[OpenSIPS-Users] Routing from PSTN A back to PSTN A

[Nagorny, Dimitry]

Thank you very much Liviu!

Mit freundlichen Grüßen
Dimitry Nagorny
Trainee

Von: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] Im Auftrag von Liviu Chircu
Gesendet: Mittwoch, 16. März 2016 12:18
An: users at lists.opensips.org
Betreff: Re: [OpenSIPS-Users] Routing from PSTN A back to PSTN A

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 at PBX! $rU@$rd:$rp via $si");
                $rU="185511";
                $rd="192.168.1.30";
                $rp="5060";
                $du="sip:185511 at 192.168.1.30:5060";
                xlog("PBX to UA at 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:1 at 192.168.1.150;user=phone<mailto:sip:1 at 192.168.1.150;user=phone> SIP/2.0
  To: sip:1 at 192.168.1.150;user=phone<mailto:sip:1 at 192.168.1.150;user=phone>
  From: "bla" <sip:2031 at 192.168.1.30;user=phone><mailto:sip:2031 at 192.168.1.30;user=phone>;tag=7d6bed4b820d54e0c0ae4cb86f442b81
#
U 192.168.1.150:5060 -> 192.168.1.30:5060
  SIP/2.0 100 Giving a try
  To: sip:1 at 192.168.1.150;user=phone<mailto:sip:1 at 192.168.1.150;user=phone>
  From: "bla" <sip:2031 at 192.168.1.3  0;user=phone><mailto:sip:2031 at 192.168.1.3 0;user=phone>;tag=7d6bed4b820d54e0c0ae4cb86f442b81
#
U 192.168.1.150:5060 -> 192.168.1.30:5060
  INVITE sip:185511 at 192.168.1.30:5060;user=phone SIP/2.0
  To: sip:1 at 192.168.1.150;user=phone
  From: "bla" <sip:2031 @192.168.1.30;user=phone<sip:2031%20 at 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 at 192.168.1.150;user=phone
  From: "bla" <sip:2031 at 192.168.1.30;user=phone><mailto:sip:2031 at 192.168.1.30;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

robot5 GmbH





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