[2016-07-06 17:20:27] SECURITY[14222]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-07-06T17:20:27.263-0500",Severity="Informational",Service="AMI",EventVersion="1",AccountID="admin",SessionID="0x7f67c24b1918",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/56823",UsingPassword="0",SessionTV="2016-07-06T17:20:27.263-0500"

<--- SIP read from UDP:166.173.57.228:55998 --->

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

<--- SIP read from UDP:166.173.57.228:55998 --->
INVITE sip:9724247977@71.244.49.87 SIP/2.0
Via: SIP/2.0/UDP 10.54.24.68:49672;rport;branch=z9hG4bKPjHXR.R7rD.BgRd-.LmYl--K72BSccsJZi
Max-Forwards: 70
From: <sip:2@71.244.49.87>;tag=emeyyFIMa0qFK6xHKSvfxAeACoi6NWuk
To: <sip:9724247977@71.244.49.87>
Contact: <sip:2@166.173.57.228:55998;ob>
Call-ID: x7RRuTvHl-IlnwZoKOKDwZahnQaGCTmY
CSeq: 12486 INVITE
Route: <sip:71.244.49.87:5061;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_zeroflteatt-23/r2457
Content-Type: application/sdp
Content-Length: 358

v=0
o=- 3676832431 3676832431 IN IP4 10.54.24.68
s=pjmedia
c=IN IP4 10.54.24.68
t=0 0
m=audio 4002 RTP/AVP 96 3 0 8 101
c=IN IP4 10.54.24.68
a=rtcp:4003 IN IP4 10.54.24.68
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (16 headers 16 lines) ---
Sending to 166.173.57.228:55998 (NAT)
Sending to 166.173.57.228:55998 (NAT)
Using INVITE request as basis request - x7RRuTvHl-IlnwZoKOKDwZahnQaGCTmY
Found peer '2' for '2' from 166.173.57.228:55998

<--- Reliably Transmitting (NAT) to 166.173.57.228:55998 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.54.24.68:49672;branch=z9hG4bKPjHXR.R7rD.BgRd-.LmYl--K72BSccsJZi;received=166.173.57.228;rport=55998
From: <sip:2@71.244.49.87>;tag=emeyyFIMa0qFK6xHKSvfxAeACoi6NWuk
To: <sip:9724247977@71.244.49.87>;tag=as67c48d49
Call-ID: x7RRuTvHl-IlnwZoKOKDwZahnQaGCTmY
CSeq: 12486 INVITE
Server: FPBX-13.0.151(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="51a064db"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'x7RRuTvHl-IlnwZoKOKDwZahnQaGCTmY' in 6400 ms (Method: INVITE)
[2016-07-06 17:20:32] SECURITY[14222]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="ChallengeSent",EventTV="2016-07-06T17:20:32.370-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:2@71.244.49.87",SessionID="0x6109088",LocalAddress="IPV4/UDP/71.244.49.87/5061",RemoteAddress="IPV4/UDP/166.173.57.228/49672",Challenge="51a064db"

<--- SIP read from UDP:166.173.57.228:55998 --->
ACK sip:9724247977@71.244.49.87 SIP/2.0
Via: SIP/2.0/UDP 10.54.24.68:49672;rport;branch=z9hG4bKPjHXR.R7rD.BgRd-.LmYl--K72BSccsJZi
Max-Forwards: 70
From: <sip:2@71.244.49.87>;tag=emeyyFIMa0qFK6xHKSvfxAeACoi6NWuk
To: <sip:9724247977@71.244.49.87>;tag=as67c48d49
Call-ID: x7RRuTvHl-IlnwZoKOKDwZahnQaGCTmY
CSeq: 12486 ACK
Route: <sip:71.244.49.87:5061;lr>
Content-Length: 0

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

<--- SIP read from UDP:166.173.57.228:55998 --->
INVITE sip:9724247977@71.244.49.87 SIP/2.0
Via: SIP/2.0/UDP 10.54.24.68:49672;rport;branch=z9hG4bKPjuxL897wWY-1tNn003zvBL7R-gxmYwMvU
Max-Forwards: 70
From: <sip:2@71.244.49.87>;tag=emeyyFIMa0qFK6xHKSvfxAeACoi6NWuk
To: <sip:9724247977@71.244.49.87>
Contact: <sip:2@166.173.57.228:55998;ob>
Call-ID: x7RRuTvHl-IlnwZoKOKDwZahnQaGCTmY
CSeq: 12487 INVITE
Route: <sip:71.244.49.87:5061;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_zeroflteatt-23/r2457
Authorization: Digest username="2", realm="asterisk", nonce="51a064db", uri="sip:9724247977@71.244.49.87", response="943240399078b8d87f60f722a269980e", algorithm=MD5
Content-Type: application/sdp
Content-Length: 358

v=0
o=- 3676832431 3676832431 IN IP4 10.54.24.68
s=pjmedia
c=IN IP4 10.54.24.68
t=0 0
m=audio 4002 RTP/AVP 96 3 0 8 101
c=IN IP4 10.54.24.68
a=rtcp:4003 IN IP4 10.54.24.68
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (17 headers 16 lines) ---
Sending to 166.173.57.228:55998 (NAT)
Using INVITE request as basis request - x7RRuTvHl-IlnwZoKOKDwZahnQaGCTmY
Found peer '2' for '2' from 166.173.57.228:55998
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 96
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found unknown media description format SILK for ID 96
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|g722|alaw|speex|opus|g726aal2), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.54.24.68:4002
Looking for 9724247977 in from-internal (domain 71.244.49.87)
sip_route_dump: route/path hop: <sip:2@166.173.57.228:55998;ob>

<--- Transmitting (NAT) to 166.173.57.228:55998 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.54.24.68:49672;branch=z9hG4bKPjuxL897wWY-1tNn003zvBL7R-gxmYwMvU;received=166.173.57.228;rport=55998
From: <sip:2@71.244.49.87>;tag=emeyyFIMa0qFK6xHKSvfxAeACoi6NWuk
To: <sip:9724247977@71.244.49.87>
Call-ID: x7RRuTvHl-IlnwZoKOKDwZahnQaGCTmY
CSeq: 12487 INVITE
Server: FPBX-13.0.151(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:9724247977@71.244.49.87:5061>
Content-Length: 0


<------------>
[2016-07-06 17:20:32] SECURITY[14222]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-07-06T17:20:32.462-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="9724247977",SessionID="0x6109088",LocalAddress="IPV4/UDP/71.244.49.87/5061",RemoteAddress="IPV4/UDP/166.173.57.228/49672",UsingPassword="1"
    -- Executing [9724247977@from-internal:1] Macro("SIP/2-000000bc", "user-callerid,LIMIT,EXTERNAL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/2-000000bc", "TOUCH_MONITOR=1467843632.223") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/2-000000bc", "AMPUSER=2") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/2-000000bc", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/2-000000bc", "1?Set(REALCALLERIDNUM=2)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/2-000000bc", "AMPUSER=2") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/2-000000bc", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/2-000000bc", "AMPUSERCIDNAME=Ryan Post") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/2-000000bc", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/2-000000bc", "AMPUSERCID=2") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/2-000000bc", "__DIAL_OPTIONS=") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/2-000000bc", "CALLERID(all)="Ryan Post" <2>") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/2-000000bc", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("SIP/2-000000bc", "1?Set(GROUP(concurrency_limit)=2)") in new stack
    -- Executing [s@macro-user-callerid:14] ExecIf("SIP/2-000000bc", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:15] GotoIf("SIP/2-000000bc", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,29)
    -- Executing [s@macro-user-callerid:29] Set("SIP/2-000000bc", "CALLERID(number)=2") in new stack
    -- Executing [s@macro-user-callerid:30] Set("SIP/2-000000bc", "CALLERID(name)=Ryan Post") in new stack
    -- Executing [s@macro-user-callerid:31] Set("SIP/2-000000bc", "CDR(cnum)=2") in new stack
    -- Executing [s@macro-user-callerid:32] Set("SIP/2-000000bc", "CDR(cnam)=Ryan Post") in new stack
    -- Executing [s@macro-user-callerid:33] Set("SIP/2-000000bc", "CHANNEL(language)=en") in new stack
    -- Executing [9724247977@from-internal:2] Gosub("SIP/2-000000bc", "sub-record-check,s,1(out,9724247977,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("SIP/2-000000bc", "0?initialized") in new stack
    -- Executing [s@sub-record-check:2] Set("SIP/2-000000bc", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:3] Set("SIP/2-000000bc", "NOW=1467843632") in new stack
    -- Executing [s@sub-record-check:4] Set("SIP/2-000000bc", "__DAY=06") in new stack
    -- Executing [s@sub-record-check:5] Set("SIP/2-000000bc", "__MONTH=07") in new stack
    -- Executing [s@sub-record-check:6] Set("SIP/2-000000bc", "__YEAR=2016") in new stack
    -- Executing [s@sub-record-check:7] Set("SIP/2-000000bc", "__TIMESTR=20160706-172032") in new stack
    -- Executing [s@sub-record-check:8] Set("SIP/2-000000bc", "__FROMEXTEN=2") in new stack
    -- Executing [s@sub-record-check:9] Set("SIP/2-000000bc", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:10] NoOp("SIP/2-000000bc", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("SIP/2-000000bc", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("SIP/2-000000bc", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("SIP/2-000000bc", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("SIP/2-000000bc", "3?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("SIP/2-000000bc", "1?sub-record-check,out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] NoOp("SIP/2-000000bc", "Outbound Recording Check from 2 to 9724247977") in new stack
    -- Executing [out@sub-record-check:2] Set("SIP/2-000000bc", "RECMODE=dontcare") in new stack
    -- Executing [out@sub-record-check:3] ExecIf("SIP/2-000000bc", "1?Goto(routewins)") in new stack
    -- Goto (sub-record-check,out,7)
    -- Executing [out@sub-record-check:7] Gosub("SIP/2-000000bc", "recordcheck,1(dontcare,out,9724247977)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/2-000000bc", "Starting recording check against dontcare") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("SIP/2-000000bc", "dontcare") in new stack
    -- Goto (sub-record-check,recordcheck,3)
    -- Executing [recordcheck@sub-record-check:3] Return("SIP/2-000000bc", "") in new stack
    -- Executing [out@sub-record-check:8] Return("SIP/2-000000bc", "") in new stack
    -- Executing [9724247977@from-internal:3] ExecIf("SIP/2-000000bc", "0 ?Set(CDR(accountcode)=)") in new stack
    -- Executing [9724247977@from-internal:4] Set("SIP/2-000000bc", "MOHCLASS=default") in new stack
    -- Executing [9724247977@from-internal:5] Set("SIP/2-000000bc", "_NODEST=") in new stack
    -- Executing [9724247977@from-internal:6] Macro("SIP/2-000000bc", "dialout-trunk,2,9724247977,,off") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/2-000000bc", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/2-000000bc", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/2-000000bc", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/2-000000bc", "DIAL_NUMBER=9724247977") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/2-000000bc", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/2-000000bc", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/2-000000bc", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/2-000000bc", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/2-000000bc", "DIAL_TRUNK_OPTIONS=Tt") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/2-000000bc", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/2-000000bc", "0?Set(CALLERPRES(name-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/2-000000bc", "0?Set(CALLERPRES(num-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/2-000000bc", "1?Set(REALCALLERIDNUM=2)") in new stack
    -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/2-000000bc", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,7)
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/2-000000bc", "USEROUTCID=4693049888") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/2-000000bc", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:9] Set("SIP/2-000000bc", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/2-000000bc", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,15)
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/2-000000bc", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/2-000000bc", "1?Set(CALLERID(all)=4693049888)") in new stack
    -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/2-000000bc", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/2-000000bc", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/2-000000bc", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:20] Set("SIP/2-000000bc", "CDR(outbound_cnum)=4693049888") in new stack
    -- Executing [s@macro-outbound-callerid:21] Set("SIP/2-000000bc", "CDR(outbound_cnam)=") in new stack
[2016-07-06 17:20:32] WARNING[14060]: func_cdr.c:377 cdr_write_callback: CDR requires a value (CDR(variable)=value)
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/2-000000bc", "0?sub-flp-2,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/2-000000bc", "OUTNUM=9724247977") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/2-000000bc", "custom=SIP/fpbx-1-cdB7e8PklPds") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/2-000000bc", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/2-000000bc", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/2-000000bc", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/2-000000bc", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/2-000000bc", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/2-000000bc", "1?Set(CONNECTEDLINE(num,i)=9724247977)") in new stack
    -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/2-000000bc", "1?Set(CONNECTEDLINE(name,i)=CID:4693049888)") in new stack
    -- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/2-000000bc", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)4693049888)") in new stack
    -- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/2-000000bc", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:23] Dial("SIP/2-000000bc", "SIP/fpbx-1-cdB7e8PklPds/9724247977,300,Tt") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 11934
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.159.66.3:5060:
INVITE sip:9724247977@trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK6dd565d5;rport
Max-Forwards: 70
From: <sip:4693049888@71.244.49.87:5061>;tag=as16df33b4
To: <sip:9724247977@trunk1.freepbx.com>
Contact: <sip:4693049888@71.244.49.87:5061>
Call-ID: 0bde579e147e0a05206d6ae110c883cb@71.244.49.87:5061
CSeq: 102 INVITE
User-Agent: FPBX-13.0.151(13.9.1)
Date: Wed, 06 Jul 2016 22:20:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "4693049888" <sip:4693049888@71.244.49.87>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 1814590057 1814590057 IN IP4 71.244.49.87
s=Asterisk PBX 13.9.1
c=IN IP4 71.244.49.87
t=0 0
m=audio 11934 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/fpbx-1-cdB7e8PklPds/9724247977

<--- SIP read from UDP:192.159.66.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK6dd565d5;rport=5061
From: <sip:4693049888@71.244.49.87:5061>;tag=as16df33b4
To: <sip:9724247977@trunk1.freepbx.com>
Call-ID: 0bde579e147e0a05206d6ae110c883cb@71.244.49.87:5061
CSeq: 102 INVITE
User-Agent: SIPStation 2.11.3
Content-Length: 0

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

<--- SIP read from UDP:192.159.66.3:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK6dd565d5;rport=5061
From: <sip:4693049888@71.244.49.87:5061>;tag=as16df33b4
To: <sip:9724247977@trunk1.freepbx.com>;tag=88D973SBjXNeK
Call-ID: 0bde579e147e0a05206d6ae110c883cb@71.244.49.87:5061
CSeq: 102 INVITE
User-Agent: SIPStation 2.11.3
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Proxy-Authenticate: Digest realm="71.244.49.87", nonce="da8f3bc6-43c7-11e6-9b4b-0732f924a662", algorithm=MD5, qop="auth"
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Transmitting (NAT) to 192.159.66.3:5060:
ACK sip:9724247977@trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK6dd565d5;rport
Max-Forwards: 70
From: <sip:4693049888@71.244.49.87:5061>;tag=as16df33b4
To: <sip:9724247977@trunk1.freepbx.com>;tag=88D973SBjXNeK
Contact: <sip:4693049888@71.244.49.87:5061>
Call-ID: 0bde579e147e0a05206d6ae110c883cb@71.244.49.87:5061
CSeq: 102 ACK
User-Agent: FPBX-13.0.151(13.9.1)
Content-Length: 0


---
Audio is at 11934
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.159.66.3:5060:
INVITE sip:9724247977@trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK023dd7bd;rport
Max-Forwards: 70
From: <sip:4693049888@71.244.49.87:5061>;tag=as16df33b4
To: <sip:9724247977@trunk1.freepbx.com>
Contact: <sip:4693049888@71.244.49.87:5061>
Call-ID: 0bde579e147e0a05206d6ae110c883cb@71.244.49.87:5061
CSeq: 103 INVITE
User-Agent: FPBX-13.0.151(13.9.1)
Proxy-Authorization: Digest username="cdB7e8PklPds", realm="71.244.49.87", algorithm=MD5, uri="sip:9724247977@trunk1.freepbx.com", nonce="da8f3bc6-43c7-11e6-9b4b-0732f924a662", response="3b2fa608ecbeef3c1daea9c80e1df9bd", qop=auth, cnonce="305f00ef", nc=00000001
Date: Wed, 06 Jul 2016 22:20:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "4693049888" <sip:4693049888@71.244.49.87>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 1814590057 1814590058 IN IP4 71.244.49.87
s=Asterisk PBX 13.9.1
c=IN IP4 71.244.49.87
t=0 0
m=audio 11934 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:192.159.66.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK023dd7bd;rport=5061
From: <sip:4693049888@71.244.49.87:5061>;tag=as16df33b4
To: <sip:9724247977@trunk1.freepbx.com>
Call-ID: 0bde579e147e0a05206d6ae110c883cb@71.244.49.87:5061
CSeq: 103 INVITE
User-Agent: SIPStation 2.11.3
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
       > 0x7f67c2d2c490 -- Probation passed - setting RTP source address to 67.231.13.79:13102

<--- SIP read from UDP:192.159.66.3:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK023dd7bd;rport=5061
From: <sip:4693049888@71.244.49.87:5061>;tag=as16df33b4
To: <sip:9724247977@trunk1.freepbx.com>;tag=B4SKDNcp9Qr6N
Call-ID: 0bde579e147e0a05206d6ae110c883cb@71.244.49.87:5061
CSeq: 103 INVITE
Contact: <sip:9724247977@192.159.66.3:5060;transport=udp>
User-Agent: SIPStation 2.11.3
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 224
Remote-Party-ID: "9724247977" <sip:9724247977@trunk1.freepbx.com>;party=calling;privacy=off;screen=no

v=0
o=Sonus_UAC 911700 764768 IN IP4 67.231.13.113
s=SIP Media Capabilities
c=IN IP4 67.231.13.79
t=0 0
m=audio 13102 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
--- (16 headers 10 lines) ---
sip_route_dump: route/path hop: <sip:9724247977@192.159.66.3:5060;transport=udp>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 67.231.13.79:13102
    -- SIP/fpbx-1-cdB7e8PklPds-000000bd is making progress passing it to SIP/2-000000bc
Audio is at 13666
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 166.173.57.228:55998 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.54.24.68:49672;branch=z9hG4bKPjuxL897wWY-1tNn003zvBL7R-gxmYwMvU;received=166.173.57.228;rport=55998
From: <sip:2@71.244.49.87>;tag=emeyyFIMa0qFK6xHKSvfxAeACoi6NWuk
To: <sip:9724247977@71.244.49.87>;tag=as14e39c07
Call-ID: x7RRuTvHl-IlnwZoKOKDwZahnQaGCTmY
CSeq: 12487 INVITE
Server: FPBX-13.0.151(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:9724247977@71.244.49.87:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 275

v=0
o=root 2044284684 2044284684 IN IP4 71.244.49.87
s=Asterisk PBX 13.9.1
c=IN IP4 71.244.49.87
t=0 0
m=audio 13666 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
       > 0x7f67c2d2c490 -- Probation passed - setting RTP source address to 67.231.13.79:13102
       > 0x6123d00 -- Probation passed - setting RTP source address to 166.173.57.228:57498

<--- SIP read from UDP:192.168.1.170:5061 --->
REGISTER sip:192.168.1.210:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-ca6362b0
From: "Cisco" <sip:6@192.168.1.210>;tag=a97236f356a9bb21o0
To: "Cisco" <sip:6@192.168.1.210>
Call-ID: 2539dd76-dae509b4@192.168.1.170
CSeq: 35640 REGISTER
Max-Forwards: 70
Authorization: Digest username="6",realm="asterisk",nonce="6a180cf7",uri="sip:192.168.1.210:5061",algorithm=MD5,response="fe1b90f6b0c4c6521dbd07015e187283"
Contact: "Cisco" <sip:6@192.168.1.170:5061>;expires=600
User-Agent: Cisco/SPA501G-7.6.1
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.1.170:5061 (NAT)
Sending to 192.168.1.170:5061 (NAT)

<--- Transmitting (no NAT) to 192.168.1.170:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-ca6362b0;received=192.168.1.170
From: "Cisco" <sip:6@192.168.1.210>;tag=a97236f356a9bb21o0
To: "Cisco" <sip:6@192.168.1.210>;tag=as1eed29ce
Call-ID: 2539dd76-dae509b4@192.168.1.170
CSeq: 35640 REGISTER
Server: FPBX-13.0.151(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37b8ac91"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2539dd76-dae509b4@192.168.1.170' in 32000 ms (Method: REGISTER)
[2016-07-06 17:20:38] SECURITY[14222]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="ChallengeSent",EventTV="2016-07-06T17:20:38.616-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="6",SessionID="0x612f558",LocalAddress="IPV4/UDP/192.168.1.210/5061",RemoteAddress="IPV4/UDP/192.168.1.170/5061",Challenge="37b8ac91"

<--- SIP read from UDP:192.168.1.170:5061 --->
REGISTER sip:192.168.1.210:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-a9358f28
From: "Cisco" <sip:6@192.168.1.210>;tag=a97236f356a9bb21o0
To: "Cisco" <sip:6@192.168.1.210>
Call-ID: 2539dd76-dae509b4@192.168.1.170
CSeq: 35641 REGISTER
Max-Forwards: 70
Authorization: Digest username="6",realm="asterisk",nonce="37b8ac91",uri="sip:192.168.1.210:5061",algorithm=MD5,response="a67ee363dd6fee484e9072b6801c99aa"
Contact: "Cisco" <sip:6@192.168.1.170:5061>;expires=600
User-Agent: Cisco/SPA501G-7.6.1
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.1.170:5061 (no NAT)
Reliably Transmitting (no NAT) to 192.168.1.170:5061:
OPTIONS sip:6@192.168.1.170:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK79c4b72f
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.210:5061>;tag=as238364aa
To: <sip:6@192.168.1.170:5061>
Contact: <sip:Unknown@192.168.1.210:5061>
Call-ID: 7a4de959312fe12b4aa60a6f035e1e68@192.168.1.210:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.151(13.9.1)
Date: Wed, 06 Jul 2016 22:20:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.1.170:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-a9358f28;received=192.168.1.170
From: "Cisco" <sip:6@192.168.1.210>;tag=a97236f356a9bb21o0
To: "Cisco" <sip:6@192.168.1.210>;tag=as1eed29ce
Call-ID: 2539dd76-dae509b4@192.168.1.170
CSeq: 35641 REGISTER
Server: FPBX-13.0.151(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: <sip:6@192.168.1.170:5061>;expires=600
Date: Wed, 06 Jul 2016 22:20:38 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2539dd76-dae509b4@192.168.1.170' in 32000 ms (Method: REGISTER)
[2016-07-06 17:20:38] SECURITY[14222]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-07-06T17:20:38.631-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="6",SessionID="0x612f558",LocalAddress="IPV4/UDP/192.168.1.210/5061",RemoteAddress="IPV4/UDP/192.168.1.170/5061",UsingPassword="1"

<--- SIP read from UDP:192.168.1.170:5061 --->
SIP/2.0 200 OK
To: <sip:6@192.168.1.170:5061>;tag=817e58dbf025c259i0
From: "Unknown" <sip:Unknown@192.168.1.210:5061>;tag=as238364aa
Call-ID: 7a4de959312fe12b4aa60a6f035e1e68@192.168.1.210:5061
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.210:5061;branch=z9hG4bK79c4b72f
Server: Cisco/SPA501G-7.6.1
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7a4de959312fe12b4aa60a6f035e1e68@192.168.1.210:5061' Method: OPTIONS
Reliably Transmitting (NAT) to 192.159.66.3:5060:
OPTIONS sip:trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK41ad1836;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@71.244.49.87:5061>;tag=as2f3e2811
To: <sip:trunk1.freepbx.com>
Contact: <sip:Unknown@71.244.49.87:5061>
Call-ID: 085c00ab2dc18d68304353a0362e633c@71.244.49.87:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.151(13.9.1)
Date: Wed, 06 Jul 2016 22:20:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.159.66.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK41ad1836;rport=5061
From: "Unknown" <sip:Unknown@71.244.49.87:5061>;tag=as2f3e2811
To: <sip:trunk1.freepbx.com>;tag=Z146F2gmFg64N
Call-ID: 085c00ab2dc18d68304353a0362e633c@71.244.49.87:5061
CSeq: 102 OPTIONS
Contact: <sip:192.159.66.3>
User-Agent: SIPStation 2.11.3
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '085c00ab2dc18d68304353a0362e633c@71.244.49.87:5061' Method: OPTIONS

<--- SIP read from UDP:192.168.1.6:10000 --->


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

<--- SIP read from UDP:192.168.1.170:5061 --->
NOTIFY sip:192.168.1.210:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-72813477
From: "Cisco" <sip:6@192.168.1.210>;tag=a97236f356a9bb21o0
To: <sip:192.168.1.210>
Call-ID: 5f2a7c5c-9da78ccd@192.168.1.170
CSeq: 200986 NOTIFY
Max-Forwards: 70
Contact: "Cisco" <sip:6@192.168.1.170:5061>
Event: keep-alive
User-Agent: Cisco/SPA501G-7.6.1
Content-Length: 0

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

<--- Transmitting (NAT) to 192.168.1.170:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.170:5061;branch=z9hG4bK-72813477;received=192.168.1.170;rport=5061
From: "Cisco" <sip:6@192.168.1.210>;tag=a97236f356a9bb21o0
To: <sip:192.168.1.210>;tag=as6d638907
Call-ID: 5f2a7c5c-9da78ccd@192.168.1.170
CSeq: 200986 NOTIFY
Server: FPBX-13.0.151(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5f2a7c5c-9da78ccd@192.168.1.170' in 32000 ms (Method: NOTIFY)

<--- SIP read from UDP:192.159.66.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK023dd7bd;rport=5061
From: <sip:4693049888@71.244.49.87:5061>;tag=as16df33b4
To: <sip:9724247977@trunk1.freepbx.com>;tag=B4SKDNcp9Qr6N
Call-ID: 0bde579e147e0a05206d6ae110c883cb@71.244.49.87:5061
CSeq: 103 INVITE
Contact: <sip:9724247977@192.159.66.3:5060;transport=udp>
User-Agent: SIPStation 2.11.3
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 224
Remote-Party-ID: "Outbound Call" <sip:+19724247977@trunk1.freepbx.com>;party=calling;privacy=off;screen=no

v=0
o=Sonus_UAC 911700 764768 IN IP4 67.231.13.113
s=SIP Media Capabilities
c=IN IP4 67.231.13.79
t=0 0
m=audio 13102 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
--- (15 headers 10 lines) ---
sip_route_dump: route/path hop: <sip:9724247977@192.159.66.3:5060;transport=udp>
Transmitting (NAT) to 192.159.66.3:5060:
ACK sip:9724247977@192.159.66.3:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 71.244.49.87:5061;branch=z9hG4bK6d5e9a10;rport
Max-Forwards: 70
From: <sip:4693049888@71.244.49.87:5061>;tag=as16df33b4
To: <sip:9724247977@trunk1.freepbx.com>;tag=B4SKDNcp9Qr6N
Contact: <sip:4693049888@71.244.49.87:5061>
Call-ID: 0bde579e147e0a05206d6ae110c883cb@71.244.49.87:5061
CSeq: 103 ACK
User-Agent: FPBX-13.0.151(13.9.1)
Content-Length: 0


---
    -- SIP/fpbx-1-cdB7e8PklPds-000000bd answered SIP/2-000000bc
Audio is at 13666
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 166.173.57.228:55998 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.54.24.68:49672;branch=z9hG4bKPjuxL897wWY-1tNn003zvBL7R-gxmYwMvU;received=166.173.57.228;rport=55998
From: <sip:2@71.244.49.87>;tag=emeyyFIMa0qFK6xHKSvfxAeACoi6NWuk
To: <sip:9724247977@71.244.49.87>;tag=as14e39c07
Call-ID: x7RRuTvHl-IlnwZoKOKDwZahnQaGCTmY
CSeq: 12487 INVITE
Server: FPBX-13.0.151(13.9.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:9724247977@71.244.49.87:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 275

v=0
o=root 2044284684 2044284684 IN IP4 71.244.49.87
s=Asterisk PBX 13.9.1
c=IN IP4 71.244.49.87
t=0 0
m=audio 13666 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/fpbx-1-cdB7e8PklPds-000000bd joined 'simple_bridge' basic-bridge <386120a1-2077-4b61-8792-e25b4f8d4a56>
    -- Channel SIP/2-000000bc joined 'simple_bridge' basic-bridge <386120a1-2077-4b61-8792-e25b4f8d4a56>

<--- SIP read from UDP:166.173.57.228:55998 --->
ACK sip:9724247977@192.168.1.210:5061 SIP/2.0
Via: SIP/2.0/UDP 10.54.24.68:49672;rport;branch=z9hG4bKPjOcxZ56n2.AZW94BUKzAAutWzXDuL8I2z
Max-Forwards: 70
From: <sip:2@71.244.49.87>;tag=emeyyFIMa0qFK6xHKSvfxAeACoi6NWuk
To: <sip:9724247977@71.244.49.87>;tag=as14e39c07
Call-ID: x7RRuTvHl-IlnwZoKOKDwZahnQaGCTmY
CSeq: 12487 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
localhost*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@localhost ~]#