<--- SIP read from UDP:192.168.1.5:12592 --->
INVITE sip:299@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-2e5158118100bf63-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:201@192.168.1.5:12592>
To: "299"<sip:299@192.168.1.111>
From: <sip:201@192.168.1.111>;tag=11788805
Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 214

v=0
o=- 1 2 IN IP4 192.168.1.5
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.5
t=0 0
m=audio 26162 RTP/AVP 107 0 8 3 101
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 10 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
Sending to 192.168.1.5 : 12592 (no NAT)
Using INVITE request as basis request - MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.
Found peer '201' for '201' from 192.168.1.5:12592

<--- Reliably Transmitting (NAT) to 192.168.1.5:12592 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-2e5158118100bf63-1---d8754z-;received=192.168.1.5;rport=12592
From: <sip:201@192.168.1.111>;tag=11788805
To: "299"<sip:299@192.168.1.111>;tag=as5637521e
Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7998902f"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.' in 6656 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.5:12592 --->
ACK sip:299@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-2e5158118100bf63-1---d8754z-;rport
To: "299"<sip:299@192.168.1.111>;tag=as5637521e
From: <sip:201@192.168.1.111>;tag=11788805
Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.5:12592 --->
INVITE sip:299@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-1134f10557266c2e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:201@192.168.1.5:12592>
To: "299"<sip:299@192.168.1.111>
From: <sip:201@192.168.1.111>;tag=11788805
Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username="201",realm="asterisk",nonce="7998902f",uri="sip:299@192.168.1.111",response="5f04108bc1f9ac781bb4a15f210781f7",algorithm=MD5
Content-Length: 214

v=0
o=- 1 2 IN IP4 192.168.1.5
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.5
t=0 0
m=audio 26162 RTP/AVP 107 0 8 3 101
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (13 headers 10 lines) ---
Sending to 192.168.1.5 : 12592 (NAT)
Using INVITE request as basis request - MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.
Found peer '201' for '201' from 192.168.1.5:12592
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100a (gsm|alaw|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xa (gsm|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.5:26162
Looking for 299 in from-internal (domain 192.168.1.111)
list_route: hop: <sip:201@192.168.1.5:12592>

<--- Transmitting (NAT) to 192.168.1.5:12592 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-1134f10557266c2e-1---d8754z-;received=192.168.1.5;rport=12592
From: <sip:201@192.168.1.111>;tag=11788805
To: "299"<sip:299@192.168.1.111>
Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:299@192.168.1.111>
Content-Length: 0


<------------>
    -- Executing [299@from-internal:1] Macro("SIP/201-0000000e", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/201-0000000e", "AMPUSER=201") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/201-0000000e", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/201-0000000e", "1?Set(REALCALLERIDNUM=201)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/201-0000000e", "AMPUSER=201") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/201-0000000e", "AMPUSERCIDNAME=201") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/201-0000000e", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/201-0000000e", "AMPUSERCID=201") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/201-0000000e", "CALLERID(all)="201" <201>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/201-0000000e", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/201-0000000e", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/201-0000000e", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/201-0000000e", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/201-0000000e", "Using CallerID "201" <201>") in new stack
    -- Executing [299@from-internal:2] Answer("SIP/201-0000000e", "") in new stack
Audio is at 192.168.1.111 port 10320
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.1.5:12592 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-1134f10557266c2e-1---d8754z-;received=192.168.1.5;rport=12592
From: <sip:201@192.168.1.111>;tag=11788805
To: "299"<sip:299@192.168.1.111>;tag=as303ba4df
Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:299@192.168.1.111>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 658433285 658433285 IN IP4 192.168.1.111
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.1.111
t=0 0
m=audio 10320 RTP/AVP 8 3 101
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

<------------>
    -- Executing [299@from-internal:3] Set("SIP/201-0000000e", "__BLKVM_OVERRIDE=BLKVM/299/SIP/201-0000000e") in new stack
    -- Executing [299@from-internal:4] Set("SIP/201-0000000e", "__BLKVM_BASE=299") in new stack
    -- Executing [299@from-internal:5] Set("SIP/201-0000000e", "DB(BLKVM/299/SIP/201-0000000e)=TRUE") in new stack
    -- Executing [299@from-internal:6] ExecIf("SIP/201-0000000e", "1?Set(_DIAL_OPTIONS=tkM(auto-blkvm))") in new stack
    -- Executing [299@from-internal:7] Set("SIP/201-0000000e", "__NODEST=299") in new stack
    -- Executing [299@from-internal:8] Set("SIP/201-0000000e", "MONITOR_FILENAME=/var/spool/asterisk/monitor/q299-20100819-144755-1282222075.19") in new stack
    -- Executing [299@from-internal:9] Queue("SIP/201-0000000e", "299,t,,") in new stack
    -- Started music on hold, class 'default', on SIP/201-0000000e
    -- Executing [202@from-queue:1] Set("Local/202@from-queue-af93;2", "QAGENT=202") in new stack
    -- Executing [202@from-queue:2] Goto("Local/202@from-queue-af93;2", "299,1") in new stack
    -- Goto (from-queue,299,1)
    -- Executing [299@from-queue:1] Goto("Local/202@from-queue-af93;2", "from-internal,202,1") in new stack
    -- Goto (from-internal,202,1)
    -- Executing [202@from-internal:1] Macro("Local/202@from-queue-af93;2", "exten-vm,novm,202") in new stack
    -- Executing [s@macro-exten-vm:1] GotoIf("Local/202@from-queue-af93;2", "0?ext-intercom,nointercom202,1") in new stack
    -- Executing [s@macro-exten-vm:2] GotoIf("Local/202@from-queue-af93;2", "0?ext-intercom,nointercom202,1") in new stack
    -- Executing [s@macro-exten-vm:3] Macro("Local/202@from-queue-af93;2", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("Local/202@from-queue-af93;2", "AMPUSER=201") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("Local/202@from-queue-af93;2", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
    -- Executing [s@macro-user-callerid:10] GotoIf("Local/202@from-queue-af93;2", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("Local/202@from-queue-af93;2", "__TTL=63") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("Local/202@from-queue-af93;2", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("Local/202@from-queue-af93;2", "Using CallerID "201" <201>") in new stack
    -- Executing [s@macro-exten-vm:4] Set("Local/202@from-queue-af93;2", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:5] Set("Local/202@from-queue-af93;2", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:6] Set("Local/202@from-queue-af93;2", "EXTTOCALL=202") in new stack
    -- Executing [s@macro-exten-vm:7] Set("Local/202@from-queue-af93;2", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:8] Set("Local/202@from-queue-af93;2", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:9] Set("Local/202@from-queue-af93;2", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:10] Macro("Local/202@from-queue-af93;2", "record-enable,202,IN") in new stack
    -- Executing [s@macro-record-enable:1] MacroExit("Local/202@from-queue-af93;2", "") in new stack
    -- Executing [s@macro-exten-vm:11] Macro("Local/202@from-queue-af93;2", "dial,,tkM(auto-blkvm),202") in new stack
    -- Executing [s@macro-dial:1] GotoIf("Local/202@from-queue-af93;2", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("Local/202@from-queue-af93;2", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
    -- Executing [203@from-queue:1] Set("Local/203@from-queue-9f48;2", "QAGENT=203") in new stack
    -- Executing [203@from-queue:2] Goto("Local/203@from-queue-9f48;2", "299,1") in new stack
    -- Goto (from-queue,299,1)
    -- Executing [299@from-queue:1] Goto("Local/203@from-queue-9f48;2", "from-internal,203,1") in new stack
    -- Goto (from-internal,203,1)
    -- Executing [203@from-internal:1] Macro("Local/203@from-queue-9f48;2", "exten-vm,novm,203") in new stack

<--- SIP read from UDP:192.168.1.5:12592 --->
ACK sip:299@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:12592;branch=z9hG4bK-d8754z-cf21da15e1698211-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:201@192.168.1.5:12592>
To: "299"<sip:299@192.168.1.111>;tag=as303ba4df
From: <sip:201@192.168.1.111>;tag=11788805
Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.
CSeq: 2 ACK
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username="201",realm="asterisk",nonce="7998902f",uri="sip:299@192.168.1.111",response="5f04108bc1f9ac781bb4a15f210781f7",algorithm=MD5
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
    -- Executing [s@macro-exten-vm:1] GotoIf("Local/203@from-queue-9f48;2", "0?ext-intercom,nointercom203,1") in new stack
    -- Executing [s@macro-exten-vm:2] GotoIf("Local/203@from-queue-9f48;2", "0?ext-intercom,nointercom203,1") in new stack
    -- Executing [s@macro-exten-vm:3] Macro("Local/203@from-queue-9f48;2", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("Local/203@from-queue-9f48;2", "AMPUSER=201") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("Local/203@from-queue-9f48;2", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
    -- Executing [s@macro-user-callerid:10] GotoIf("Local/203@from-queue-9f48;2", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("Local/203@from-queue-9f48;2", "__TTL=63") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("Local/203@from-queue-9f48;2", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("Local/203@from-queue-9f48;2", "Using CallerID "201" <201>") in new stack
    -- Executing [s@macro-exten-vm:4] Set("Local/203@from-queue-9f48;2", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:5] Set("Local/203@from-queue-9f48;2", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:6] Set("Local/203@from-queue-9f48;2", "EXTTOCALL=203") in new stack
    -- Executing [s@macro-exten-vm:7] Set("Local/203@from-queue-9f48;2", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:8] Set("Local/203@from-queue-9f48;2", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:9] Set("Local/203@from-queue-9f48;2", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:10] Macro("Local/203@from-queue-9f48;2", "record-enable,203,IN") in new stack
    -- Executing [s@macro-record-enable:1] MacroExit("Local/203@from-queue-9f48;2", "") in new stack
    -- Executing [s@macro-exten-vm:11] Macro("Local/203@from-queue-9f48;2", "dial,,tkM(auto-blkvm),203") in new stack
    -- Executing [s@macro-dial:1] GotoIf("Local/203@from-queue-9f48;2", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("Local/203@from-queue-9f48;2", "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 '201' number is '201'
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 202 to extension map
    -- dialparties.agi: Extension 202 cf is disabled
    -- dialparties.agi: Extension 202 do not disturb is disabled
 dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
 dialparties.agi: Extension 202 has ExtensionState: 0
    -- dialparties.agi: Checking CW and CFB status for extension 202
    -- dialparties.agi: dbset CALLTRACE/202 to 201
    -- dialparties.agi: Filtered ARG3: 202
    -- <Local/202@from-queue-af93;2>AGI Script dialparties.agi completed, returning 0
 dialparties.agi: Starting New Dialparties.agi
    -- Executing [s@macro-dial:7] Dial("Local/202@from-queue-af93;2", "SIP/202,,tkM(auto-blkvm)") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
Audio is at 192.168.1.111 port 10426
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.52:5060:
INVITE sip:202@192.168.1.52:5060;line=budunjf2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK3e5aa209;rport
Max-Forwards: 70
From: "201" <sip:201@192.168.1.111>;tag=as0ec9c5cf
To: <sip:202@192.168.1.52:5060;line=budunjf2>
Contact: <sip:201@192.168.1.111>
Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Thu, 19 Aug 2010 12:47:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1941640983 1941640983 IN IP4 192.168.1.111
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.1.111
t=0 0
m=audio 10426 RTP/AVP 8 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called 202
 dialparties.agi: Caller ID name is '201' number is '201'
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 203 to extension map
    -- dialparties.agi: Extension 203 cf is disabled
    -- dialparties.agi: Extension 203 do not disturb is disabled
 dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
 dialparties.agi: Extension 203 has ExtensionState: 0
    -- dialparties.agi: Checking CW and CFB status for extension 203
    -- dialparties.agi: dbset CALLTRACE/203 to 201
    -- dialparties.agi: Filtered ARG3: 203
    -- <Local/203@from-queue-9f48;2>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("Local/203@from-queue-9f48;2", "SIP/203,,tkM(auto-blkvm)") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
Audio is at 192.168.1.111 port 10220
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.213:5060:
NVITE sip:203@192.168.1.213:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport
Max-Forwards: 70
From: "201" <sip:201@192.168.1.111>;tag=as2330a1d9
To: <sip:203@192.168.1.213:5060;transport=udp>
Contact: <sip:201@192.168.1.111>
Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Thu, 19 Aug 2010 12:47:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1910113364 1910113364 IN IP4 192.168.1.111
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.1.111
t=0 0
m=audio 10220 RTP/AVP 8 3 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called 203

<--- SIP read from UDP:192.168.1.52:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK3e5aa209;rport=5060
From: "201" <sip:201@192.168.1.111>;tag=as0ec9c5cf
To: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=nvbtn81cu6
Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111
CSeq: 102 INVITE
Contact: <sip:202@192.168.1.52:5060;line=budunjf2>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- SIP/202-0000000f is ringing
    -- Local/202@from-queue-af93;1 is ringing

<--- SIP read from UDP:192.168.1.213:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport
From: "201" <sip:201@192.168.1.111>;tag=as2330a1d9
To: <sip:203@192.168.1.213:5060;transport=udp>
Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.2.2.19
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.213:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport
From: "201" <sip:201@192.168.1.111>;tag=as2330a1d9
To: <sip:203@192.168.1.213:5060;transport=udp>;tag=e516915c3d248474
Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.2.2.19
Contact: <sip:203@192.168.1.213:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- SIP/203-00000010 is ringing
    -- Local/203@from-queue-9f48;1 is ringing

<--- SIP read from UDP:192.168.1.52:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK3e5aa209;rport=5060
From: "201" <sip:201@192.168.1.111>;tag=as0ec9c5cf
To: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=nvbtn81cu6
Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111
CSeq: 102 INVITE
Contact: <sip:202@192.168.1.52:5060;line=budunjf2>;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- SIP/202-0000000f is ringing

<--- SIP read from UDP:192.168.1.52:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK3e5aa209;rport=5060
From: "201" <sip:201@192.168.1.111>;tag=as0ec9c5cf
To: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=nvbtn81cu6
Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111
CSeq: 102 INVITE
Contact: <sip:202@192.168.1.52:5060;line=budunjf2>;reg-id=1
User-Agent: snom300/7.3.30
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1230806085 1230806086 IN IP4 192.168.1.52
s=call
c=IN IP4 192.168.1.52
t=0 0
m=audio 51580 RTP/AVP 8 3 9 101
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:9 g722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 9
Found RTP audio format 101
Found audio description format pcma for ID 8
Found audio description format gsm for ID 3
Found audio description format g722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100a (gsm|alaw|g722), peer - audio=0x100a (gsm|alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100a (gsm|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.52:51580
list_route: hop: <sip:202@192.168.1.52:5060;line=budunjf2>
set_destination: Parsing <sip:202@192.168.1.52:5060;line=budunjf2> for address/port to send to
set_destination: set destination to 192.168.1.52, port 5060
Transmitting (NAT) to 192.168.1.52:5060:
ACK sip:202@192.168.1.52:5060;line=budunjf2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK475078ee;rport
Max-Forwards: 70
From: "201" <sip:201@192.168.1.111>;tag=as0ec9c5cf
To: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=nvbtn81cu6
Contact: <sip:201@192.168.1.111>
Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0


---
    -- SIP/202-0000000f answered Local/202@from-queue-af93;2
    -- Executing [s@macro-auto-blkvm:1] Set("SIP/202-0000000f", "__MACRO_RESULT=") in new stack
    -- Executing [s@macro-auto-blkvm:2] NoOp("SIP/202-0000000f", "Deleting: BLKVM/299/SIP/201-0000000e TRUE") in new stack
    -- Local/202@from-queue-af93;1 answered SIP/201-0000000e
Scheduling destruction of SIP dialog '3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.1.213:5060:
CANCEL sip:203@192.168.1.213:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport
Max-Forwards: 70
From: "201" <sip:201@192.168.1.111>;tag=as2330a1d9
To: <sip:203@192.168.1.213:5060;transport=udp>
Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0


---
Scheduling destruction of SIP dialog '3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111' in 6400 ms (Method: INVITE)
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'Local/203@from-queue-9f48;2' in macro 'dial'
  == Spawn extension (macro-exten-vm, s, 11) exited non-zero on 'Local/203@from-queue-9f48;2' in macro 'exten-vm'
  == Spawn extension (from-internal, 203, 1) exited non-zero on 'Local/203@from-queue-9f48;2'
    -- Executing [h@from-internal:1] Macro("Local/203@from-queue-9f48;2", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("Local/203@from-queue-9f48;2", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("Local/203@from-queue-9f48;2", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("Local/203@from-queue-9f48;2", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("Local/203@from-queue-9f48;2", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'Local/203@from-queue-9f48;2' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/203@from-queue-9f48;2'
    -- Stopped music on hold on SIP/201-0000000e

<--- SIP read from UDP:192.168.1.213:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport
From: "201" <sip:201@192.168.1.111>;tag=as2330a1d9
To: <sip:203@192.168.1.213:5060;transport=udp>;tag=e516915c3d248474
Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111
CSeq: 102 CANCEL
User-Agent: Grandstream BT200 1.2.2.19
Supported: replaces, timer
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.213:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport
From: "201" <sip:201@192.168.1.111>;tag=as2330a1d9
To: <sip:203@192.168.1.213:5060;transport=udp>;tag=e516915c3d248474
Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.2.2.19
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.213:5060:
ACK sip:203@192.168.1.213:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5e0e1565;rport
Max-Forwards: 70
From: "201" <sip:201@192.168.1.111>;tag=as2330a1d9
To: <sip:203@192.168.1.213:5060;transport=udp>;tag=e516915c3d248474
Contact: <sip:201@192.168.1.111>
Call-ID: 3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0


---
Really destroying SIP dialog '3f9298175c425a3d5b4a517d6360e1c4@192.168.1.111' Method: INVITE
Reliably Transmitting (NAT) to 192.168.1.213:5060:
OPTIONS sip:203@192.168.1.213:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK77822428;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.111>;tag=as54fb82c2
To: <sip:203@192.168.1.213:5060;transport=udp>
Contact: <sip:Unknown@192.168.1.111>
Call-ID: 52f325c63a89535f425a461a45d8908a@192.168.1.111
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.11
Date: Thu, 19 Aug 2010 12:47:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.213:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK77822428;rport
From: "Unknown" <sip:Unknown@192.168.1.111>;tag=as54fb82c2
To: <sip:203@192.168.1.213:5060;transport=udp>;tag=ac178e1d691de328
Call-ID: 52f325c63a89535f425a461a45d8908a@192.168.1.111
CSeq: 102 OPTIONS
User-Agent: Grandstream BT200 1.2.2.19
Contact: <sip:203@192.168.1.213:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '52f325c63a89535f425a461a45d8908a@192.168.1.111' Method: OPTIONS

<--- SIP read from UDP:192.168.1.52:5060 --->
BYE sip:201@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.52:5060;branch=z9hG4bK-jvjz0rd9x021;rport
From: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=nvbtn81cu6
To: "201" <sip:201@192.168.1.111>;tag=as0ec9c5cf
Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111
CSeq: 1 BYE
Max-Forwards: 70
Contact: <sip:202@192.168.1.52:5060;line=budunjf2>;reg-id=1
User-Agent: snom300/7.3.30
RTP-RxStat: Total_Rx_Pkts=68,Rx_Pkts=67,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=67,Tx_Pkts=67,Remote_Tx_Pkts=0
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.1.52 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.1.52:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.52:5060;branch=z9hG4bK-jvjz0rd9x021;received=192.168.1.52;rport=5060
From: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=nvbtn81cu6
To: "201" <sip:201@192.168.1.111>;tag=as0ec9c5cf
Call-ID: 429f11b251cb8f722c27bbfd215259ec@192.168.1.111
CSeq: 1 BYE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Executing [h@macro-dial:1] Macro("Local/202@from-queue-af93;2", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("Local/202@from-queue-af93;2", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("Local/202@from-queue-af93;2", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("Local/202@from-queue-af93;2", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("Local/202@from-queue-af93;2", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'Local/202@from-queue-af93;2' in macro 'hangupcall'
  == Spawn extension (macro-dial, h, 1) exited non-zero on 'Local/202@from-queue-af93;2'
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'Local/202@from-queue-af93;2' in macro 'dial'
  == Spawn extension (macro-exten-vm, s, 11) exited non-zero on 'Local/202@from-queue-af93;2' in macro 'exten-vm'
  == Spawn extension (from-internal, 202, 1) exited non-zero on 'Local/202@from-queue-af93;2'
    -- Executing [h@from-internal:1] Macro("Local/202@from-queue-af93;2", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("Local/202@from-queue-af93;2", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("Local/202@from-queue-af93;2", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("Local/202@from-queue-af93;2", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("Local/202@from-queue-af93;2", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'Local/202@from-queue-af93;2' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/202@from-queue-af93;2'
    -- Executing [h@from-internal:1] Macro("SIP/201-0000000e", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/201-0000000e", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/201-0000000e", "0?skipblkvm") in new stack
    -- Executing [s@macro-hangupcall:5] NoOp("SIP/201-0000000e", "Cleaning Up Block VM Flag: BLKVM/299/SIP/201-0000000e") in new stack
    -- Executing [s@macro-hangupcall:6] NoOp("SIP/201-0000000e", "Deleting: BLKVM/299/SIP/201-0000000e ") in new stack
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/201-0000000e", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/201-0000000e", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/201-0000000e' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-0000000e'
  == Spawn extension (from-internal, 299, 9) exited non-zero on 'SIP/201-0000000e'
    -- Executing [h@from-internal:1] Macro("SIP/201-0000000e", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/201-0000000e", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/201-0000000e", "0?skipblkvm") in new stack
    -- Executing [s@macro-hangupcall:5] NoOp("SIP/201-0000000e", "Cleaning Up Block VM Flag: BLKVM/299/SIP/201-0000000e") in new stack
    -- Executing [s@macro-hangupcall:6] NoOp("SIP/201-0000000e", "Deleting: BLKVM/299/SIP/201-0000000e ") in new stack
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/201-0000000e", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/201-0000000e", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/201-0000000e' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-0000000e'
Scheduling destruction of SIP dialog 'MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.' in 6656 ms (Method: ACK)
set_destination: Parsing <sip:201@192.168.1.5:12592> for address/port to send to
set_destination: set destination to 192.168.1.5, port 12592
Reliably Transmitting (NAT) to 192.168.1.5:12592:
BYE sip:201@192.168.1.5:12592 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5fd6ef35;rport
Max-Forwards: 70
From: "299"<sip:299@192.168.1.111>;tag=as303ba4df
To: <sip:201@192.168.1.111>;tag=11788805
Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.11
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.5:12592 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK5fd6ef35;rport=5060
Contact: <sip:201@192.168.1.5:12592>
To: <sip:201@192.168.1.111>;tag=11788805
From: "299"<sip:299@192.168.1.111>;tag=as303ba4df
Call-ID: MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.
CSeq: 102 BYE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '429f11b251cb8f722c27bbfd215259ec@192.168.1.111' Method: BYE
Really destroying SIP dialog 'MTRiZDVhNjM5Y2I5Y2Q2OWVhNTU2OGJiNTg4YWIyNzM.' Method: ACK