Thank you very much Liviu! Mit freundlichen Grüßen Dimitry Nagorny Trainee
Von: [email protected] [mailto:[email protected]] Im Auftrag von Liviu Chircu Gesendet: Mittwoch, 16. März 2016 12:18 An: [email protected] 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@PBX! $rU@$rd:$rp via $si"); $rU="185511"; $rd="192.168.1.30"; $rp="5060"; $du="sip:[email protected]: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<mailto:sip:[email protected];user=phone> SIP/2.0 To: sip:[email protected];user=phone<mailto:sip:[email protected];user=phone> From: "bla" <sip:[email protected];user=phone><mailto: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<mailto:sip:[email protected];user=phone> From: "bla" <sip:[email protected] 0;user=phone><mailto:sip:[email protected] 0;user=phone>;tag=7d6bed4b820d54e0c0ae4cb86f442b81 # U 192.168.1.150:5060 -> 192.168.1.30:5060 INVITE sip:[email protected]:5060;user=phone SIP/2.0 To: sip:[email protected];user=phone From: "bla" <sip:2031 @192.168.1.30;user=phone<sip:2031%[email protected];user=phone>>;tag=7d6bed4b820d54e0c0ae4cb86f442b81 # U 192.168.1.30:5060 -> 192.168.1.150:5060 SIP/2.0 100 Trying To: sip:[email protected];user=phone From: "bla" <sip:[email protected];user=phone><mailto: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 robot5 GmbH _______________________________________________ Users mailing list [email protected]<mailto:[email protected]> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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