<--- SIP read from UDP:216.82.224.202:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 173.8.103.157:5060;branch=z9hG4bK6d128790
From: <sip:+16516811421@67.231.8.93>;tag=as3d638546
To: "TAYLOR TELEPHON " <sip:+16512454836@192.168.37.68;isup-oli=62>;tag=gK0e2b34c9
Call-ID: 2114908381_22202251@192.168.37.68
CSeq: 102 BYE
Record-Route: <sip:67.231.8.93:5060;lr=on;ftag=as3d638546>
Record-Route: <sip:216.82.224.202:5060;lr;ftag=as3d638546>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '2114908381_22202251@192.168.37.68' Method: ACK
 '<sip:101@192.168.183.1;transport=UDP>;tag=101b6a18-8fb7a8c0-13c4-5016b5c9-3dd8a0c0-5016b5c9'

<--- SIP read from UDP:216.82.224.202:5060 --->
INVITE sip:+16516811421@173.8.103.157:5060;transport=udp SIP/2.0
Record-Route: <sip:216.82.224.202;lr;ftag=gK0e75e2ad>
Record-Route: <sip:67.231.4.93;lr=on;ftag=gK0e75e2ad>
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK04ec.6083acc6.0
Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bK04ec.2d9aa18.1
Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK0eB37f231489bdacd35
From: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
To: <sip:+16516811421@67.231.4.93>
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 11396 INVITE
Max-Forwards: 68
Contact: <sip:+16512454836@192.168.47.68:5060>
Content-Length: 327
Content-Disposition: session; handling=required
Content-Type: application/sdp
Remote-Party-ID: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68:5060>;privacy=off
P-Asserted-Identity: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68:5060>

v=0
o=Sonus_UAC 17796 5887 IN IP4 192.168.47.68
s=SIP Media Capabilities
c=IN IP4 67.231.4.99
t=0 0
m=audio 10046 RTP/AVP 0 18 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:30
<------------->
--- (17 headers 15 lines) ---
Sending to 216.82.224.202:5060 (NAT)
Using INVITE request as basis request - 621692546_46699090@192.168.47.68
Found peer 'bandwidth' for '+16512454836' from 216.82.224.202:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format iLBC for ID 96
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|g729|ilbc)/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 67.231.4.99:10046
Looking for +16516811421 in from-pstn-e164-us (domain 173.8.103.157)
list_route: hop: <sip:216.82.224.202;lr;ftag=gK0e75e2ad>
list_route: hop: <sip:67.231.4.93;lr=on;ftag=gK0e75e2ad>

<--- Transmitting (no NAT) to 216.82.224.202:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK04ec.6083acc6.0;received=216.82.224.202
Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bK04ec.2d9aa18.1
Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK0eB37f231489bdacd35
Record-Route: <sip:216.82.224.202;lr;ftag=gK0e75e2ad>
Record-Route: <sip:67.231.4.93;lr=on;ftag=gK0e75e2ad>
From: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
To: <sip:+16516811421@67.231.4.93>
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 11396 INVITE
Server: FPBX-2.10.1(10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+16516811421@173.8.103.157:5060>
Content-Length: 0


<------------>
    -- Executing [+16516811421@from-pstn-e164-us:1] Set("SIP/bandwidth-00000003", "CALLERID(number)=6512454836") in new stack
    -- Executing [+16516811421@from-pstn-e164-us:2] Goto("SIP/bandwidth-00000003", "from-pstn,6516811421,1") in new stack
    -- Goto (from-pstn,6516811421,1)
    -- Executing [6516811421@from-pstn:1] Set("SIP/bandwidth-00000003", "__FROM_DID=6516811421") in new stack
    -- Executing [6516811421@from-pstn:2] Gosub("SIP/bandwidth-00000003", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/bandwidth-00000003", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/bandwidth-00000003", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/bandwidth-00000003", "") in new stack
    -- Executing [6516811421@from-pstn:3] Set("SIP/bandwidth-00000003", "CDR(did)=6516811421") in new stack
    -- Executing [6516811421@from-pstn:4] ExecIf("SIP/bandwidth-00000003", "0 ?Set(CALLERID(name)=6512454836)") in new stack
    -- Executing [6516811421@from-pstn:5] Set("SIP/bandwidth-00000003", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [6516811421@from-pstn:6] Set("SIP/bandwidth-00000003", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [6516811421@from-pstn:7] Goto("SIP/bandwidth-00000003", "ext-group,600,1") in new stack
    -- Goto (ext-group,600,1)
    -- Executing [600@ext-group:1] Macro("SIP/bandwidth-00000003", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/bandwidth-00000003", "AMPUSER=6512454836") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/bandwidth-00000003", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/bandwidth-00000003", "1?Set(REALCALLERIDNUM=6512454836)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/bandwidth-00000003", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/bandwidth-00000003", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/bandwidth-00000003", "1?report") in new stack
    -- Goto (macro-user-callerid,s,11)
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/bandwidth-00000003", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:12] Set("SIP/bandwidth-00000003", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:13] GotoIf("SIP/bandwidth-00000003", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,24)
    -- Executing [s@macro-user-callerid:24] Set("SIP/bandwidth-00000003", "CALLERID(number)=6512454836") in new stack
    -- Executing [s@macro-user-callerid:25] Set("SIP/bandwidth-00000003", "CALLERID(name)=TAYLOR TELEPHON ") in new stack
    -- Executing [s@macro-user-callerid:26] Set("SIP/bandwidth-00000003", "CHANNEL(language)=en") in new stack
    -- Executing [600@ext-group:2] Macro("SIP/bandwidth-00000003", "blkvm-setifempty,") in new stack
    -- Executing [s@macro-blkvm-setifempty:1] GotoIf("SIP/bandwidth-00000003", "1?init") in new stack
    -- Goto (macro-blkvm-setifempty,s,4)
    -- Executing [s@macro-blkvm-setifempty:4] Set("SIP/bandwidth-00000003", "__BLKVM_CHANNEL=SIP/bandwidth-00000003") in new stack
    -- Executing [s@macro-blkvm-setifempty:5] Set("SIP/bandwidth-00000003", "SHARED(BLKVM,SIP/bandwidth-00000003)=TRUE") in new stack
    -- Executing [s@macro-blkvm-setifempty:6] Set("SIP/bandwidth-00000003", "GOSUB_RETVAL=TRUE") in new stack
    -- Executing [s@macro-blkvm-setifempty:7] MacroExit("SIP/bandwidth-00000003", "") in new stack
    -- Executing [600@ext-group:3] GotoIf("SIP/bandwidth-00000003", "1?skipov") in new stack
    -- Goto (ext-group,600,6)
    -- Executing [600@ext-group:6] Set("SIP/bandwidth-00000003", "RRNODEST=") in new stack
    -- Executing [600@ext-group:7] Set("SIP/bandwidth-00000003", "__NODEST=600") in new stack
    -- Executing [600@ext-group:8] GosubIf("SIP/bandwidth-00000003", "0?sub-rgsetcid,s,1()") in new stack
    -- Executing [600@ext-group:9] Set("SIP/bandwidth-00000003", "__PICKUPMARK=600") in new stack
    -- Executing [600@ext-group:10] Gosub("SIP/bandwidth-00000003", "sub-record-check,s,1(rg,600,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("SIP/bandwidth-00000003", "1?check") in new stack
    -- Goto (sub-record-check,s,6)
    -- Executing [s@sub-record-check:6] Set("SIP/bandwidth-00000003", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:7] GotoIf("SIP/bandwidth-00000003", "1?next") in new stack
    -- Goto (sub-record-check,s,10)
    -- Executing [s@sub-record-check:10] ExecIf("SIP/bandwidth-00000003", "0?Return()") in new stack
    -- Executing [s@sub-record-check:11] GotoIf("SIP/bandwidth-00000003", "0?rg,1") in new stack
    -- Executing [s@sub-record-check:12] Set("SIP/bandwidth-00000003", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("SIP/bandwidth-00000003", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
    -- Executing [s@sub-record-check:14] Set("SIP/bandwidth-00000003", "NOW=1343666285") in new stack
    -- Executing [s@sub-record-check:15] Set("SIP/bandwidth-00000003", "__DAY=30") in new stack
    -- Executing [s@sub-record-check:16] Set("SIP/bandwidth-00000003", "__MONTH=07") in new stack
    -- Executing [s@sub-record-check:17] Set("SIP/bandwidth-00000003", "__YEAR=2012") in new stack
    -- Executing [s@sub-record-check:18] Set("SIP/bandwidth-00000003", "__TIMESTR=20120730-113805") in new stack
    -- Executing [s@sub-record-check:19] Set("SIP/bandwidth-00000003", "__FROMEXTEN=6512454836") in new stack
    -- Executing [s@sub-record-check:20] Set("SIP/bandwidth-00000003", "__CALLFILENAME=rg-600-6512454836-20120730-113805-1343666285.4") in new stack
    -- Executing [s@sub-record-check:21] Goto("SIP/bandwidth-00000003", "rg,1") in new stack
    -- Goto (sub-record-check,rg,1)
    -- Executing [rg@sub-record-check:1] GosubIf("SIP/bandwidth-00000003", "0?record,1(rg,dontcare,6512454836)") in new stack
    -- Executing [rg@sub-record-check:2] Return("SIP/bandwidth-00000003", "") in new stack
    -- Executing [600@ext-group:11] Set("SIP/bandwidth-00000003", "RingGroupMethod=ringall") in new stack
    -- Executing [600@ext-group:12] Macro("SIP/bandwidth-00000003", "dial,30,tr,101-103-106-107-108-109-110") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/bandwidth-00000003", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/bandwidth-00000003", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Starting New Dialparties.agi
 dialparties.agi: Caller ID name is 'TAYLOR TELEPHON' number is '6512454836'
       > dialparties.agi: USE_CONFIRMATION:  'FALSE'
       > dialparties.agi: RINGGROUP_INDEX:   ''
 dialparties.agi: Methodology of ring is  'ringall'
    -- dialparties.agi: Added extension 101 to extension map
    -- dialparties.agi: Added extension 103 to extension map
    -- dialparties.agi: Added extension 106 to extension map
    -- dialparties.agi: Added extension 107 to extension map
    -- dialparties.agi: Added extension 108 to extension map
    -- dialparties.agi: Added extension 109 to extension map
    -- dialparties.agi: Added extension 110 to extension map
    -- dialparties.agi: Extension 101 cf is disabled
    -- dialparties.agi: Extension 103 cf is disabled
    -- dialparties.agi: Extension 106 cf is disabled
    -- dialparties.agi: Extension 107 cf is disabled
    -- dialparties.agi: Extension 108 cf is disabled
    -- dialparties.agi: Extension 109 cf is disabled
    -- dialparties.agi: Extension 110 cf is disabled
    -- dialparties.agi: Extension 101 do not disturb is disabled
    -- dialparties.agi: Extension 103 do not disturb is disabled
    -- dialparties.agi: Extension 106 do not disturb is disabled
    -- dialparties.agi: Extension 107 do not disturb is disabled
    -- dialparties.agi: Extension 108 do not disturb is disabled
    -- dialparties.agi: Extension 109 do not disturb is disabled
    -- dialparties.agi: Extension 110 do not disturb is disabled
       > dialparties.agi: extnum 101 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/101 to 6512454836
       > dialparties.agi: extnum 103 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/103 to 6512454836
       > dialparties.agi: extnum 106 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/106 to 6512454836
       > dialparties.agi: extnum 107 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/107 to 6512454836
       > dialparties.agi: extnum 108 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/108 to 6512454836
       > dialparties.agi: extnum 109 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/109 to 6512454836
       > dialparties.agi: extnum 110 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/110 to 6512454836
    -- dialparties.agi: Filtered ARG3: 101-103-106-107-108-109-110
       > dialparties.agi: NODEST: 600 adding M(auto-blkvm) to dialopts: trM(auto-blkvm)
       > dialparties.agi: NODEST: 600 blkvm enabled macro already in dialopts: trM(auto-blkvm)
    -- <SIP/bandwidth-00000003>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/bandwidth-00000003", "SIP/101&SIP/103&SIP/106&USTM/107@107&USTM/108@108&USTM/109@109&USTM/109@4002&SIP/110,30,trM(auto-blkvm)") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
[2012-07-30 11:38:05] WARNING[29976][C-00000001]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Using UNISTIM RTP CoS mark 5
[2012-07-30 11:38:05] WARNING[29976][C-00000001]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'USTM' (cause 34 - Circuit/channel congestion)
[2012-07-30 11:38:05] WARNING[29976][C-00000001]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'USTM' (cause 34 - Circuit/channel congestion)
  == Using UNISTIM RTP CoS mark 5
[2012-07-30 11:38:05] WARNING[29976][C-00000001]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
    -- Called SIP/101
    -- Called SIP/103
    -- Called USTM/107@107
    -- Called USTM/109@4002

<--- Transmitting (no NAT) to 216.82.224.202:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK04ec.6083acc6.0;received=216.82.224.202
Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bK04ec.2d9aa18.1
Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK0eB37f231489bdacd35
Record-Route: <sip:216.82.224.202;lr;ftag=gK0e75e2ad>
Record-Route: <sip:67.231.4.93;lr=on;ftag=gK0e75e2ad>
From: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
To: <sip:+16516811421@67.231.4.93>;tag=as6200a592
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 11396 INVITE
Server: FPBX-2.10.1(10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+16516811421@173.8.103.157:5060>
Content-Length: 0


<------------>
    -- USTM/109@4002-0xb63137c8 is ringing

<--- Transmitting (no NAT) to 216.82.224.202:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK04ec.6083acc6.0;received=216.82.224.202
Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bK04ec.2d9aa18.1
Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK0eB37f231489bdacd35
Record-Route: <sip:216.82.224.202;lr;ftag=gK0e75e2ad>
Record-Route: <sip:67.231.4.93;lr=on;ftag=gK0e75e2ad>
From: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
To: <sip:+16516811421@67.231.4.93>;tag=as6200a592
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 11396 INVITE
Server: FPBX-2.10.1(10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+16516811421@173.8.103.157:5060>
Content-Length: 0


<------------>
    -- USTM/107@107-0x89e5c40 is ringing
    -- SIP/103-00000005 connected line has changed. Saving it until answer for SIP/bandwidth-00000003
    -- SIP/101-00000004 connected line has changed. Saving it until answer for SIP/bandwidth-00000003
    -- SIP/103-00000005 is ringing
    -- SIP/101-00000004 is ringing
    -- USTM/109@4002-0xb63137c8 answered SIP/bandwidth-00000003
    -- Executing [s@macro-auto-blkvm:1] Set("USTM/109@4002-0xb63137c8", "__MACRO_RESULT=") in new stack
    -- Executing [s@macro-auto-blkvm:2] Macro("USTM/109@4002-0xb63137c8", "blkvm-clr,") in new stack
    -- Executing [s@macro-blkvm-clr:1] Set("USTM/109@4002-0xb63137c8", "SHARED(BLKVM,SIP/bandwidth-00000003)=") in new stack
    -- Executing [s@macro-blkvm-clr:2] Set("USTM/109@4002-0xb63137c8", "GOSUB_RETVAL=") in new stack
    -- Executing [s@macro-blkvm-clr:3] MacroExit("USTM/109@4002-0xb63137c8", "") in new stack
    -- Executing [s@macro-auto-blkvm:3] ExecIf("USTM/109@4002-0xb63137c8", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=109@4002)") in new stack
    -- Executing [s@macro-auto-blkvm:4] ExecIf("USTM/109@4002-0xb63137c8", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=)") in new stack
Audio is at 13720
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 216.82.224.202:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK04ec.6083acc6.0;received=216.82.224.202
Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bK04ec.2d9aa18.1
Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK0eB37f231489bdacd35
Record-Route: <sip:216.82.224.202;lr;ftag=gK0e75e2ad>
Record-Route: <sip:67.231.4.93;lr=on;ftag=gK0e75e2ad>
From: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
To: <sip:+16516811421@67.231.4.93>;tag=as6200a592
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 11396 INVITE
Server: FPBX-2.10.1(10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+16516811421@173.8.103.157:5060>
Content-Type: application/sdp
Content-Length: 248

v=0
o=root 1495464827 1495464827 IN IP4 173.8.103.157
s=Asterisk PBX SVN-trunk-r370518
c=IN IP4 173.8.103.157
t=0 0
m=audio 13720 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (no NAT) to 216.82.224.202:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK04ec.6083acc6.0;received=216.82.224.202
Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bK04ec.2d9aa18.1
Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK0eB37f231489bdacd35
Record-Route: <sip:216.82.224.202;lr;ftag=gK0e75e2ad>
Record-Route: <sip:67.231.4.93;lr=on;ftag=gK0e75e2ad>
From: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
To: <sip:+16516811421@67.231.4.93>;tag=as6200a592
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 11396 INVITE
Server: FPBX-2.10.1(10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:+16516811421@173.8.103.157:5060>
Content-Type: application/sdp
Content-Length: 248

v=0
o=root 1495464827 1495464827 IN IP4 173.8.103.157
s=Asterisk PBX SVN-trunk-r370518
c=IN IP4 173.8.103.157
t=0 0
m=audio 13720 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:216.82.224.202:5060 --->
ACK sip:+16516811421@173.8.103.157:5060 SIP/2.0
Record-Route: <sip:216.82.224.202;lr;ftag=gK0e75e2ad>
Record-Route: <sip:67.231.4.93;lr=on;ftag=gK0e75e2ad>
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK04ec.6083acc6.2
Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bK04ec.2d9aa18.3
Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK0eB38015dc19bdacd35
From: <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
To: <sip:+16516811421@67.231.4.93>;tag=as6200a592
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 11396 ACK
Max-Forwards: 68
Content-Length: 0

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

<--- SIP read from UDP:216.82.224.202:5060 --->
ACK sip:+16516811421@173.8.103.157:5060 SIP/2.0
Record-Route: <sip:216.82.224.202;lr;ftag=gK0e75e2ad>
Record-Route: <sip:67.231.4.93;lr=on;ftag=gK0e75e2ad>
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK04ec.6083acc6.2
Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bK04ec.2d9aa18.3
Via: SIP/2.0/UDP 192.168.47.68:5060;branch=z9hG4bK0eB38026b5f9bdacd35
From: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
To: <sip:+16516811421@67.231.4.93>;tag=as6200a592
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 11396 ACK
Max-Forwards: 68
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
    -- Executing [h@macro-dial:1] Macro("SIP/bandwidth-00000003", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/bandwidth-00000003", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("SIP/bandwidth-00000003", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] Hangup("SIP/bandwidth-00000003", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/bandwidth-00000003' in macro 'hangupcall'
  == Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/bandwidth-00000003'
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/bandwidth-00000003' in macro 'dial'
  == Spawn extension (ext-group, 600, 12) exited non-zero on 'SIP/bandwidth-00000003'
Scheduling destruction of SIP dialog '621692546_46699090@192.168.47.68' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:216.82.224.202;lr;ftag=gK0e75e2ad> for address/port to send to
set_destination: set destination to 216.82.224.202:5060
Reliably Transmitting (no NAT) to 216.82.224.202:5060:
BYE sip:+16512454836@192.168.47.68:5060 SIP/2.0
Via: SIP/2.0/UDP 173.8.103.157:5060;branch=z9hG4bK4aee9e91
Route: <sip:216.82.224.202;lr;ftag=gK0e75e2ad>,<sip:67.231.4.93;lr=on;ftag=gK0e75e2ad>
Max-Forwards: 70
From: <sip:+16516811421@67.231.4.93>;tag=as6200a592
To: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 102 BYE
User-Agent: FPBX-2.10.1(10.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Retransmitting #1 (no NAT) to 216.82.224.202:5060:
BYE sip:+16512454836@192.168.47.68:5060 SIP/2.0
Via: SIP/2.0/UDP 173.8.103.157:5060;branch=z9hG4bK4aee9e91
Route: <sip:216.82.224.202;lr;ftag=gK0e75e2ad>,<sip:67.231.4.93;lr=on;ftag=gK0e75e2ad>
Max-Forwards: 70
From: <sip:+16516811421@67.231.4.93>;tag=as6200a592
To: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 102 BYE
User-Agent: FPBX-2.10.1(10.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:216.82.224.202:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 173.8.103.157:5060;branch=z9hG4bK4aee9e91
From: <sip:+16516811421@67.231.4.93>;tag=as6200a592
To: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 102 BYE
Record-Route: <sip:67.231.4.93:5060;lr=on;ftag=as6200a592>
Record-Route: <sip:216.82.224.202:5060;lr;ftag=as6200a592>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '621692546_46699090@192.168.47.68' Method: ACK

<--- SIP read from UDP:216.82.224.202:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 173.8.103.157:5060;branch=z9hG4bK4aee9e91
From: <sip:+16516811421@67.231.4.93>;tag=as6200a592
To: "TAYLOR TELEPHON " <sip:+16512454836@192.168.47.68;isup-oli=62>;tag=gK0e75e2ad
Call-ID: 621692546_46699090@192.168.47.68
CSeq: 102 BYE
Record-Route: <sip:67.231.4.93:5060;lr=on;ftag=as6200a592>
Record-Route: <sip:216.82.224.202:5060;lr;ftag=as6200a592>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (no NAT) to 216.82.224.202:5060:
OPTIONS sip:216.82.224.202 SIP/2.0
Via: SIP/2.0/UDP 173.8.103.157:5060;branch=z9hG4bK28a2a0ae
Max-Forwards: 70
From: "Unknown" <sip:Unknown@173.8.103.157>;tag=as385e5f72
To: <sip:216.82.224.202>
Contact: <sip:Unknown@173.8.103.157:5060>
Call-ID: 06209d0b59b012d864d4fb0d7ef630e2@173.8.103.157:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(10.0)
Date: Mon, 30 Jul 2012 16:38:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:216.82.224.202:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 173.8.103.157:5060;branch=z9hG4bK28a2a0ae
From: "Unknown" <sip:Unknown@173.8.103.157>;tag=as385e5f72
To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.a9fa
Call-ID: 06209d0b59b012d864d4fb0d7ef630e2@173.8.103.157:5060
CSeq: 102 OPTIONS
Server: Bandwidth.com TRM (bw7.gold.13)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '06209d0b59b012d864d4fb0d7ef630e2@173.8.103.157:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 216.82.224.202:5060:
OPTIONS sip:216.82.224.202 SIP/2.0
Via: SIP/2.0/UDP 173.8.103.157:5060;branch=z9hG4bK38054838
Max-Forwards: 70
From: "Unknown" <sip:Unknown@173.8.103.157>;tag=as2164a5c0
To: <sip:216.82.224.202>
Contact: <sip:Unknown@173.8.103.157:5060>
Call-ID: 76e5c4995a52cd2529e8cecc6f18d1f3@173.8.103.157:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(10.0)
Date: Mon, 30 Jul 2012 16:38:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:216.82.224.202:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 173.8.103.157:5060;branch=z9hG4bK38054838
From: "Unknown" <sip:Unknown@173.8.103.157>;tag=as2164a5c0
To: <sip:216.82.224.202>;tag=f5da119de3db22dcaa2abb8ea9fec0ce.71e4
Call-ID: 76e5c4995a52cd2529e8cecc6f18d1f3@173.8.103.157:5060
CSeq: 102 OPTIONS
Server: Bandwidth.com TRM (bw7.gold.13)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '76e5c4995a52cd2529e8cecc6f18d1f3@173.8.103.157:5060' Method: OPTIONS