Hi! I'am trying to create a fake call to extension passing some additional information in custom SIP header "User-to-User". Command syntax: originate {sip_h_X-User-to-User=48656c6c6f;encoding=hex}sofia/internal/1002@10.10.104.125 &playback('test.wav') XML test Call originated but according to sofia siptrace my custom header doesn't transmit to 1002 endpoint. SIP trace looks as follows: send 1298 bytes to udp/[10.10.104.125]:5060 at 08:12:48.524061: ------------------------------------------------------------------------ INVITE sip:1002@10.10.104.125 SIP/2.0 Via: SIP/2.0/UDP 10.10.104.125;rport;branch=z9hG4bKU0j6a2D3Zg02g Max-Forwards: 70 From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:mod_sofia@10.10.104.125:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 373 X-User-to-User: 48656c6c6f;encoding=hex X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@10.10.104.125>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1371708570 1371708571 IN IP4 10.10.104.125 s=FreeSWITCH c=IN IP4 10.10.104.125 t=0 0 m=audio 7398 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 7198 RTP/AVP 98 99 34 31 100 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 a=rtpmap:100 H263-2000/90000 ------------------------------------------------------------------------ recv 1298 bytes from udp/[10.10.104.125]:5060 at 08:12:48.525061: ------------------------------------------------------------------------ INVITE sip:1002@10.10.104.125 SIP/2.0 Via: SIP/2.0/UDP 10.10.104.125;rport;branch=z9hG4bKU0j6a2D3Zg02g Max-Forwards: 70 From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:mod_sofia@10.10.104.125:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 373 X-User-to-User: 48656c6c6f;encoding=hex X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@10.10.104.125>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1371708570 1371708571 IN IP4 10.10.104.125 s=FreeSWITCH c=IN IP4 10.10.104.125 t=0 0 m=audio 7398 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 7198 RTP/AVP 98 99 34 31 100 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 a=rtpmap:100 H263-2000/90000 ------------------------------------------------------------------------ send 319 bytes to udp/[10.10.104.125]:5060 at 08:12:48.525061: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ recv 319 bytes from udp/[10.10.104.125]:5060 at 08:12:48.525061: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125> Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ send 1352 bytes to tcp/[10.10.104.125]:30101 at 08:12:48.551064: ------------------------------------------------------------------------ INVITE sip:1002@10.10.104.125:30101;transport=TCP;ob SIP/2.0 Via: SIP/2.0/TCP 10.10.104.125;branch=z9hG4bKv9BZcXy6vSpNc Max-Forwards: 69 From: "" <sip:0000000000@10.10.104.125>;tag=UvKaX51QD0XSp To: <sip:1002@10.10.104.125:30101;transport=TCP;ob> Call-ID: 09412ee0-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:mod_sofia@10.10.104.125:5060;transport=tcp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 373 X-User-to-User: 48656c6c6f;encoding=hex X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@10.10.104.125>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1371708570 1371708571 IN IP4 10.10.104.125 s=FreeSWITCH c=IN IP4 10.10.104.125 t=0 0 m=audio 7398 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 7198 RTP/AVP 98 99 34 31 100 a=rtpmap:98 H264/90000 a=rtpmap:99 H263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 a=rtpmap:100 H263-2000/90000 ------------------------------------------------------------------------ recv 284 bytes from tcp/[10.10.104.125]:30101 at 08:12:48.558064: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.10.104.125;received=10.10.104.125;branch=z9hG4bKv9BZcXy6vSpNc Call-ID: 09412ee0-5424-1231-178e-171268c989f7 From: <sip:0000000000@10.10.104.125>;tag=UvKaX51QD0XSp To: <sip:1002@10.10.104.125;ob> CSeq: 45521920 INVITE Content-Length: 0 ------------------------------------------------------------------------ recv 479 bytes from tcp/[10.10.104.125]:30101 at 08:12:48.577066: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/TCP 10.10.104.125;received=10.10.104.125;branch=z9hG4bKv9BZcXy6vSpNc Call-ID: 09412ee0-5424-1231-178e-171268c989f7 From: <sip:0000000000@10.10.104.125>;tag=UvKaX51QD0XSp To: <sip:1002@10.10.104.125;ob>;tag=d1fb24be0fba4269bbf3fce36d8bb485 CSeq: 45521920 INVITE Contact: "1002" <sip:1002@10.10.104.125:30101;transport=TCP;ob> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 930 bytes to udp/[10.10.104.125]:5060 at 08:12:48.580067: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125>;tag=tKtHUaHmgQ76a Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:1002@10.10.104.125:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Length: 0 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1002@10.10.104.125 X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:1002@10.10.104.125>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ recv 930 bytes from udp/[10.10.104.125]:5060 at 08:12:48.580067: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.104.125;rport=5060;branch=z9hG4bKU0j6a2D3Zg02g From: "" <sip:0000000000@10.10.104.125>;tag=Sa1rSF0gKeHmF To: <sip:1002@10.10.104.125>;tag=tKtHUaHmgQ76a Call-ID: 093d3725-5424-1231-178e-171268c989f7 CSeq: 45521920 INVITE Contact: <sip:1002@10.10.104.125:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summar y, refer Content-Length: 0 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:1002@10.10.104.125 X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:1002@10.10.104.125>;party=calling;privacy=off;screen=no Maybe there is some originate command option to translate SIP headers, or some sort of variable to do that? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Passing-custom-SIP-header-thro... Sent from the freeswitch-users mailing list archive at Nabble.com.