
<--- SIP read from UDP:64.61.93.190:5060 --->
INVITE sip:15618076052@66.29.254.132:5060 SIP/2.0
Record-Route: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKeb76.1aa7cda3.0
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKeb76.a312bdd4.0
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK5d092554;rport=5060
From: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
To: <sip:15618076052@nycinpro01.voicepulse.net>
Contact: <sip:5613389696@64.61.93.170>
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 68
Remote-Party-ID: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;privacy=off;screen=no
Date: Fri, 09 Dec 2011 13:18:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 410

v=0
o=root 25089 25089 IN IP4 64.61.93.170
s=session
c=IN IP4 64.61.93.170
t=0 0
m=audio 12354 RTP/AVP 0 8 3 97 111 5 7 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (18 headers 19 lines) ---
Sending to 64.61.93.190:5060 (no NAT)
Using INVITE request as basis request - 3f3fe48d526819262128243b738bf35a@64.61.93.170
Found peer '1VP-SIPJFKA' for '5613389696' from 64.61.93.190:5060
  == 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 97
Found RTP audio format 111
Found RTP audio format 5
Found RTP audio format 7
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 iLBC for ID 97
Found audio description format G726-32 for ID 111
Found audio description format DVI4 for ID 5
Found audio description format LPC for ID 7
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0xcae (gsm|ulaw|alaw|g726|adpcm|lpc10|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.61.93.170:12354
Looking for 15618076052 in from-trunk (domain 66.29.254.132:5060)
list_route: hop: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>

<--- Transmitting (no NAT) to 64.61.93.190:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKeb76.1aa7cda3.0;received=64.61.93.190
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKeb76.a312bdd4.0
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK5d092554;rport=5060
Record-Route: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
From: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
To: <sip:15618076052@nycinpro01.voicepulse.net>
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:15618076052@66.29.254.132:5060>
Content-Length: 0


<------------>
    -- Executing [15618076052@from-trunk:1] Set("SIP/1VP-SIPJFKA-0000a05d", "__FROM_DID=15618076052") in new stack
    -- Executing [15618076052@from-trunk:2] ExecIf("SIP/1VP-SIPJFKA-0000a05d", "0 ?Set(CALLERID(name)=5613389696)") in new stack
    -- Executing [15618076052@from-trunk:3] Set("SIP/1VP-SIPJFKA-0000a05d", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [15618076052@from-trunk:4] Set("SIP/1VP-SIPJFKA-0000a05d", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [15618076052@from-trunk:5] Goto("SIP/1VP-SIPJFKA-0000a05d", "ivr-4,s,1") in new stack
    -- Goto (ivr-4,s,1)
    -- Executing [s@ivr-4:1] Set("SIP/1VP-SIPJFKA-0000a05d", "MSG=custom/main-greeting") in new stack
    -- Executing [s@ivr-4:2] Set("SIP/1VP-SIPJFKA-0000a05d", "LOOPCOUNT=0") in new stack
    -- Executing [s@ivr-4:3] Set("SIP/1VP-SIPJFKA-0000a05d", "__DIR-CONTEXT=") in new stack
    -- Executing [s@ivr-4:4] Set("SIP/1VP-SIPJFKA-0000a05d", "_IVR_CONTEXT_ivr-4=") in new stack
    -- Executing [s@ivr-4:5] Set("SIP/1VP-SIPJFKA-0000a05d", "_IVR_CONTEXT=ivr-4") in new stack
    -- Executing [s@ivr-4:6] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "0?begin") in new stack
    -- Executing [s@ivr-4:7] Answer("SIP/1VP-SIPJFKA-0000a05d", "") in new stack
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 64.61.93.190:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKeb76.1aa7cda3.0;received=64.61.93.190
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKeb76.a312bdd4.0
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK5d092554;rport=5060
Record-Route: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
From: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
To: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:15618076052@66.29.254.132:5060>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1754475368 1754475368 IN IP4 66.29.254.132
s=Asterisk PBX 1.8.6.0
c=IN IP4 66.29.254.132
t=0 0
m=audio 11922 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 64.61.93.190:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKeb76.1aa7cda3.0;received=64.61.93.190
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKeb76.a312bdd4.0
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK5d092554;rport=5060
Record-Route: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
From: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
To: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:15618076052@66.29.254.132:5060>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1754475368 1754475368 IN IP4 66.29.254.132
s=Asterisk PBX 1.8.6.0
c=IN IP4 66.29.254.132
t=0 0
m=audio 11922 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:64.61.93.190:5060 --->
ACK sip:15618076052@66.29.254.132:5060 SIP/2.0
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKeb76.1aa7cda3.2
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKeb76.a312bdd4.2
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK5c7cc756;rport=5060
From: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
To: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
Contact: <sip:5613389696@64.61.93.170>
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 68
Remote-Party-ID: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;privacy=off;screen=no
Content-Length: 0
VP-hint: loose-routed

<------------->
--- (14 headers 0 lines) ---

<--- SIP read from UDP:64.61.93.190:5060 --->
ACK sip:15618076052@66.29.254.132:5060 SIP/2.0
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKeb76.1aa7cda3.2
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKeb76.a312bdd4.2
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK7f90c385;rport=5060
From: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
To: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
Contact: <sip:5613389696@64.61.93.170>
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 68
Remote-Party-ID: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;privacy=off;screen=no
Content-Length: 0
VP-hint: loose-routed

<------------->
--- (14 headers 0 lines) ---
    -- Executing [s@ivr-4:8] Wait("SIP/1VP-SIPJFKA-0000a05d", "1") in new stack
Really destroying SIP dialog '58ee31c04726534c284ae07b1f9dced0@127.0.0.1' Method: REGISTER
    -- Executing [s@ivr-4:9] Set("SIP/1VP-SIPJFKA-0000a05d", "TIMEOUT(digit)=3") in new stack
    -- Digit timeout set to 3.000
    -- Executing [s@ivr-4:10] Set("SIP/1VP-SIPJFKA-0000a05d", "TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10.000
    -- Executing [s@ivr-4:11] Set("SIP/1VP-SIPJFKA-0000a05d", "__IVR_RETVM=") in new stack
    -- Executing [s@ivr-4:12] ExecIf("SIP/1VP-SIPJFKA-0000a05d", "1?Background(custom/main-greeting)") in new stack
    -- <SIP/1VP-SIPJFKA-0000a05d> Playing 'custom/main-greeting.slin' (language 'en')
Reliably Transmitting (no NAT) to 192.168.16.36:5060:
OPTIONS sip:7522@192.168.16.36:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK4fb2df8f
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.16.231>;tag=as28523052
To: <sip:7522@192.168.16.36:5060>
Contact: <sip:Unknown@192.168.16.231:5060>
Call-ID: 7f35a96273e85cad41c521f55645b4ae@192.168.16.231:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:18:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.36:5060 --->
SIP/2.0 200 OK
To: <sip:7522@192.168.16.36:5060>;tag=c5344bbf726bd4cdi0
From: "Unknown" <sip:Unknown@192.168.16.231>;tag=as28523052
Call-ID: 7f35a96273e85cad41c521f55645b4ae@192.168.16.231:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK4fb2df8f
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7f35a96273e85cad41c521f55645b4ae@192.168.16.231:5060' Method: OPTIONS
    -- Executing [1@ivr-4:1] Macro("SIP/1VP-SIPJFKA-0000a05d", "blkvm-clr,") in new stack
    -- Executing [s@macro-blkvm-clr:1] Set("SIP/1VP-SIPJFKA-0000a05d", "SHARED(BLKVM,)=") in new stack
    -- Executing [s@macro-blkvm-clr:2] Set("SIP/1VP-SIPJFKA-0000a05d", "GOSUB_RETVAL=") in new stack
    -- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/1VP-SIPJFKA-0000a05d", "") in new stack
    -- Executing [1@ivr-4:2] Set("SIP/1VP-SIPJFKA-0000a05d", "__NODEST=") in new stack
    -- Executing [1@ivr-4:3] Goto("SIP/1VP-SIPJFKA-0000a05d", "ext-group,601,1") in new stack
    -- Goto (ext-group,601,1)
    -- Executing [601@ext-group:1] Macro("SIP/1VP-SIPJFKA-0000a05d", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/1VP-SIPJFKA-0000a05d", "AMPUSER=5613389696") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1VP-SIPJFKA-0000a05d", "1?Set(REALCALLERIDNUM=5613389696)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/1VP-SIPJFKA-0000a05d", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/1VP-SIPJFKA-0000a05d", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?report") in new stack
    -- Goto (macro-user-callerid,s,12)
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:13] Set("SIP/1VP-SIPJFKA-0000a05d", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,25)
    -- Executing [s@macro-user-callerid:25] Set("SIP/1VP-SIPJFKA-0000a05d", "CALLERID(number)=5613389696") in new stack
    -- Executing [s@macro-user-callerid:26] Set("SIP/1VP-SIPJFKA-0000a05d", "CALLERID(name)=BYTE SOLUTIONS ") in new stack
    -- Executing [s@macro-user-callerid:27] Set("SIP/1VP-SIPJFKA-0000a05d", "CHANNEL(language)=en") in new stack
    -- Executing [601@ext-group:2] Macro("SIP/1VP-SIPJFKA-0000a05d", "blkvm-setifempty,") in new stack
    -- Executing [s@macro-blkvm-setifempty:1] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?init") in new stack
    -- Goto (macro-blkvm-setifempty,s,4)
    -- Executing [s@macro-blkvm-setifempty:4] Set("SIP/1VP-SIPJFKA-0000a05d", "__BLKVM_CHANNEL=SIP/1VP-SIPJFKA-0000a05d") in new stack
    -- Executing [s@macro-blkvm-setifempty:5] Set("SIP/1VP-SIPJFKA-0000a05d", "SHARED(BLKVM,SIP/1VP-SIPJFKA-0000a05d)=TRUE") in new stack
    -- Executing [s@macro-blkvm-setifempty:6] Set("SIP/1VP-SIPJFKA-0000a05d", "GOSUB_RETVAL=TRUE") in new stack
    -- Executing [s@macro-blkvm-setifempty:7] MacroExit("SIP/1VP-SIPJFKA-0000a05d", "") in new stack
    -- Executing [601@ext-group:3] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?skipov") in new stack
    -- Goto (ext-group,601,6)
    -- Executing [601@ext-group:6] Set("SIP/1VP-SIPJFKA-0000a05d", "RRNODEST=") in new stack
    -- Executing [601@ext-group:7] Set("SIP/1VP-SIPJFKA-0000a05d", "__NODEST=601") in new stack
    -- Executing [601@ext-group:8] GosubIf("SIP/1VP-SIPJFKA-0000a05d", "0?sub-rgsetcid,s,1") in new stack
    -- Executing [601@ext-group:9] Macro("SIP/1VP-SIPJFKA-0000a05d", "prepend-cid,AS:") in new stack
    -- Executing [s@macro-prepend-cid:1] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?REPCID") in new stack
    -- Goto (macro-prepend-cid,s,5)
    -- Executing [s@macro-prepend-cid:5] Set("SIP/1VP-SIPJFKA-0000a05d", "_RGPREFIX=AS:") in new stack
    -- Executing [s@macro-prepend-cid:6] Set("SIP/1VP-SIPJFKA-0000a05d", "CALLERID(name)=AS:BYTE SOLUTIONS") in new stack
    -- Executing [601@ext-group:10] Set("SIP/1VP-SIPJFKA-0000a05d", "_CFIGNORE=TRUE") in new stack
    -- Executing [601@ext-group:11] Set("SIP/1VP-SIPJFKA-0000a05d", "_FORWARD_CONTEXT=block-cf") in new stack
    -- Executing [601@ext-group:12] Set("SIP/1VP-SIPJFKA-0000a05d", "RecordMethod=Group") in new stack
    -- Executing [601@ext-group:13] Macro("SIP/1VP-SIPJFKA-0000a05d", "record-enable,7600-7535-7540-7518-7595-7527-7405-7528-7607-7604-7602-7999,Group") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/1VP-SIPJFKA-0000a05d", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,6)
    -- Executing [s@macro-record-enable:6] Set("SIP/1VP-SIPJFKA-0000a05d", "LOOPCNT=12") in new stack
    -- Executing [s@macro-record-enable:7] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=1") in new stack
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=2") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?begin") in new stack
    -- Goto (macro-record-enable,s,8)
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=3") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?begin") in new stack
    -- Goto (macro-record-enable,s,8)
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=4") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?begin") in new stack
    -- Goto (macro-record-enable,s,8)
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=5") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?begin") in new stack
    -- Goto (macro-record-enable,s,8)
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=6") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?begin") in new stack
    -- Goto (macro-record-enable,s,8)
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=7") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?begin") in new stack
    -- Goto (macro-record-enable,s,8)
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=8") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?begin") in new stack
    -- Goto (macro-record-enable,s,8)
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=9") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?begin") in new stack
    -- Goto (macro-record-enable,s,8)
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=10") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?begin") in new stack
    -- Goto (macro-record-enable,s,8)
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=11") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?begin") in new stack
    -- Goto (macro-record-enable,s,8)
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=12") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?begin") in new stack
    -- Goto (macro-record-enable,s,8)
    -- Executing [s@macro-record-enable:8] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?continue") in new stack
    -- Goto (macro-record-enable,s,12)
    -- Executing [s@macro-record-enable:12] Set("SIP/1VP-SIPJFKA-0000a05d", "ITER=13") in new stack
    -- Executing [s@macro-record-enable:13] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "0?begin") in new stack
    -- Executing [s@macro-record-enable:14] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "0?IN") in new stack
    -- Executing [s@macro-record-enable:15] ExecIf("SIP/1VP-SIPJFKA-0000a05d", "1?MacroExit()") in new stack
    -- Executing [601@ext-group:14] Set("SIP/1VP-SIPJFKA-0000a05d", "RingGroupMethod=ringall") in new stack
    -- Executing [601@ext-group:15] Macro("SIP/1VP-SIPJFKA-0000a05d", "dial,20,tr,7600-7535-7540-7518-7595-7527-7405-7528-7607-7604-7602-7999") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/1VP-SIPJFKA-0000a05d", "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 'AS:BYTE SOLUTIONS' number is '5613389696'
       > dialparties.agi: USE_CONFIRMATION:  'FALSE'
       > dialparties.agi: RINGGROUP_INDEX:   ''
 dialparties.agi: Methodology of ring is  'ringall'
    -- dialparties.agi: Added extension 7600 to extension map
    -- dialparties.agi: Added extension 7535 to extension map
    -- dialparties.agi: Added extension 7540 to extension map
    -- dialparties.agi: Added extension 7518 to extension map
    -- dialparties.agi: Added extension 7595 to extension map
    -- dialparties.agi: Added extension 7527 to extension map
    -- dialparties.agi: Added extension 7405 to extension map
    -- dialparties.agi: Added extension 7528 to extension map
    -- dialparties.agi: Added extension 7607 to extension map
    -- dialparties.agi: Added extension 7604 to extension map
    -- dialparties.agi: Added extension 7602 to extension map
    -- dialparties.agi: Added extension 7999 to extension map
    -- dialparties.agi: Extension 7600 cf is disabled
    -- dialparties.agi: Extension 7535 cf is disabled
    -- dialparties.agi: Extension 7540 cf is disabled
    -- dialparties.agi: Extension 7518 cf is disabled
    -- dialparties.agi: Extension 7595 cf is disabled
    -- dialparties.agi: Extension 7527 cf is disabled
    -- dialparties.agi: Extension 7405 cf is disabled
    -- dialparties.agi: Extension 7528 cf is disabled
    -- dialparties.agi: Extension 7607 cf is disabled
    -- dialparties.agi: Extension 7604 cf is disabled
    -- dialparties.agi: Extension 7602 cf is disabled
    -- dialparties.agi: Extension 7999 cf is disabled
    -- dialparties.agi: Extension 7600 do not disturb is disabled
    -- dialparties.agi: Extension 7535 do not disturb is disabled
    -- dialparties.agi: Extension 7540 do not disturb is disabled
    -- dialparties.agi: Extension 7518 do not disturb is disabled
    -- dialparties.agi: Extension 7595 do not disturb is disabled
    -- dialparties.agi: Extension 7527 do not disturb is disabled
    -- dialparties.agi: Extension 7405 do not disturb is disabled
    -- dialparties.agi: Extension 7528 do not disturb is disabled
    -- dialparties.agi: Extension 7607 do not disturb is disabled
    -- dialparties.agi: Extension 7604 do not disturb is disabled
    -- dialparties.agi: Extension 7602 do not disturb is disabled
    -- dialparties.agi: Extension 7999 do not disturb is disabled
       > dialparties.agi: extnum 7600 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/7600 to 5613389696
       > dialparties.agi: extnum 7535 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/7535 to 5613389696
       > dialparties.agi: extnum 7540 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/7540 to 5613389696
       > dialparties.agi: extnum 7518 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/7518 to 5613389696
       > dialparties.agi: extnum 7595 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
Reliably Transmitting (no NAT) to 192.168.16.58:5060:
OPTIONS sip:7902@192.168.16.58:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK0d8a8c0f
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.16.231>;tag=as3ba5780c
To: <sip:7902@192.168.16.58:5060>
Contact: <sip:Unknown@192.168.16.231:5060>
Call-ID: 41b5123854c7100d6d63041a567154a3@192.168.16.231:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
    -- dialparties.agi: dbset CALLTRACE/7595 to 5613389696
       > dialparties.agi: extnum 7527 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/7527 to 5613389696
       > dialparties.agi: extnum 7405 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/7405 to 5613389696
       > dialparties.agi: extnum 7528 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/7528 to 5613389696
       > dialparties.agi: extnum 7607 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/7607 to 5613389696
       > dialparties.agi: extnum 7604 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/7604 to 5613389696
       > dialparties.agi: extnum 7602 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/7602 to 5613389696
       > dialparties.agi: extnum 7999 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
    -- dialparties.agi: dbset CALLTRACE/7999 to 5613389696
    -- dialparties.agi: Filtered ARG3: 7600-7535-7540-7518-7595-7527-7405-7528-7607-7604-7602-7999
       > dialparties.agi: NODEST: 601 adding M(auto-blkvm) to dialopts: trM(auto-blkvm)
       > dialparties.agi: NODEST: 601 blkvm enabled macro already in dialopts: trM(auto-blkvm)
    -- <SIP/1VP-SIPJFKA-0000a05d>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/1VP-SIPJFKA-0000a05d", "SIP/7600&SIP/7535&SIP/7540&SIP/7518&SIP/7595&SIP/7527&SIP/7405&SIP/7528&SIP/7607&SIP/7604&SIP/7602&SIP/7999,20,trM(auto-blkvm)") 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 192.168.16.12:5060:
INVITE sip:7600@192.168.16.12:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK75fcaa2f
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as0043d69a
To: <sip:7600@192.168.16.12:5060>
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 0ffa186c62ac5bee3b6ee8913c33a5b5@192.168.16.231:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 2128275450 2128275450 IN IP4 192.168.16.231
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.16.231
t=0 0
m=audio 14500 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/7600
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5

<--- SIP read from UDP:192.168.16.58:5060 --->
SIP/2.0 200 OK
To: <sip:7902@192.168.16.58:5060>;tag=5f8a585868bb03a2i0
From: "Unknown" <sip:Unknown@192.168.16.231>;tag=as3ba5780c
Call-ID: 41b5123854c7100d6d63041a567154a3@192.168.16.231:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK0d8a8c0f
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '41b5123854c7100d6d63041a567154a3@192.168.16.231:5060' Method: OPTIONS
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 192.168.16.18:5060:
INVITE sip:7535@192.168.16.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK488e8b9b
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as4e74c6e3
To: <sip:7535@192.168.16.18:5060>
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 3130aba62002b0fc4965a927085465dc@192.168.16.231:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1657333477 1657333477 IN IP4 192.168.16.231
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.16.231
t=0 0
m=audio 18578 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/7535
  == 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 192.168.16.19:5060:
INVITE sip:7540@192.168.16.19:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK572087ba
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as29ae4279
To: <sip:7540@192.168.16.19:5060>
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 767ad07267dfd0eb00efbbb662939f32@192.168.16.231:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 251962358 251962358 IN IP4 192.168.16.231
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.16.231
t=0 0
m=audio 10634 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/7540
  == 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 192.168.16.22:5060:
INVITE sip:7518@192.168.16.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK07749c5a
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as05a99507
To: <sip:7518@192.168.16.22:5060>
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 11272d3c0efbf2960ce80ca47e06fd5d@192.168.16.231:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1169605281 1169605281 IN IP4 192.168.16.231
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.16.231
t=0 0
m=audio 16932 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/7518
  == 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 192.168.16.24:5060:
INVITE sip:7595@192.168.16.24:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK599e868e
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as1011922d
To: <sip:7595@192.168.16.24:5060>
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 48256b63443879587368104d2f3c8183@192.168.16.231:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 276352001 276352001 IN IP4 192.168.16.231
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.16.231
t=0 0
m=audio 19812 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/7595
  == 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 192.168.16.28:5060:
INVITE sip:7527@192.168.16.28:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK67a38bcf
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as5965c2f4
To: <sip:7527@192.168.16.28:5060>
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 1fd8966476e1a16f38e2f00c78569160@192.168.16.231:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 187459501 187459501 IN IP4 192.168.16.231
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.16.231
t=0 0
m=audio 17734 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/7527
  == 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 192.168.16.29:5060:
INVITE sip:7405@192.168.16.29:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK7199859d
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as72548d41
To: <sip:7405@192.168.16.29:5060>
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 71755e777e060e817ef387ce174adf76@192.168.16.231:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1223223407 1223223407 IN IP4 192.168.16.231
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.16.231
t=0 0
m=audio 19018 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/7405
  == 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 192.168.16.44:5060:
INVITE sip:7528@192.168.16.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK4ad5dbe3
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as775835c2
To: <sip:7528@192.168.16.44:5060>
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 4fb2c1867c19799f3b668d5767f09c5e@192.168.16.231:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1577372816 1577372816 IN IP4 192.168.16.231
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.16.231
t=0 0
m=audio 11998 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/7528
  == 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 192.168.16.47:5060:
INVITE sip:7607@192.168.16.47:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK57e138fb
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as55b24378
To: <sip:7607@192.168.16.47:5060>
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 242edea57a30873c292fd3c4500119ea@192.168.16.231:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 270195740 270195740 IN IP4 192.168.16.231
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.16.231
t=0 0
m=audio 17706 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/7607
  == 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 192.168.16.55:5060:
INVITE sip:7604@192.168.16.55:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK06d1c07d
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as3712d4ca
To: <sip:7604@192.168.16.55:5060>
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 23319ca852fada281231e27578618fc7@192.168.16.231:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1278044313 1278044313 IN IP4 192.168.16.231
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.16.231
t=0 0
m=audio 17742 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/7604
Really destroying SIP dialog '2810749a3d6548b907251d5172455609@127.0.0.1:0' Method: INVITE
  == 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 173.12.99.113:61211:
INVITE sip:7999@173.12.99.113:61211 SIP/2.0
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK38a63b48
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@66.29.254.132>;tag=as154ef6e0
To: <sip:7999@173.12.99.113:61211>
Contact: <sip:5613389696@66.29.254.132:5060>
Call-ID: 3d2410507cef09b350aac2ee231c4231@66.29.254.132:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1916633901 1916633901 IN IP4 66.29.254.132
s=Asterisk PBX 1.8.6.0
c=IN IP4 66.29.254.132
t=0 0
m=audio 18466 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/7999
    -- SIP/7600-0000a05e connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d
    -- SIP/7535-0000a05f connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d
    -- SIP/7540-0000a060 connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d
    -- SIP/7518-0000a061 connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d
    -- SIP/7595-0000a062 connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d
    -- SIP/7527-0000a063 connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d
    -- SIP/7405-0000a064 connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d
    -- SIP/7528-0000a065 connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d
    -- SIP/7607-0000a066 connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d
    -- SIP/7604-0000a067 connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d
    -- SIP/7999-0000a068 connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d

<--- SIP read from UDP:192.168.16.19:5060 --->
SIP/2.0 100 Trying
To: <sip:7540@192.168.16.19:5060>
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as29ae4279
Call-ID: 767ad07267dfd0eb00efbbb662939f32@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK572087ba
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.16.29:5060 --->
SIP/2.0 100 Trying
To: <sip:7405@192.168.16.29:5060>
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as72548d41
Call-ID: 71755e777e060e817ef387ce174adf76@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK7199859d
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.16.12:5060 --->
SIP/2.0 100 Trying
To: <sip:7600@192.168.16.12:5060>
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as0043d69a
Call-ID: 0ffa186c62ac5bee3b6ee8913c33a5b5@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK75fcaa2f
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.16.24:5060 --->
SIP/2.0 100 Trying
To: <sip:7595@192.168.16.24:5060>
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as1011922d
Call-ID: 48256b63443879587368104d2f3c8183@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK599e868e
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.16.18:5060 --->
SIP/2.0 100 Trying
To: <sip:7535@192.168.16.18:5060>
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as4e74c6e3
Call-ID: 3130aba62002b0fc4965a927085465dc@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK488e8b9b
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.16.44:5060 --->
SIP/2.0 100 Trying
To: <sip:7528@192.168.16.44:5060>
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as775835c2
Call-ID: 4fb2c1867c19799f3b668d5767f09c5e@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK4ad5dbe3
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.16.55:5060 --->
SIP/2.0 100 Trying
To: <sip:7604@192.168.16.55:5060>
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as3712d4ca
Call-ID: 23319ca852fada281231e27578618fc7@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK06d1c07d
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.16.47:5060 --->
SIP/2.0 100 Trying
To: <sip:7607@192.168.16.47:5060>
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as55b24378
Call-ID: 242edea57a30873c292fd3c4500119ea@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK57e138fb
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.16.28:5060 --->
SIP/2.0 100 Trying
To: <sip:7527@192.168.16.28:5060>
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as5965c2f4
Call-ID: 1fd8966476e1a16f38e2f00c78569160@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK67a38bcf
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.16.19:5060 --->
SIP/2.0 302 Moved Temporarily
To: <sip:7540@192.168.16.19:5060>;tag=e458ec456861421ci0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as29ae4279
Call-ID: 767ad07267dfd0eb00efbbb662939f32@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK572087ba
Contact: <sip:915617031114@192.168.16.231>
Diversion: "7540" <sip:7540@192.168.16.231>;reason=unconditional
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Got SIP response 302 "Moved Temporarily" back from 192.168.16.19:5060
RDNIS for this call is 7540 (reason unconditional)
Transmitting (no NAT) to 192.168.16.19:5060:
ACK sip:7540@192.168.16.19:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK572087ba
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as29ae4279
To: <sip:7540@192.168.16.19:5060>;tag=e458ec456861421ci0
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 767ad07267dfd0eb00efbbb662939f32@192.168.16.231:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---
    -- Now forwarding SIP/1VP-SIPJFKA-0000a05d to 'Local/915617031114@block-cf' (thanks to SIP/7540-0000a060)

<--- SIP read from UDP:192.168.16.29:5060 --->
SIP/2.0 302 Moved Temporarily
To: <sip:7405@192.168.16.29:5060>;tag=b980073826b627b9i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as72548d41
Call-ID: 71755e777e060e817ef387ce174adf76@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK7199859d
Contact: <sip:7518@192.168.16.231>
Diversion: "7405" <sip:7405@192.168.16.231>;reason=unconditional
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Got SIP response 302 "Moved Temporarily" back from 192.168.16.29:5060
RDNIS for this call is 7405 (reason unconditional)
Transmitting (no NAT) to 192.168.16.29:5060:
ACK sip:7405@192.168.16.29:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK7199859d
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as72548d41
To: <sip:7405@192.168.16.29:5060>;tag=b980073826b627b9i0
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 71755e777e060e817ef387ce174adf76@192.168.16.231:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.18:5060 --->
SIP/2.0 302 Moved Temporarily
To: <sip:7535@192.168.16.18:5060>;tag=23256420bf7230b8i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as4e74c6e3
Call-ID: 3130aba62002b0fc4965a927085465dc@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK488e8b9b
Contact: <sip:915618666690@192.168.16.231>
Diversion: "7535" <sip:7535@192.168.16.231>;reason=unconditional
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Got SIP response 302 "Moved Temporarily" back from 192.168.16.18:5060
RDNIS for this call is 7535 (reason unconditional)
Transmitting (no NAT) to 192.168.16.18:5060:
ACK sip:7535@192.168.16.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK488e8b9b
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as4e74c6e3
To: <sip:7535@192.168.16.18:5060>;tag=23256420bf7230b8i0
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 3130aba62002b0fc4965a927085465dc@192.168.16.231:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---
Really destroying SIP dialog '767ad07267dfd0eb00efbbb662939f32@192.168.16.231:5060' Method: INVITE
    -- Now forwarding SIP/1VP-SIPJFKA-0000a05d to 'Local/915618666690@block-cf' (thanks to SIP/7535-0000a05f)
    -- Now forwarding SIP/1VP-SIPJFKA-0000a05d to 'Local/7518@block-cf' (thanks to SIP/7405-0000a064)

<--- SIP read from UDP:192.168.16.22:5060 --->
SIP/2.0 100 Trying
To: <sip:7518@192.168.16.22:5060>
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as05a99507
Call-ID: 11272d3c0efbf2960ce80ca47e06fd5d@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK07749c5a
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '3130aba62002b0fc4965a927085465dc@192.168.16.231:5060' Method: INVITE
Really destroying SIP dialog '71755e777e060e817ef387ce174adf76@192.168.16.231:5060' Method: INVITE

<--- SIP read from UDP:192.168.16.12:5060 --->
SIP/2.0 180 Ringing
To: <sip:7600@192.168.16.12:5060>;tag=e2ae6387bc48deb1i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as0043d69a
Call-ID: 0ffa186c62ac5bee3b6ee8913c33a5b5@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK75fcaa2f
Contact: "7600" <sip:7600@192.168.16.12:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/7600-0000a05e is ringing

<--- SIP read from UDP:192.168.16.44:5060 --->
SIP/2.0 180 Ringing
To: <sip:7528@192.168.16.44:5060>;tag=7b37acd4de3534c8i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as775835c2
Call-ID: 4fb2c1867c19799f3b668d5767f09c5e@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK4ad5dbe3
Contact: "7528" <sip:7528@192.168.16.44:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/7528-0000a065 is ringing

<--- SIP read from UDP:192.168.16.55:5060 --->
SIP/2.0 180 Ringing
To: <sip:7604@192.168.16.55:5060>;tag=b19f68f6b709775di0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as3712d4ca
Call-ID: 23319ca852fada281231e27578618fc7@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK06d1c07d
Contact: "7604" <sip:7604@192.168.16.55:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/7604-0000a067 is ringing

<--- SIP read from UDP:192.168.16.47:5060 --->
SIP/2.0 180 Ringing
To: <sip:7607@192.168.16.47:5060>;tag=e35fb6b514d5df94i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as55b24378
Call-ID: 242edea57a30873c292fd3c4500119ea@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK57e138fb
Contact: "7607" <sip:7607@192.168.16.47:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/7607-0000a066 is ringing

<--- SIP read from UDP:192.168.16.24:5060 --->
SIP/2.0 180 Ringing
To: <sip:7595@192.168.16.24:5060>;tag=c73818c223efac3ai0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as1011922d
Call-ID: 48256b63443879587368104d2f3c8183@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK599e868e
Contact: "7595" <sip:7595@192.168.16.24:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/7595-0000a062 is ringing

<--- SIP read from UDP:192.168.16.28:5060 --->
SIP/2.0 180 Ringing
To: <sip:7527@192.168.16.28:5060>;tag=b4b4f8cd81dea291i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as5965c2f4
Call-ID: 1fd8966476e1a16f38e2f00c78569160@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK67a38bcf
Contact: "7527" <sip:7527@192.168.16.28:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/7527-0000a063 is ringing
Retransmitting #1 (no NAT) to 173.12.99.113:61211:
INVITE sip:7999@173.12.99.113:61211 SIP/2.0
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK38a63b48
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@66.29.254.132>;tag=as154ef6e0
To: <sip:7999@173.12.99.113:61211>
Contact: <sip:5613389696@66.29.254.132:5060>
Call-ID: 3d2410507cef09b350aac2ee231c4231@66.29.254.132:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1916633901 1916633901 IN IP4 66.29.254.132
s=Asterisk PBX 1.8.6.0
c=IN IP4 66.29.254.132
t=0 0
m=audio 18466 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:173.12.99.113:61211 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.29.254.132:5060;received=66.29.254.132;branch=z9hG4bK38a63b48
Call-ID: 3d2410507cef09b350aac2ee231c4231@66.29.254.132:5060
From: "AS:BYTE SOLUTIONS" <sip:5613389696@66.29.254.132>;tag=as154ef6e0
To: <sip:7999@173.12.99.113>;tag=SHu8.aBkXIl8yzK-8h9Xyz2pSmBjb7RE
CSeq: 102 INVITE
Contact: "Gary Herbstman" <sip:7999@173.12.99.113:61211>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/7999-0000a068 is ringing

<--- SIP read from UDP:192.168.16.22:5060 --->
SIP/2.0 180 Ringing
To: <sip:7518@192.168.16.22:5060>;tag=c3658eb1661b6fc9i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as05a99507
Call-ID: 11272d3c0efbf2960ce80ca47e06fd5d@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK07749c5a
Contact: "7518" <sip:7518@192.168.16.22:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/7518-0000a061 is ringing
Reliably Transmitting (no NAT) to 209.31.18.12:5060:
OPTIONS sip:jfk-backup.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK52625f63
Max-Forwards: 70
From: "Unknown" <sip:Unknown@66.29.254.132>;tag=as3340d719
To: <sip:jfk-backup.voicepulse.com>
Contact: <sip:Unknown@66.29.254.132:5060>
Call-ID: 72cf47f778659e3f376d8be058b75985@66.29.254.132:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:209.31.18.12:5060 --->
SIP/2.0 200 OK to keepalive
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK52625f63;rport=5060
From: "Unknown" <sip:Unknown@66.29.254.132>;tag=as3340d719
To: <sip:jfk-backup.voicepulse.com>;tag=bf91ba8e754533baedb51a458caf3219.7440
Call-ID: 72cf47f778659e3f376d8be058b75985@66.29.254.132:5060
CSeq: 102 OPTIONS
Server: OpenSIPS (1.6.2-notls (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '72cf47f778659e3f376d8be058b75985@66.29.254.132:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.16.34:5060:
OPTIONS sip:7606@192.168.16.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK641c7c8f
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.16.231>;tag=as53c65003
To: <sip:7606@192.168.16.34:5060>
Contact: <sip:Unknown@192.168.16.231:5060>
Call-ID: 2c9034241f63085d5a90fcd13afae272@192.168.16.231:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.34:5060 --->
SIP/2.0 200 OK
To: <sip:7606@192.168.16.34:5060>;tag=fc60ad385ddbe7di0
From: "Unknown" <sip:Unknown@192.168.16.231>;tag=as53c65003
Call-ID: 2c9034241f63085d5a90fcd13afae272@192.168.16.231:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK641c7c8f
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2c9034241f63085d5a90fcd13afae272@192.168.16.231:5060' Method: OPTIONS

<--- SIP read from UDP:173.12.99.113:61211 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.29.254.132:5060;received=66.29.254.132;branch=z9hG4bK38a63b48
Call-ID: 3d2410507cef09b350aac2ee231c4231@66.29.254.132:5060
From: "AS:BYTE SOLUTIONS" <sip:5613389696@66.29.254.132>;tag=as154ef6e0
To: <sip:7999@173.12.99.113>;tag=SHu8.aBkXIl8yzK-8h9Xyz2pSmBjb7RE
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: "Gary Herbstman" <sip:7999@173.12.99.113:61211>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 209

v=0
o=- 3532425542 3532425543 IN IP4 173.12.99.113
s=cpc_med
c=IN IP4 173.12.99.113
t=0 0
m=audio 4000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (11 headers 10 lines) ---
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 - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 173.12.99.113:4000
list_route: hop: <sip:7999@173.12.99.113:61211>
set_destination: Parsing <sip:7999@173.12.99.113:61211> for address/port to send to
set_destination: set destination to 173.12.99.113:61211
Transmitting (no NAT) to 173.12.99.113:61211:
ACK sip:7999@173.12.99.113:61211 SIP/2.0
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK58945659
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@66.29.254.132>;tag=as154ef6e0
To: <sip:7999@173.12.99.113:61211>;tag=SHu8.aBkXIl8yzK-8h9Xyz2pSmBjb7RE
Contact: <sip:5613389696@66.29.254.132:5060>
Call-ID: 3d2410507cef09b350aac2ee231c4231@66.29.254.132:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---
    -- SIP/7999-0000a068 connected line has changed. Saving it until answer for SIP/1VP-SIPJFKA-0000a05d
    -- SIP/7999-0000a068 answered SIP/1VP-SIPJFKA-0000a05d
set_destination: Parsing <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA> for address/port to send to
set_destination: set destination to 64.61.93.190:5060
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.61.93.190:5060:
INVITE sip:5613389696@64.61.93.170 SIP/2.0
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK5f119bdf
Route: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
Max-Forwards: 70
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Contact: <sip:15618076052@66.29.254.132:5060>
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "device" <sip:7999@64.61.93.170>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1754475368 1754475369 IN IP4 66.29.254.132
s=Asterisk PBX 1.8.6.0
c=IN IP4 66.29.254.132
t=0 0
m=audio 11922 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

---
Scheduling destruction of SIP dialog '23319ca852fada281231e27578618fc7@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.16.55:5060:
CANCEL sip:7604@192.168.16.55:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK06d1c07d
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as3712d4ca
To: <sip:7604@192.168.16.55:5060>
Call-ID: 23319ca852fada281231e27578618fc7@192.168.16.231:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.6.0)
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '23319ca852fada281231e27578618fc7@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '242edea57a30873c292fd3c4500119ea@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.16.47:5060:
CANCEL sip:7607@192.168.16.47:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK57e138fb
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as55b24378
To: <sip:7607@192.168.16.47:5060>
Call-ID: 242edea57a30873c292fd3c4500119ea@192.168.16.231:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.6.0)
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '242edea57a30873c292fd3c4500119ea@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '4fb2c1867c19799f3b668d5767f09c5e@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.16.44:5060:
CANCEL sip:7528@192.168.16.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK4ad5dbe3
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as775835c2
To: <sip:7528@192.168.16.44:5060>
Call-ID: 4fb2c1867c19799f3b668d5767f09c5e@192.168.16.231:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.6.0)
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '4fb2c1867c19799f3b668d5767f09c5e@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '1fd8966476e1a16f38e2f00c78569160@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.16.28:5060:
CANCEL sip:7527@192.168.16.28:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK67a38bcf
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as5965c2f4
To: <sip:7527@192.168.16.28:5060>
Call-ID: 1fd8966476e1a16f38e2f00c78569160@192.168.16.231:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.6.0)
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '1fd8966476e1a16f38e2f00c78569160@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '48256b63443879587368104d2f3c8183@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.16.24:5060:
CANCEL sip:7595@192.168.16.24:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK599e868e
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as1011922d
To: <sip:7595@192.168.16.24:5060>
Call-ID: 48256b63443879587368104d2f3c8183@192.168.16.231:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.6.0)
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '48256b63443879587368104d2f3c8183@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '11272d3c0efbf2960ce80ca47e06fd5d@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.16.22:5060:
CANCEL sip:7518@192.168.16.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK07749c5a
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as05a99507
To: <sip:7518@192.168.16.22:5060>
Call-ID: 11272d3c0efbf2960ce80ca47e06fd5d@192.168.16.231:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.6.0)
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '11272d3c0efbf2960ce80ca47e06fd5d@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '0ffa186c62ac5bee3b6ee8913c33a5b5@192.168.16.231:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.16.12:5060:
CANCEL sip:7600@192.168.16.12:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK75fcaa2f
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as0043d69a
To: <sip:7600@192.168.16.12:5060>
Call-ID: 0ffa186c62ac5bee3b6ee8913c33a5b5@192.168.16.231:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.6.0)
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '0ffa186c62ac5bee3b6ee8913c33a5b5@192.168.16.231:5060' in 6400 ms (Method: INVITE)
    -- Executing [s@macro-auto-blkvm:1] Set("SIP/7999-0000a068", "__MACRO_RESULT=") in new stack
    -- Executing [s@macro-auto-blkvm:2] Macro("SIP/7999-0000a068", "blkvm-clr,") in new stack
    -- Executing [s@macro-blkvm-clr:1] Set("SIP/7999-0000a068", "SHARED(BLKVM,SIP/1VP-SIPJFKA-0000a05d)=") in new stack
    -- Executing [s@macro-blkvm-clr:2] Set("SIP/7999-0000a068", "GOSUB_RETVAL=") in new stack
    -- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/7999-0000a068", "") in new stack
    -- Executing [s@macro-auto-blkvm:3] Set("SIP/7999-0000a068", "MASTER_CHANNEL(CONNECTEDLINE(num))=7999") in new stack
    -- Executing [s@macro-auto-blkvm:4] Set("SIP/7999-0000a068", "MASTER_CHANNEL(CONNECTEDLINE(name))=Gary Test Phone") in new stack

<--- SIP read from UDP:192.168.16.55:5060 --->
SIP/2.0 487 Request Terminated
To: <sip:7604@192.168.16.55:5060>;tag=b19f68f6b709775di0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as3712d4ca
Call-ID: 23319ca852fada281231e27578618fc7@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK06d1c07d
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.16.55:5060:
ACK sip:7604@192.168.16.55:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK06d1c07d
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as3712d4ca
To: <sip:7604@192.168.16.55:5060>;tag=b19f68f6b709775di0
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 23319ca852fada281231e27578618fc7@192.168.16.231:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.47:5060 --->
SIP/2.0 487 Request Terminated
To: <sip:7607@192.168.16.47:5060>;tag=e35fb6b514d5df94i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as55b24378
Call-ID: 242edea57a30873c292fd3c4500119ea@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK57e138fb
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.16.47:5060:
ACK sip:7607@192.168.16.47:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK57e138fb
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as55b24378
To: <sip:7607@192.168.16.47:5060>;tag=e35fb6b514d5df94i0
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 242edea57a30873c292fd3c4500119ea@192.168.16.231:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.44:5060 --->
SIP/2.0 487 Request Terminated
To: <sip:7528@192.168.16.44:5060>;tag=7b37acd4de3534c8i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as775835c2
Call-ID: 4fb2c1867c19799f3b668d5767f09c5e@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK4ad5dbe3
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.16.44:5060:
ACK sip:7528@192.168.16.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK4ad5dbe3
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as775835c2
To: <sip:7528@192.168.16.44:5060>;tag=7b37acd4de3534c8i0
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 4fb2c1867c19799f3b668d5767f09c5e@192.168.16.231:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.28:5060 --->
SIP/2.0 487 Request Terminated
To: <sip:7527@192.168.16.28:5060>;tag=b4b4f8cd81dea291i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as5965c2f4
Call-ID: 1fd8966476e1a16f38e2f00c78569160@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK67a38bcf
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.16.28:5060:
ACK sip:7527@192.168.16.28:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK67a38bcf
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as5965c2f4
To: <sip:7527@192.168.16.28:5060>;tag=b4b4f8cd81dea291i0
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 1fd8966476e1a16f38e2f00c78569160@192.168.16.231:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.12:5060 --->
SIP/2.0 487 Request Terminated
To: <sip:7600@192.168.16.12:5060>;tag=e2ae6387bc48deb1i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as0043d69a
Call-ID: 0ffa186c62ac5bee3b6ee8913c33a5b5@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK75fcaa2f
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.16.12:5060:
ACK sip:7600@192.168.16.12:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK75fcaa2f
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as0043d69a
To: <sip:7600@192.168.16.12:5060>;tag=e2ae6387bc48deb1i0
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 0ffa186c62ac5bee3b6ee8913c33a5b5@192.168.16.231:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.55:5060 --->
SIP/2.0 200 OK
To: <sip:7604@192.168.16.55:5060>;tag=b19f68f6b709775di0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as3712d4ca
Call-ID: 23319ca852fada281231e27578618fc7@192.168.16.231:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK06d1c07d
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '23319ca852fada281231e27578618fc7@192.168.16.231:5060' Method: INVITE

<--- SIP read from UDP:192.168.16.24:5060 --->
SIP/2.0 487 Request Terminated
To: <sip:7595@192.168.16.24:5060>;tag=c73818c223efac3ai0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as1011922d
Call-ID: 48256b63443879587368104d2f3c8183@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK599e868e
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.16.24:5060:
ACK sip:7595@192.168.16.24:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK599e868e
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as1011922d
To: <sip:7595@192.168.16.24:5060>;tag=c73818c223efac3ai0
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 48256b63443879587368104d2f3c8183@192.168.16.231:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.47:5060 --->
SIP/2.0 200 OK
To: <sip:7607@192.168.16.47:5060>;tag=e35fb6b514d5df94i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as55b24378
Call-ID: 242edea57a30873c292fd3c4500119ea@192.168.16.231:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK57e138fb
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '242edea57a30873c292fd3c4500119ea@192.168.16.231:5060' Method: INVITE

<--- SIP read from UDP:192.168.16.22:5060 --->
SIP/2.0 487 Request Terminated
To: <sip:7518@192.168.16.22:5060>;tag=c3658eb1661b6fc9i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as05a99507
Call-ID: 11272d3c0efbf2960ce80ca47e06fd5d@192.168.16.231:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK07749c5a
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.16.22:5060:
ACK sip:7518@192.168.16.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK07749c5a
Max-Forwards: 70
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as05a99507
To: <sip:7518@192.168.16.22:5060>;tag=c3658eb1661b6fc9i0
Contact: <sip:5613389696@192.168.16.231:5060>
Call-ID: 11272d3c0efbf2960ce80ca47e06fd5d@192.168.16.231:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.44:5060 --->
SIP/2.0 200 OK
To: <sip:7528@192.168.16.44:5060>;tag=7b37acd4de3534c8i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as775835c2
Call-ID: 4fb2c1867c19799f3b668d5767f09c5e@192.168.16.231:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK4ad5dbe3
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4fb2c1867c19799f3b668d5767f09c5e@192.168.16.231:5060' Method: INVITE

<--- SIP read from UDP:192.168.16.28:5060 --->
SIP/2.0 200 OK
To: <sip:7527@192.168.16.28:5060>;tag=b4b4f8cd81dea291i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as5965c2f4
Call-ID: 1fd8966476e1a16f38e2f00c78569160@192.168.16.231:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK67a38bcf
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1fd8966476e1a16f38e2f00c78569160@192.168.16.231:5060' Method: INVITE

<--- SIP read from UDP:192.168.16.12:5060 --->
SIP/2.0 200 OK
To: <sip:7600@192.168.16.12:5060>;tag=e2ae6387bc48deb1i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as0043d69a
Call-ID: 0ffa186c62ac5bee3b6ee8913c33a5b5@192.168.16.231:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK75fcaa2f
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '0ffa186c62ac5bee3b6ee8913c33a5b5@192.168.16.231:5060' Method: INVITE

<--- SIP read from UDP:192.168.16.24:5060 --->
SIP/2.0 200 OK
To: <sip:7595@192.168.16.24:5060>;tag=c73818c223efac3ai0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as1011922d
Call-ID: 48256b63443879587368104d2f3c8183@192.168.16.231:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK599e868e
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '48256b63443879587368104d2f3c8183@192.168.16.231:5060' Method: INVITE

<--- SIP read from UDP:192.168.16.22:5060 --->
SIP/2.0 200 OK
To: <sip:7518@192.168.16.22:5060>;tag=c3658eb1661b6fc9i0
From: "AS:BYTE SOLUTIONS" <sip:5613389696@192.168.16.231>;tag=as05a99507
Call-ID: 11272d3c0efbf2960ce80ca47e06fd5d@192.168.16.231:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK07749c5a
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '11272d3c0efbf2960ce80ca47e06fd5d@192.168.16.231:5060' Method: INVITE

<--- SIP read from UDP:64.61.93.190:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK5f119bdf;rport=5060
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 102 INVITE
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:64.61.93.190:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.29.254.132:5060;rport=5060;branch=z9hG4bK5f119bdf
Record-Route: <sip:64.61.93.190;lr=on;ftag=as6d75ab70>
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5613389696@64.61.93.170:5060>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 25089 25090 IN IP4 64.61.93.170
s=session
c=IN IP4 64.61.93.170
t=0 0
m=audio 12354 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
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 - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.61.93.170:12354
set_destination: Parsing <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA> for address/port to send to
set_destination: set destination to 64.61.93.190:5060
Transmitting (no NAT) to 64.61.93.190:5060:
ACK sip:5613389696@64.61.93.170:5060 SIP/2.0
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK26e05599
Route: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
Max-Forwards: 70
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Contact: <sip:15618076052@66.29.254.132:5060>
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---
set_destination: Parsing <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA> for address/port to send to
set_destination: set destination to 64.61.93.190:5060
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.61.93.190:5060:
INVITE sip:5613389696@64.61.93.170:5060 SIP/2.0
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK2315a247
Route: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
Max-Forwards: 70
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Contact: <sip:15618076052@66.29.254.132:5060>
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 103 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Remote-Party-ID: "Gary Test Phone" <sip:7999@64.61.93.170>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1754475368 1754475370 IN IP4 66.29.254.132
s=Asterisk PBX 1.8.6.0
c=IN IP4 66.29.254.132
t=0 0
m=audio 11922 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:64.61.93.190:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK2315a247;rport=5060
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 103 INVITE
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:64.61.93.190:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 66.29.254.132:5060;rport=5060;branch=z9hG4bK2315a247
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

<------------->
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA> for address/port to send to
set_destination: set destination to 64.61.93.190:5060
Transmitting (no NAT) to 64.61.93.190:5060:
ACK sip:5613389696@64.61.93.170:5060 SIP/2.0
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK2315a247
Route: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
Max-Forwards: 70
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Contact: <sip:15618076052@66.29.254.132:5060>
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 103 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---
set_destination: Parsing <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA> for address/port to send to
set_destination: set destination to 64.61.93.190:5060
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.61.93.190:5060:
INVITE sip:5613389696@64.61.93.170:5060 SIP/2.0
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK7ea0c0d8
Route: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
Max-Forwards: 70
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Contact: <sip:15618076052@66.29.254.132:5060>
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 104 INVITE
User-Agent: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Remote-Party-ID: "Gary Test Phone" <sip:7999@64.61.93.170>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1754475368 1754475371 IN IP4 66.29.254.132
s=Asterisk PBX 1.8.6.0
c=IN IP4 66.29.254.132
t=0 0
m=audio 11922 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:64.61.93.190:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK7ea0c0d8;rport=5060
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 104 INVITE
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:64.61.93.190:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.29.254.132:5060;rport=5060;branch=z9hG4bK7ea0c0d8
Record-Route: <sip:64.61.93.190;lr=on;ftag=as6d75ab70>
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5613389696@64.61.93.170:5060>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 25089 25091 IN IP4 64.61.93.170
s=session
c=IN IP4 64.61.93.170
t=0 0
m=audio 12354 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
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 - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.61.93.170:12354
set_destination: Parsing <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA> for address/port to send to
set_destination: set destination to 64.61.93.190:5060
Transmitting (no NAT) to 64.61.93.190:5060:
ACK sip:5613389696@64.61.93.170:5060 SIP/2.0
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK048c3dc5
Route: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
Max-Forwards: 70
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Contact: <sip:15618076052@66.29.254.132:5060>
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 104 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---
Reliably Transmitting (no NAT) to 192.168.16.17:5060:
OPTIONS sip:7565@192.168.16.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK2e5fa933
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.16.231>;tag=as3843c412
To: <sip:7565@192.168.16.17:5060>
Contact: <sip:Unknown@192.168.16.231:5060>
Call-ID: 5842bfb7468616436876ebad35d0df41@192.168.16.231:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.17:5060 --->
SIP/2.0 200 OK
To: <sip:7565@192.168.16.17:5060>;tag=a5eed23477ef790ai0
From: "Unknown" <sip:Unknown@192.168.16.231>;tag=as3843c412
Call-ID: 5842bfb7468616436876ebad35d0df41@192.168.16.231:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK2e5fa933
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5842bfb7468616436876ebad35d0df41@192.168.16.231:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.16.44:5060:
OPTIONS sip:7528@192.168.16.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK465141f1
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.16.231>;tag=as1b79a3e0
To: <sip:7528@192.168.16.44:5060>
Contact: <sip:Unknown@192.168.16.231:5060>
Call-ID: 1f72dd94120e4e997df0725303f3ad82@192.168.16.231:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Fri, 09 Dec 2011 13:19:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.16.44:5060 --->
SIP/2.0 200 OK
To: <sip:7528@192.168.16.44:5060>;tag=a01f936df242813ai0
From: "Unknown" <sip:Unknown@192.168.16.231>;tag=as1b79a3e0
Call-ID: 1f72dd94120e4e997df0725303f3ad82@192.168.16.231:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.16.231:5060;branch=z9hG4bK465141f1
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1f72dd94120e4e997df0725303f3ad82@192.168.16.231:5060' Method: OPTIONS
Really destroying SIP dialog '7ef9a9380a783a1e07d0a1633601bd3c@192.168.16.231:5060' Method: OPTIONS

<--- SIP read from UDP:64.61.93.190:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 66.29.254.132:5060;rport=5060;branch=z9hG4bK2315a247
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16

<------------->
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA> for address/port to send to
set_destination: set destination to 64.61.93.190:5060
Transmitting (no NAT) to 64.61.93.190:5060:
ACK sip:5613389696@64.61.93.170:5060 SIP/2.0
Via: SIP/2.0/UDP 66.29.254.132:5060;branch=z9hG4bK048c3dc5
Route: <sip:64.61.93.190;lr=on;ftag=as2e813a8a;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
Max-Forwards: 70
From: <sip:15618076052@nycinpro01.voicepulse.net>;tag=as6d75ab70
To: "BYTE SOLUTIONS " <sip:5613389696@64.61.93.170>;tag=as2e813a8a
Contact: <sip:15618076052@66.29.254.132:5060>
Call-ID: 3f3fe48d526819262128243b738bf35a@64.61.93.170
CSeq: 103 ACK
User-Agent: FPBX-2.9.0(1.8.6.0)
Content-Length: 0


---

Really destroying SIP dialog '0232bb791bf0c8d96028054f58074108@192.168.16.231:5060' Method: OPTIONS
Really destroying SIP dialog 'nBw2uQBAam0LORrX5bPggjksgaE.9PP.' Method: REGISTER
Really destroying SIP dialog 'EX4aJveCYhCpakZYZACZj49hQaBxBO5D' Method: REGISTER
pbx*CLI> sip set debug off
SIP Debugging Disabled
    -- Executing [h@macro-dial:1] Macro("SIP/1VP-SIPJFKA-0000a05d", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1VP-SIPJFKA-0000a05d", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] Hangup("SIP/1VP-SIPJFKA-0000a05d", "") in new stack
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/1VP-SIPJFKA-0000a05d' in macro 'hangupcall'
  == Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/1VP-SIPJFKA-0000a05d'
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/1VP-SIPJFKA-0000a05d' in macro 'dial'
  == Spawn extension (ext-group, 601, 15) exited non-zero on 'SIP/1VP-SIPJFKA-0000a05d'
       > doing dnsmgr_lookup for 'jfk-primary.voicepulse.com'
       > doing dnsmgr_lookup for 'atlanta.voip.ms'
       > doing dnsmgr_lookup for 'atlanta.voip.ms'
       > doing dnsmgr_lookup for 'jfk-backup.voicepulse.com'
       > doing dnsmgr_lookup for 'jfk-backup.voicepulse.com'
pbx*CLI> core set verbose 0
Verbosity is now OFF
pbx*CLI>