<--- SIP read from UDP:66.54.140.46:5060 --->
INVITE sip:2532429604@96.255.142.63 SIP/2.0
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK6895be24;rport
From: "ALEXANDRIA VA" <sip:7035772867@66.54.140.46>;tag=as76942305
To: <sip:2532429604@96.255.142.63>
Contact: <sip:7035772867@66.54.140.46>
Call-ID: 14b6da2d0da63e07651f42fb689fafeb@66.54.140.46
CSeq: 102 INVITE
User-Agent: IPKall
Max-Forwards: 70
Date: Mon, 16 Jan 2012 03:43:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 334

v=0
o=root 14028 14028 IN IP4 66.54.140.46
s=session
c=IN IP4 66.54.140.46
t=0 0
m=audio 19596 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 16 lines) ---
Sending to 66.54.140.46:5060 (NAT)
Using INVITE request as basis request - 14b6da2d0da63e07651f42fb689fafeb@66.54.140.46
No matching peer for '7035772867' from '66.54.140.46:5060'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 66.54.140.46:19596
Looking for 2532429604 in from-sip-external (domain 96.255.142.63)
list_route: hop: <sip:7035772867@66.54.140.46>

<--- Transmitting (NAT) to 66.54.140.46:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK6895be24;received=66.54.140.46;rport=5060
From: "ALEXANDRIA VA" <sip:7035772867@66.54.140.46>;tag=as76942305
To: <sip:2532429604@96.255.142.63>
Call-ID: 14b6da2d0da63e07651f42fb689fafeb@66.54.140.46
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2532429604@96.255.142.63:5060>
Content-Length: 0


<------------>
    -- Executing [2532429604@from-sip-external:1] NoOp("SIP/66.54.140.46-00000038", "Received incoming SIP connection from unknown peer to 2532429604") in new stack
    -- Executing [2532429604@from-sip-external:2] Set("SIP/66.54.140.46-00000038", "DID=2532429604") in new stack
    -- Executing [2532429604@from-sip-external:3] Goto("SIP/66.54.140.46-00000038", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/66.54.140.46-00000038", "1?checklang:noanonymous") in new stack
    -- Goto (from-sip-external,s,2)
    -- Executing [s@from-sip-external:2] GotoIf("SIP/66.54.140.46-00000038", "0?setlanguage:from-trunk,2532429604,1") in new stack
    -- Goto (from-trunk,2532429604,1)
    -- Executing [2532429604@from-trunk:1] Set("SIP/66.54.140.46-00000038", "__FROM_DID=2532429604") in new stack
    -- Executing [2532429604@from-trunk:2] Gosub("SIP/66.54.140.46-00000038", "app-blacklist-check,s,1") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/66.54.140.46-00000038", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/66.54.140.46-00000038", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/66.54.140.46-00000038", "") in new stack
    -- Executing [2532429604@from-trunk:3] ExecIf("SIP/66.54.140.46-00000038", "0 ?Set(CALLERID(name)=7035772867)") in new stack
    -- Executing [2532429604@from-trunk:4] Ringing("SIP/66.54.140.46-00000038", "") in new stack

<--- Transmitting (NAT) to 66.54.140.46:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK6895be24;received=66.54.140.46;rport=5060
From: "ALEXANDRIA VA" <sip:7035772867@66.54.140.46>;tag=as76942305
To: <sip:2532429604@96.255.142.63>;tag=as7c5e0317
Call-ID: 14b6da2d0da63e07651f42fb689fafeb@66.54.140.46
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2532429604@96.255.142.63:5060>
Content-Length: 0


<------------>
    -- Executing [2532429604@from-trunk:5] Set("SIP/66.54.140.46-00000038", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [2532429604@from-trunk:6] Set("SIP/66.54.140.46-00000038", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [2532429604@from-trunk:7] Goto("SIP/66.54.140.46-00000038", "from-did-direct,2002,1") in new stack
    -- Goto (from-did-direct,2002,1)
    -- Executing [2002@from-did-direct:1] ExecIf("SIP/66.54.140.46-00000038", "0?Set(__RINGTIMER=0)") in new stack
    -- Executing [2002@from-did-direct:2] Macro("SIP/66.54.140.46-00000038", "exten-vm,2002,2002,0,0,0") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/66.54.140.46-00000038", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/66.54.140.46-00000038", "AMPUSER=7035772867") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/66.54.140.46-00000038", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/66.54.140.46-00000038", "1?Set(REALCALLERIDNUM=7035772867)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/66.54.140.46-00000038", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/66.54.140.46-00000038", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/66.54.140.46-00000038", "1?report") in new stack
    -- Goto (macro-user-callerid,s,13)
    -- Executing [s@macro-user-callerid:13] GotoIf("SIP/66.54.140.46-00000038", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:14] Set("SIP/66.54.140.46-00000038", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:15] GotoIf("SIP/66.54.140.46-00000038", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,26)
    -- Executing [s@macro-user-callerid:26] Set("SIP/66.54.140.46-00000038", "CALLERID(number)=7035772867") in new stack
    -- Executing [s@macro-user-callerid:27] Set("SIP/66.54.140.46-00000038", "CALLERID(name)=ALEXANDRIA VA") in new stack
    -- Executing [s@macro-user-callerid:28] Set("SIP/66.54.140.46-00000038", "CHANNEL(language)=en") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/66.54.140.46-00000038", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/66.54.140.46-00000038", "__EXTTOCALL=2002") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/66.54.140.46-00000038", "__PICKUPMARK=2002") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/66.54.140.46-00000038", "RT=15") in new stack
    -- Executing [s@macro-exten-vm:6] Macro("SIP/66.54.140.46-00000038", "record-enable,2002,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/66.54.140.46-00000038", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/66.54.140.46-00000038", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/66.54.140.46-00000038", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,14)
    -- Executing [s@macro-record-enable:14] GotoIf("SIP/66.54.140.46-00000038", "1?IN") in new stack
    -- Goto (macro-record-enable,s,18)
    -- Executing [s@macro-record-enable:18] ExecIf("SIP/66.54.140.46-00000038", "1?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:7] GotoIf("SIP/66.54.140.46-00000038", "1?macrodial") in new stack
    -- Goto (macro-exten-vm,s,13)
    -- Executing [s@macro-exten-vm:13] GosubIf("SIP/66.54.140.46-00000038", "0?clrheader,1") in new stack
    -- Executing [s@macro-exten-vm:14] Macro("SIP/66.54.140.46-00000038", "dial-one,15,tr,2002") in new stack
    -- Executing [s@macro-dial-one:1] Set("SIP/66.54.140.46-00000038", "DEXTEN=2002") in new stack
    -- Executing [s@macro-dial-one:2] Set("SIP/66.54.140.46-00000038", "DIALSTATUS_CW=") in new stack
    -- Executing [s@macro-dial-one:3] GosubIf("SIP/66.54.140.46-00000038", "0?screen,1") in new stack
    -- Executing [s@macro-dial-one:4] GosubIf("SIP/66.54.140.46-00000038", "0?cf,1") in new stack
    -- Executing [s@macro-dial-one:5] GotoIf("SIP/66.54.140.46-00000038", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,8)
    -- Executing [s@macro-dial-one:8] GotoIf("SIP/66.54.140.46-00000038", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:9] GotoIf("SIP/66.54.140.46-00000038", "0?continue") in new stack
    -- Executing [s@macro-dial-one:10] Set("SIP/66.54.140.46-00000038", "EXTHASCW=ENABLED") in new stack
    -- Executing [s@macro-dial-one:11] GotoIf("SIP/66.54.140.46-00000038", "0?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,23)
    -- Executing [s@macro-dial-one:23] GotoIf("SIP/66.54.140.46-00000038", "1?next3:continue") in new stack
    -- Goto (macro-dial-one,s,24)
    -- Executing [s@macro-dial-one:24] ExecIf("SIP/66.54.140.46-00000038", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
    -- Executing [s@macro-dial-one:25] GotoIf("SIP/66.54.140.46-00000038", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:26] GosubIf("SIP/66.54.140.46-00000038", "1?dstring,1:dlocal,1") in new stack
    -- Executing [dstring@macro-dial-one:1] Set("SIP/66.54.140.46-00000038", "DSTRING=") in new stack
    -- Executing [dstring@macro-dial-one:2] Set("SIP/66.54.140.46-00000038", "DEVICES=2002") in new stack
    -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/66.54.140.46-00000038", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/66.54.140.46-00000038", "0?Set(DEVICES=002)") in new stack
    -- Executing [dstring@macro-dial-one:5] Set("SIP/66.54.140.46-00000038", "LOOPCNT=1") in new stack
    -- Executing [dstring@macro-dial-one:6] Set("SIP/66.54.140.46-00000038", "ITER=1") in new stack
    -- Executing [dstring@macro-dial-one:7] Set("SIP/66.54.140.46-00000038", "THISDIAL=SIP/2002") in new stack
    -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/66.54.140.46-00000038", "1?zap2dahdi,1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/66.54.140.46-00000038", "0?Return()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/66.54.140.46-00000038", "NEWDIAL=") in new stack
    -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/66.54.140.46-00000038", "LOOPCNT2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/66.54.140.46-00000038", "ITER2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/66.54.140.46-00000038", "THISPART2=SIP/2002") in new stack
    -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/66.54.140.46-00000038", "0?Set(THISPART2=DAHDI/2002)") in new stack
    -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/66.54.140.46-00000038", "NEWDIAL=SIP/2002&") in new stack
    -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/66.54.140.46-00000038", "ITER2=2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/66.54.140.46-00000038", "0?begin2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/66.54.140.46-00000038", "THISDIAL=SIP/2002") in new stack
    -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/66.54.140.46-00000038", "") in new stack
    -- Executing [dstring@macro-dial-one:9] Set("SIP/66.54.140.46-00000038", "DSTRING=SIP/2002&") in new stack
    -- Executing [dstring@macro-dial-one:10] Set("SIP/66.54.140.46-00000038", "ITER=2") in new stack
    -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/66.54.140.46-00000038", "0?begin") in new stack
    -- Executing [dstring@macro-dial-one:12] Set("SIP/66.54.140.46-00000038", "DSTRING=SIP/2002") in new stack
    -- Executing [dstring@macro-dial-one:13] Return("SIP/66.54.140.46-00000038", "") in new stack
    -- Executing [s@macro-dial-one:27] GotoIf("SIP/66.54.140.46-00000038", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:28] GotoIf("SIP/66.54.140.46-00000038", "1?skiptrace") in new stack
    -- Goto (macro-dial-one,s,30)
    -- Executing [s@macro-dial-one:30] Set("SIP/66.54.140.46-00000038", "D_OPTIONS=tr") in new stack
    -- Executing [s@macro-dial-one:31] ExecIf("SIP/66.54.140.46-00000038", "0?SIPAddHeader(Alert-Info: )") in new stack
    -- Executing [s@macro-dial-one:32] ExecIf("SIP/66.54.140.46-00000038", "0?SIPAddHeader()") in new stack
    -- Executing [s@macro-dial-one:33] ExecIf("SIP/66.54.140.46-00000038", "0?Set(CHANNEL(musicclass)=)") in new stack
    -- Executing [s@macro-dial-one:34] GosubIf("SIP/66.54.140.46-00000038", "0?qwait,1") in new stack
    -- Executing [s@macro-dial-one:35] Set("SIP/66.54.140.46-00000038", "__CWIGNORE=") in new stack
    -- Executing [s@macro-dial-one:36] Set("SIP/66.54.140.46-00000038", "__KEEPCID=TRUE") in new stack
    -- Executing [s@macro-dial-one:37] GotoIf("SIP/66.54.140.46-00000038", "0?usegoto,1") in new stack
    -- Executing [s@macro-dial-one:38] GotoIf("SIP/66.54.140.46-00000038", "0?godial") in new stack
    -- Executing [s@macro-dial-one:39] Set("SIP/66.54.140.46-00000038", "CONNECTEDLINE(name,i)=2002") in new stack
    -- Executing [s@macro-dial-one:40] Set("SIP/66.54.140.46-00000038", "CONNECTEDLINE(num)=2002") in new stack
    -- Executing [s@macro-dial-one:41] Set("SIP/66.54.140.46-00000038", "D_OPTIONS=trI") in new stack
    -- Executing [s@macro-dial-one:42] Dial("SIP/66.54.140.46-00000038", "SIP/2002,15,trI") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.232:58173:
INVITE sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.3:5060;branch=z9hG4bK7b4c416d
Max-Forwards: 70
From: "ALEXANDRIA VA" <sip:7035772867@192.168.254.3>;tag=as508af574
To: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>
Contact: <sip:7035772867@192.168.254.3:5060>
Call-ID: 62de6f971f5d22256274e0b972c27940@192.168.254.3:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.7.1)
Date: Mon, 16 Jan 2012 08:43:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 112272996 112272996 IN IP4 192.168.254.3
s=Asterisk PBX 1.8.7.1
c=IN IP4 192.168.254.3
t=0 0
m=audio 16476 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/2002

<--- Transmitting (NAT) to 66.54.140.46:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK6895be24;received=66.54.140.46;rport=5060
From: "ALEXANDRIA VA" <sip:7035772867@66.54.140.46>;tag=as76942305
To: <sip:2532429604@96.255.142.63>;tag=as7c5e0317
Call-ID: 14b6da2d0da63e07651f42fb689fafeb@66.54.140.46
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2532429604@96.255.142.63:5060>
Content-Length: 0


<------------>
    -- Connected line update to SIP/66.54.140.46-00000038 prevented.

<--- SIP read from UDP:192.168.2.232:58173 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.254.3:5060;branch=z9hG4bK7b4c416d
Contact: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>
To: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>;tag=6621b323
From: "ALEXANDRIA VA"<sip:7035772867@192.168.254.3>;tag=as508af574
Call-ID: 62de6f971f5d22256274e0b972c27940@192.168.254.3:5060
CSeq: 102 INVITE
User-Agent: 3CXPhone 6.0.20943.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/2002-00000039 is ringing

<--- Transmitting (NAT) to 66.54.140.46:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK6895be24;received=66.54.140.46;rport=5060
From: "ALEXANDRIA VA" <sip:7035772867@66.54.140.46>;tag=as76942305
To: <sip:2532429604@96.255.142.63>;tag=as7c5e0317
Call-ID: 14b6da2d0da63e07651f42fb689fafeb@66.54.140.46
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2532429604@96.255.142.63:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.2.232:58173 --->


<------------->

<--- SIP read from UDP:192.168.2.232:58173 --->
REGISTER sip:192.168.254.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.232:58173;branch=z9hG4bK-d8754z-9a436177126c3200-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>
To: "2002"<sip:2002@192.168.254.3:5060>
From: "2002"<sip:2002@192.168.254.3:5060>;tag=fa3d7126
Call-ID: NjU3OThkYmNiNWE2ZjQ0ZDQyOGZmMTNiOTk3YzNjYWM.
CSeq: 11 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.20943.0
Authorization: Digest username="2002",realm="asterisk",nonce="18a3a008",uri="sip:192.168.254.3:5060",response="3d3a1cb1977b89322277b714438520f1",algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.2.232:58173 (NAT)

<--- Transmitting (no NAT) to 192.168.2.232:58173 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.232:58173;branch=z9hG4bK-d8754z-9a436177126c3200-1---d8754z-;received=192.168.2.232;rport=58173
From: "2002"<sip:2002@192.168.254.3:5060>;tag=fa3d7126
To: "2002"<sip:2002@192.168.254.3:5060>
Call-ID: NjU3OThkYmNiNWE2ZjQ0ZDQyOGZmMTNiOTk3YzNjYWM.
CSeq: 11 REGISTER
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 192.168.2.232:58173 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.232:58173;branch=z9hG4bK-d8754z-9a436177126c3200-1---d8754z-;received=192.168.2.232;rport=58173
From: "2002"<sip:2002@192.168.254.3:5060>;tag=fa3d7126
To: "2002"<sip:2002@192.168.254.3:5060>;tag=as0a61f869
Call-ID: NjU3OThkYmNiNWE2ZjQ0ZDQyOGZmMTNiOTk3YzNjYWM.
CSeq: 11 REGISTER
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7e081c30"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NjU3OThkYmNiNWE2ZjQ0ZDQyOGZmMTNiOTk3YzNjYWM.' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.2.232:58173 --->
REGISTER sip:192.168.254.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.232:58173;branch=z9hG4bK-d8754z-ac2e2a730a26f836-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>
To: "2002"<sip:2002@192.168.254.3:5060>
From: "2002"<sip:2002@192.168.254.3:5060>;tag=fa3d7126
Call-ID: NjU3OThkYmNiNWE2ZjQ0ZDQyOGZmMTNiOTk3YzNjYWM.
CSeq: 12 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.20943.0
Authorization: Digest username="2002",realm="asterisk",nonce="7e081c30",uri="sip:192.168.254.3:5060",response="69b688890e1c8e21c37bf791c7292d28",algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.2.232:58173 (no NAT)

<--- Transmitting (no NAT) to 192.168.2.232:58173 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.232:58173;branch=z9hG4bK-d8754z-ac2e2a730a26f836-1---d8754z-;received=192.168.2.232;rport=58173
From: "2002"<sip:2002@192.168.254.3:5060>;tag=fa3d7126
To: "2002"<sip:2002@192.168.254.3:5060>
Call-ID: NjU3OThkYmNiNWE2ZjQ0ZDQyOGZmMTNiOTk3YzNjYWM.
CSeq: 12 REGISTER
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Reliably Transmitting (no NAT) to 192.168.2.232:58173:
OPTIONS sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.3:5060;branch=z9hG4bK73dbfaae
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.254.3>;tag=as74494ee3
To: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>
Contact: <sip:Unknown@192.168.254.3:5060>
Call-ID: 2896072974c6b1b4491a293829f7265a@192.168.254.3:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.7.1)
Date: Mon, 16 Jan 2012 08:43:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.2.232:58173 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.232:58173;branch=z9hG4bK-d8754z-ac2e2a730a26f836-1---d8754z-;received=192.168.2.232;rport=58173
From: "2002"<sip:2002@192.168.254.3:5060>;tag=fa3d7126
To: "2002"<sip:2002@192.168.254.3:5060>;tag=as0a61f869
Call-ID: NjU3OThkYmNiNWE2ZjQ0ZDQyOGZmMTNiOTk3YzNjYWM.
CSeq: 12 REGISTER
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>;expires=120
Date: Mon, 16 Jan 2012 08:43:49 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NjU3OThkYmNiNWE2ZjQ0ZDQyOGZmMTNiOTk3YzNjYWM.' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.2.232:58173 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.3:5060;branch=z9hG4bK73dbfaae
Contact: <sip:192.168.2.232:58173>
To: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>;tag=ae396235
From: "Unknown"<sip:Unknown@192.168.254.3>;tag=as74494ee3
Call-ID: 2896072974c6b1b4491a293829f7265a@192.168.254.3:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
Allow-Events: presence, message-summary, tunnel-info
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '2896072974c6b1b4491a293829f7265a@192.168.254.3:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.2.232:58173 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.3:5060;branch=z9hG4bK7b4c416d
Contact: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>
To: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>;tag=6621b323
From: "ALEXANDRIA VA"<sip:7035772867@192.168.254.3>;tag=as508af574
Call-ID: 62de6f971f5d22256274e0b972c27940@192.168.254.3:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.20943.0
Content-Length: 257

v=0
o=3cxVCE 90933270 283391820 IN IP4 192.168.2.232
s=3cxVCE Audio Call
c=IN IP4 192.168.2.232
t=0 0
m=audio 40022 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.232:40022
list_route: hop: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>
set_destination: Parsing <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657> for address/port to send to
set_destination: set destination to 192.168.2.232:58173
Transmitting (no NAT) to 192.168.2.232:58173:
ACK sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.3:5060;branch=z9hG4bK6f2e7a20
Max-Forwards: 70
From: "ALEXANDRIA VA" <sip:7035772867@192.168.254.3>;tag=as508af574
To: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>;tag=6621b323
Contact: <sip:7035772867@192.168.254.3:5060>
Call-ID: 62de6f971f5d22256274e0b972c27940@192.168.254.3:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.7.1)
Content-Length: 0


---
    -- Connected line update to SIP/66.54.140.46-00000038 prevented.
    -- SIP/2002-00000039 answered SIP/66.54.140.46-00000038
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 66.54.140.46:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK6895be24;received=66.54.140.46;rport=5060
From: "ALEXANDRIA VA" <sip:7035772867@66.54.140.46>;tag=as76942305
To: <sip:2532429604@96.255.142.63>;tag=as7c5e0317
Call-ID: 14b6da2d0da63e07651f42fb689fafeb@66.54.140.46
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2532429604@96.255.142.63:5060>
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1038112530 1038112530 IN IP4 96.255.142.63
s=Asterisk PBX 1.8.7.1
c=IN IP4 96.255.142.63
t=0 0
m=audio 16414 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:66.54.140.46:5060 --->
ACK sip:2532429604@96.255.142.63:5060 SIP/2.0
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK3398432d;rport
From: "ALEXANDRIA VA" <sip:7035772867@66.54.140.46>;tag=as76942305
To: <sip:2532429604@96.255.142.63>;tag=as7c5e0317
Contact: <sip:7035772867@66.54.140.46>
Call-ID: 14b6da2d0da63e07651f42fb689fafeb@66.54.140.46
CSeq: 102 ACK
User-Agent: IPKall
Max-Forwards: 70
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.2.200:48631 --->
°
<------------->

<--- SIP read from UDP:192.168.2.232:58173 --->


<------------->
Really destroying SIP dialog 'NjU3OThkYmNiNWE2ZjQ0ZDQyOGZmMTNiOTk3YzNjYWM.' Method: REGISTER
[2012-01-16 03:44:21] NOTICE[3796]: chan_sip.c:25080 check_rtp_timeout: Disconnecting call 'SIP/66.54.140.46-00000038' for lack of RTP activity in 31 seconds
    -- Executing [h@macro-dial-one:1] Macro("SIP/66.54.140.46-00000038", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/66.54.140.46-00000038", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] Hangup("SIP/66.54.140.46-00000038", "") in new stack
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/66.54.140.46-00000038' in macro 'hangupcall'
  == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/66.54.140.46-00000038'
Scheduling destruction of SIP dialog '62de6f971f5d22256274e0b972c27940@192.168.254.3:5060' in 7296 ms (Method: INVITE)
set_destination: Parsing <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657> for address/port to send to
set_destination: set destination to 192.168.2.232:58173
Reliably Transmitting (no NAT) to 192.168.2.232:58173:
BYE sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.3:5060;branch=z9hG4bK0f69a39d
Max-Forwards: 70
From: "ALEXANDRIA VA" <sip:7035772867@192.168.254.3>;tag=as508af574
To: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>;tag=6621b323
Call-ID: 62de6f971f5d22256274e0b972c27940@192.168.254.3:5060
CSeq: 103 BYE
User-Agent: FPBX-2.9.0(1.8.7.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dial-one, s, 42) exited non-zero on 'SIP/66.54.140.46-00000038' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/66.54.140.46-00000038' in macro 'exten-vm'
  == Spawn extension (from-did-direct, 2002, 2) exited non-zero on 'SIP/66.54.140.46-00000038'
Scheduling destruction of SIP dialog '14b6da2d0da63e07651f42fb689fafeb@66.54.140.46' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:7035772867@66.54.140.46> for address/port to send to
set_destination: set destination to 66.54.140.46:5060
Reliably Transmitting (NAT) to 66.54.140.46:5060:
BYE sip:7035772867@66.54.140.46 SIP/2.0
Via: SIP/2.0/UDP 96.255.142.63:5060;branch=z9hG4bK3584c74a;rport
Max-Forwards: 70
From: <sip:2532429604@96.255.142.63>;tag=as7c5e0317
To: "ALEXANDRIA VA" <sip:7035772867@66.54.140.46>;tag=as76942305
Call-ID: 14b6da2d0da63e07651f42fb689fafeb@66.54.140.46
CSeq: 102 BYE
User-Agent: FPBX-2.9.0(1.8.7.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:66.54.140.46:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 96.255.142.63:5060;branch=z9hG4bK3584c74a;received=96.255.142.63;rport=5060
From: <sip:2532429604@96.255.142.63>;tag=as7c5e0317
To: "ALEXANDRIA VA" <sip:7035772867@66.54.140.46>;tag=as76942305
Call-ID: 14b6da2d0da63e07651f42fb689fafeb@66.54.140.46
CSeq: 102 BYE
User-Agent: IPKall
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '14b6da2d0da63e07651f42fb689fafeb@66.54.140.46' Method: ACK

<--- SIP read from UDP:192.168.2.232:58173 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.3:5060;branch=z9hG4bK0f69a39d
Contact: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>
To: <sip:2002@192.168.2.232:58173;rinstance=b0ec34b327cd2657>;tag=6621b323
From: "ALEXANDRIA VA"<sip:7035772867@192.168.254.3>;tag=as508af574
Call-ID: 62de6f971f5d22256274e0b972c27940@192.168.254.3:5060
CSeq: 103 BYE
User-Agent: 3CXPhone 6.0.20943.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '62de6f971f5d22256274e0b972c27940@192.168.254.3:5060' Method: INVITE

<--- SIP read from UDP:192.168.2.200:48631 --->
°
<------------->
freepbx*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[root@freepbx log]#