[Freeswitch-users] Passing custom SIP header through originate command
Limit
arkaha at hotbox.ru
Thu Jun 20 12:13:01 MSD 2013
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 at 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 at 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 at 10.10.104.125>;tag=Sa1rSF0gKeHmF
To: <sip:1002 at 10.10.104.125>
Call-ID: 093d3725-5424-1231-178e-171268c989f7
CSeq: 45521920 INVITE
Contact: <sip:mod_sofia at 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 at 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 at 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 at 10.10.104.125>;tag=Sa1rSF0gKeHmF
To: <sip:1002 at 10.10.104.125>
Call-ID: 093d3725-5424-1231-178e-171268c989f7
CSeq: 45521920 INVITE
Contact: <sip:mod_sofia at 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 at 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 at 10.10.104.125>;tag=Sa1rSF0gKeHmF
To: <sip:1002 at 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 at 10.10.104.125>;tag=Sa1rSF0gKeHmF
To: <sip:1002 at 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 at 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 at 10.10.104.125>;tag=UvKaX51QD0XSp
To: <sip:1002 at 10.10.104.125:30101;transport=TCP;ob>
Call-ID: 09412ee0-5424-1231-178e-171268c989f7
CSeq: 45521920 INVITE
Contact: <sip:mod_sofia at 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 at 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 at 10.10.104.125>;tag=UvKaX51QD0XSp
To: <sip:1002 at 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 at 10.10.104.125>;tag=UvKaX51QD0XSp
To: <sip:1002 at 10.10.104.125;ob>;tag=d1fb24be0fba4269bbf3fce36d8bb485
CSeq: 45521920 INVITE
Contact: "1002" <sip:1002 at 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 at 10.10.104.125>;tag=Sa1rSF0gKeHmF
To: <sip:1002 at 10.10.104.125>;tag=tKtHUaHmgQ76a
Call-ID: 093d3725-5424-1231-178e-171268c989f7
CSeq: 45521920 INVITE
Contact: <sip:1002 at 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 at 10.10.104.125
X-FS-Support: update_display,send_info
Remote-Party-ID: "Outbound Call"
<sip:1002 at 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 at 10.10.104.125>;tag=Sa1rSF0gKeHmF
To: <sip:1002 at 10.10.104.125>;tag=tKtHUaHmgQ76a
Call-ID: 093d3725-5424-1231-178e-171268c989f7
CSeq: 45521920 INVITE
Contact: <sip:1002 at 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 at 10.10.104.125
X-FS-Support: update_display,send_info
Remote-Party-ID: "Outbound Call"
<sip:1002 at 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-through-originate-command-tp7591997.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
Join us at ClueCon 2011 Aug 9-11, 2011
More information about the FreeSWITCH-users
mailing list