[Freeswitch-users] [fs_cli] Only display SIP messages?

[Fred-145]

On Thu, 28 Jan 2010 08:57:14 -0500, Javier Aristizábal
<javieraristizabal at gmail.com> wrote:
>sofia profile internal siptrace on

Thanks for the tip, but even with this command, I get a lot more data
than just the SIP dialog:

==========
Dialplan: sofia/internal/1004 at 192.168.0.7 Regex (FAIL)
[global-intercept] destination_number(1001) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1004 at 192.168.0.7 parsing
[default->group-intercept] continue=false
[...]
2010-01-28 16:07:34.149826 [DEBUG] switch_core_state_machine.c:122
(sofia/internal/1004 at 192.168.0.7) State Change CS_ROUTING ->
CS_EXECUTE
2010-01-28 16:07:34.149826 [DEBUG] switch_core_session.c:1013 Send
signal sofia/internal/1004 at 192.168.0.7 [BREAK]
[...]
send 1369 bytes to udp/[192.168.0.1]:19354 at 15:07:34.017297:

------------------------------------------------------------------------
   INVITE sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d
SIP/2.0
   Via: SIP/2.0/UDP 192.168.0.7;rport;branch=z9hG4bKpDgjj09Qv54Xj
   Max-Forwards: 69
   From: "Extension 1004" <sip:1004 at 192.168.0.7>;tag=Nra5SSp7aat4S
   To: <sip:1001 at 192.168.0.1:19354;rinstance=804198d1614a592d>
   Call-ID: b455d480-86c1-122d-f48c-00242116bb98
   CSeq: 126227291 INVITE
   Contact: <sip:mod_sofia at 192.168.0.7:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16456
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 425
   X-FS-Support: update_display
   Remote-Party-ID: "Extension 1004"
<sip:1004 at 192.168.0.7>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 1264662214 1264662215 IN IP4 192.168.0.7
   s=FreeSWITCH
   c=IN IP4 192.168.0.7
   t=0 0
   m=audio 29040 RTP/AVP 8 115 107 9 0 3 101 13
   a=rtpmap:8 PCMA/8000
==========

Can fs_cli be configured to filter everything but the SIP protocol
itself?