[SR-Users] Kamailio 5.0 doesn't forward Authorization (Digest username) to Asterisk

Hi Georg, I believe my issue relates to this function: # Forward REGISTER to Asterisk route[REGFWD] {     if(!is_method("REGISTER"))     {         return;     }     $var(rip) = $sel(cfg_get.asterisk.bindip);     $uac_req(method)="REGISTER";     $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport);     $uac_req(furi)="sip:" + $au + "@" + $var(rip);     $uac_req(turi)="sip:" + $au + "@" + $var(rip);     $uac_req(hdrs)="Contact: <sip:" + $au + "@"                 + $sel(cfg_get.kamailio.bindip)                 + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n";     if($sel(contact.expires) != $null)         $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n";     else         $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n";     uac_req_send(); } #!endif Where should I put your code to test it? RegardsMarko ----- Forwarded Message ----- From: gh &lt;gh(a)heissa.de&gt; To: Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.kamailio.org&gt; Sent: Sunday, April 16, 2017 6:47 PM Subject: [SR-Users] Kamailio 5.0 doesn't forward Authorization (Digest username) to Asterisk Hi Marko, i work on the same scenario. It should work with: if(uac_reg_request_to("$fU", 0))     {         xlog("L_NOTICE", "Found remote user [$rU] on [$rd] via [$du]");         t_on_failure("REMOTE_AUTH");         t_relay();     } But in my case xlog() currently does not print out the variables, so i am not able to debug:-( Can you try this code and tell me if xlog() works for you? Best regards Georg MY ISSUE:---------------------- Hi Daniel, yes I want to register my SIP clients twice to have full media path through Asterisk. Now I register eg. clients 31 & 32 on Kamailio and when I try to call 31->32 then my call reaches Asterisk but it doesn't find the callee because it isn't registered in Asterisk. Am I right that Kamailio cuts a part of register message header (Authorisation - Digest name) which it receives from SIP-client and forward just such shortened message to Asterisk or there is an other problem? Is there any better approach for above requirement (full media path through Asterisk for all calls and for all users registered by Kamailio)? Thank you Regards Marko