Although the scripting variables [1] are very flexible, expect each of them to only modify a single header, URI or parameter/chunk of the current SIP message.

In your case:

* $ru and $rU only work with the Request-URI, nothing more
* $du denotes a "next hop" (outbound proxy) the request will be sent to while preserving current Request-URI
  (by not setting $du, you'll just route initial requests to $ru)
* if you want to also change the "To" header field, use uac_replace_to() [2] from the "uac" module

[1]: http://www.opensips.org/Documentation/Script-CoreVar-2-2
[2]: http://www.opensips.org/html/docs/modules/2.2.x/uac.html#id293640

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 15.03.2016 18:23, Nagorny, Dimitry wrote:

Hi all,

when I shoot the following routing rule:

        if ($rU=~"^[1]$" && src_ip==192.168.1.30) {

                xlog("PBX to UA@PBX! $rU@$rd:$rp via $si");

                $rU="*185511*";

                $rd="192.168.1.30";

                $rp="5060";

                $du="sip:*185511*@192.168.1.30:5060";

                xlog("PBX to UA@PBX! $rU@$rd:$rp via $si");

force_send_socket(udp:192.168.1.150:5060);

                t_relay();

                exit;

        }

I don’t get why OpenSIPS is reverting my changes somewhere internally so this happens:

U *192.168.1.30:5060* -> 192.168.1.150:5060   (*PSTN* to OpenSIPS)

  INVITE sip:[email protected];user=phone SIP/2.0
  To: sip:[email protected];user=phone
From: "bla" <sip:[email protected];user=phone>;tag=7d6bed4b820d54e0c0ae4cb86f442b81

#

U 192.168.1.150:5060 -> 192.168.1.30:5060

  SIP/2.0 100 Giving a try
  To: sip:[email protected];user=phone
From: "bla" <sip:[email protected] 0;user=phone>;tag=7d6bed4b820d54e0c0ae4cb86f442b81

#

U 192.168.1.150:5060 -> 192.168.1.30:5060

  INVITE sip:*185511*@192.168.1.30:5060;user=phone SIP/2.0
  To: sip:*1*@192.168.1.150;user=phone
From: "bla" <sip:2031 @192.168.1.30;user=phone>;tag=7d6bed4b820d54e0c0ae4cb86f442b81

#

U 192.168.1.30:5060 -> 192.168.1.150:5060

  SIP/2.0 100 Trying
  To: sip:*1*@192.168.1.150;user=phone
From: "bla" <sip:[email protected];user=phone>;tag=7d6bed4b820d54e0c0ae4cb86f442b81

I simply want if someone from inside or outside calls a known area of numbers that they are getting relayed to a different number. Is the above routing script part wrong for my purpose?

Very Respectfully

*Dimitry Nagorny*

Trainee

robot5GmbH



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