Re: [SR-Users] Problem with to: header

Can someone advise me on how to change the to header to show the host that we are sending the call to an not the servers ip. I am using dispatcher on my setup . i am getting this U 2013/09/23 12:57:54.576312 10.0.1.206:5060 <http://10.0.1.206:5060> -> 2.2.2.2:5060 <http://2.2.2.2:5060> INVITE sip:+42123333235@2.2.2.2:5060 <http://sip:+42123333235@2.2.2.2:5060> SIP/2.0. Record-Route: <sip:1.1.1.1;lr=on;ftag=as1f3df8b3>. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bKb29d.7399c2b3.0. Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK42a1ecaf;rport=5060. Max-Forwards: 16. From:<sip:unavailable@1.1.1.1 <mailto:sip%3Aunavailable@1.1.1.1>>. To: <sip:+4212333323@1.1.1.1 <mailto:sip%3A%2B4212333323@1.1.1.1>>. Contact: <sip:anonymous@1.1.1.5:5060 <http://sip:anonymous@1.1.1.5:5060>>. Call-ID: 0df8db614d45bae27035443c35166ba6@1.1.1.5:5060 <http://0df8db614d45bae27035443c35166ba6@1.1.1.5:5060>. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Mon, 23 Sep 2013 16:58:25 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Cisco-Guid: 7128745-3588944267-852064@msc1 Content-Type: application/sdp. Content-Length: 288. . v=0. o=root 1760548326 1760548326 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.5. t=0 0. m=audio 39794 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. I need to sent header to a carrier like this To: <sip:+4212333323@2.2.2.2 <mailto:sip%3A%2B4212333323@2.2.2.2>> instead of 1.1.1.1. i am proxing calls from asterisk to a main carrier. Please help. here is my config. if (starts_with("$var(o)","anonymous")) { ds_select_domain("$var(z)", "4");#carrier dynamic xlog("here is anonymous call <$var(o)>77777\n"); $var(n)=$(tU{s.substr,3,0}); remove_hf("From"); remove_hf("P-Asserted-Identity"); remove_hf("Privacy"); insert_hf("From:<sip:unavailable@1.1.1.1.1>\r\n", "From"); $tU=$var(n); xlog("out header CHECK ANONYMOUS BEFORE to $tu--$td - contact pai+++ <<$ct>>++ from_uri=$fu;<$tU---=$var(n)> to_uri=$tu; }pai<$ai>intid=$fU; type_call=$si; dst_ip=$ru; carriercode=$var(z);callmode=$var(out)"); if(!t_relay()){; sl_reply_error(); exit; }; ##ENDANONYMOUS exit; } Again thanks a lot for any help. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users