[Freeswitch-dev] SIP fantom calls, when calling disconnected SIP endpoint. EP called when plugged in. suppose bug?

[Anton VG]

Yea, there is siptrace, when enabled.

sofia global siptrace on

freeswitch at lab3> send 1181 bytes to udp/[192.168.5.21]:5062 at 15:16:40.952027:
   ------------------------------------------------------------------------
   INVITE sip:sip3779100 at 192.168.5.21:5062 SIP/2.0
   Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKN2QvX33eK72rr
   Route: <sip:sip3779100 at 192.168.5.21:5062>
   Max-Forwards: 70
   From: "" <sip:907388397 at 192.168.100.11>;tag=KXXUQ57gS66ZS
   To: <sip:sip3779100 at 192.168.5.21:5062>
   Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2
   CSeq: 12573020 INVITE
   Contact: <sip:mod_sofia at 192.168.100.11:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-fccbba5 2011-05-18
19-00-42 -0400
   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-summary,
refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 209
   X-FS-Support: update_display
   Remote-Party-ID:
<sip:907388397 at 192.168.100.11>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1305790653 1305790654 IN IP4 192.168.100.11
   s=FreeSWITCH
   c=IN IP4 192.168.100.11
   t=0 0
   m=audio 27516 RTP/AVP 0 8 3 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 304 bytes from udp/[192.168.5.21]:5062 at 15:16:40.968185:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   To: <sip:sip3779100 at 192.168.5.21:5062>
   From: "" <sip:907388397 at 192.168.100.11>;tag=KXXUQ57gS66ZS
   Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2
   CSeq: 12573020 INVITE
   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKN2QvX33eK72rr
   Server: Linksys/SPA942-6.1.3(a)
   Content-Length: 0

   ------------------------------------------------------------------------
send 414 bytes to udp/[192.168.5.21]:5062 at 15:16:40.968378:
   ------------------------------------------------------------------------
   CANCEL sip:sip3779100 at 192.168.5.21:5062 SIP/2.0
   Via: SIP/2.0/UDP 192.168.100.11;rport;branch=z9hG4bKN2QvX33eK72rr
   Route: <sip:sip3779100 at 192.168.5.21:5062>
   Max-Forwards: 70
   From: "" <sip:907388397 at 192.168.100.11>;tag=KXXUQ57gS66ZS
   To: <sip:sip3779100 at 192.168.5.21:5062>
   Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2
   CSeq: 12573020 CANCEL
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 356 bytes from udp/[192.168.5.21]:5062 at 15:16:40.984188:
   ------------------------------------------------------------------------
   SIP/2.0 481 Call Leg/Transaction Does Not Exist
   To: <sip:sip3779100 at 192.168.5.21:5062>;tag=1f405cc57aec781ei2
   From: "" <sip:907388397 at 192.168.100.11>;tag=KXXUQ57gS66ZS
   Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2
   CSeq: 12573020 CANCEL
   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKN2QvX33eK72rr
   Server: Linksys/SPA942-6.1.3(a)
   Content-Length: 0

   ------------------------------------------------------------------------
recv 386 bytes from udp/[192.168.5.21]:5062 at 15:16:40.994546:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   To: <sip:sip3779100 at 192.168.5.21:5062>;tag=4482dc0be7df9455i2
   From: "" <sip:907388397 at 192.168.100.11>;tag=KXXUQ57gS66ZS
   Call-ID: c47c3593-fccd-122e-c796-bcaec59901b2
   CSeq: 12573020 INVITE
   Via: SIP/2.0/UDP 192.168.100.11;branch=z9hG4bKN2QvX33eK72rr
   Contact: "sip3779100" <sip:sip3779100 at 192.168.5.21:5062>
   Server: Linksys/SPA942-6.1.3(a)
   Content-Length: 0