a known list.
> From
> >> > there, you can perform script logic based on where the request came
> >> > from.
> >> >
> >> > Hope that helps!
> >> > -Brett
> >> >
> >> >
> >> > On Mon, Jul 23, 2012 at 11:38 AM, Schneur Rosenberg
> >> > <rosenberg11219 at gmail.com> wrote:
> >> >>
> >> >> I'm using opensips as a registrar server and as a loadbalancer, all
> >> >> phones are registered to opensips and all incoming and outgoing calls
> >> >> go to Asterisk boxes via load balancing, therefore I have 3 kinds of
> >> >> calls going to opensips,
> >> >> 1) outgoing calls coming from one of the phones Registered to
> opensips,
> >> >> 2) incoming calls (we allow all incoming calls no matter from where
> >> >> they come, I call them unauthenticated DID)
> >> >> 3) Calls ringing to a phone registered to opensips, the Asterisk
> boxes
> >> >> will send the calls to the phone either after getting a call from a
> >> >> DID, or when a internal user wants to call another internal user
> >> >>
> >> >> The way I differentiate between the calls is I do a if
> >> >> (!(method=="REGISTER") && is_from_local()) this will check
> credentials
> >> >> and send call to asterisk to process outgoing call, then I do a else
> >> >> if ((method=="INVITE")) which will check if the call is going to a
> >> >> local phone by doing if (!lookup("location", "m")) if that fails
> that
> >> >> it assumes its a incoming did call, and it will send it to asterisk
> >> >> with a prefix so asterisk knows its a unauthenticated incoming call,
> >> >> bellow I pasted a skeleton of the code I'm using.
> >> >>
> >> >> Everything worked fine, until I connected a PBX to my opensips, then
> >> >> the from came in with the address of the PBX and the is_from_local()
> >> >> test was not true, so it did not work, I had the same problem when
> >> >> sending a call from a SPA3000 and blocking caller id, in that case it
> >> >> also obscured the from address, as follows "From: Anonymous
> >> >> <sip:anonymous at localhost>;tag=ea3ee097cd947aeeo0." , the only
> >> >> reference of the user or domain was in the RPID field and calls did
> >> >> not go through.
> >> >>
> >> >> Is there anyway to check if a source IP is registered to our system
> >> >> and only then it should send a 407? this way if I have a BPX
> >> >> registered it will then ask for credentials, all others it will
> assume
> >> >> that either a call to the local phone or unauthenticated DID, I
> >> >> understand that I wont be able to send calls to the system only if
> >> >> registration was done before, but I have no problem with that, I
> >> >> could do it with avp_db_query() on the subscriber table, but I want
> to
> >> >> know if there is a better way.
> >> >>
> >> >> If there is there a better solution then the above solution please
> let
> >> >> me
> >> >> know
> >> >>
> >> >> if (!(method=="REGISTER") && is_from_local())
> >> >> {
> >> >> #check credentials
> >> >> }
> >> >> else if ((method=="INVITE")) #unathenticated did or call
> >> >> going to phone registered to opensips
> >> >> {
> >> >>
> >> >> if (!lookup("location", "m")) #calling local phone
> >> >> {
> >> >> #send to phone registered to opensips
> >> >> }
> >> >> else
> >> >> {
> >> >> #incoming did send call to asterisk to
> process
> >> >> }
> >> >> }
> >> >> else
> >> >> {
> >> >> #outgoing calls route continues here
> >> >> }
> >> >> ...................................
> >> >>
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> >> >
> >> >
> >> >
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> >> >
> >>
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> >
> >
> >
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> >
>
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