Hello, Do not know the details about your setup. But you usually change the request URI if you want to route an INVITE to a certain user. Changing From/To headers is normally more for number display and this kind of topics.
If the routing does not work with the $ru changes I would suggest to do some traces and observe the SIP message flow between the systems. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <[email protected]> On Behalf Of Derek OSS Sent: Tuesday, August 23, 2022 6:26 PM To: [email protected] Subject: [SR-Users] Kamailio 5.3 call forwarding Hi, I'm using Kamailio 5.3 and I am interested in doing call forwarding. I am interested in having a user "alice" bypass the SIP REGISTER step and have their SIP INVITE request/call forwarded to another SIP user "bob" (who, let's just say for now is within the same domain, so accessible by the same SCSCF and if it makes things simpler, let's say that bob went through the register step). In the PCSCF configuration, I've tried doing something like this: route { route(TEST) ... } route[TEST] { $ru = "sip:[email protected]<mailto:sip%[email protected]>" if (!t_relay("scscf.domain.com<http://scscf.domain.com>",6060)) { sl_reply_error(); } exit; } Instead of $ru, I've also tried using uac_replace_to, providing both the To and URI fields with Bob's SIP URI. When I try using the $ru, the S-CSCF still seems to use Alice's R-URI instead of Bob's. When I try using uac_replace_to, I get a 403 error saying that I need to register with the S-CSCF first. Am I going about this the correct way? Thank you. Derek
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