<--- SIP read from UDP:192.168.1.52:5060 --->
REGISTER sip:192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.52:5060;branch=z9hG4bK-k9r8m2zm6ujk;rport
From: "202" <sip:202@192.168.1.111>;tag=txfeaogcl3
To: "202" <sip:202@192.168.1.111>
Call-ID: 3c37cf78b16b-qe7ayp4mkb4s
CSeq: 7602 REGISTER
Max-Forwards: 70
Contact: <sip:202@192.168.1.52:5060;line=budunjf2>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:81bc9a01-eb78-4530-8730-8dbe7b9d0e80>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom300/7.3.30
Supported: gruu
Allow-Events: dialog
X-Real-IP: 192.168.1.52
Authorization: Digest username="202",realm="asterisk",nonce="63ace492",uri="sip:192.168.1.111",response="10c755040a710f9c6773b422a2f4dd32",algorithm=MD5
Expires: 600
Content-Length: 0


<------------->
--- (15 headers 0 lines) ---
Sending to 192.168.1.52 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.1.52:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.52:5060;branch=z9hG4bK-k9r8m2zm6ujk;received=192.168.1.52;rport=5060
From: "202" <sip:202@192.168.1.111>;tag=txfeaogcl3
To: "202" <sip:202@192.168.1.111>
Call-ID: 3c37cf78b16b-qe7ayp4mkb4s
CSeq: 7602 REGISTER
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Reliably Transmitting (NAT) to 192.168.1.52:5060:
OPTIONS sip:202@192.168.1.52:5060;line=budunjf2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK74f4cd22;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.111>;tag=as1dc277e6
To: <sip:202@192.168.1.52:5060;line=budunjf2>
Contact: <sip:Unknown@192.168.1.111>
Call-ID: 0e1da75436908eb865b5aad63b0e1fa0@192.168.1.111
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.11
Date: Thu, 19 Aug 2010 12:51:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- 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-k9r8m2zm6ujk;received=192.168.1.52;rport=5060
From: "202" <sip:202@192.168.1.111>;tag=txfeaogcl3
To: "202" <sip:202@192.168.1.111>;tag=as518e7fed
Call-ID: 3c37cf78b16b-qe7ayp4mkb4s
CSeq: 7602 REGISTER
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 600
Contact: <sip:202@192.168.1.52:5060;line=budunjf2>;expires=600
Date: Thu, 19 Aug 2010 12:51:10 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3c37cf78b16b-qe7ayp4mkb4s' in 32000 ms (Method: REGISTER)

<--- 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=z9hG4bK74f4cd22;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.111>;tag=as1dc277e6
To: <sip:202@192.168.1.52:5060;line=budunjf2>
Call-ID: 0e1da75436908eb865b5aad63b0e1fa0@192.168.1.111
CSeq: 102 OPTIONS
Contact: <sip:202@192.168.1.52:5060;line=budunjf2>;reg-id=1
User-Agent: snom300/7.3.30
Accept-Language: en
Accept: application/sdp
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-Length: 0


<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '0e1da75436908eb865b5aad63b0e1fa0@192.168.1.111' Method: OPTIONS

<--- 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-077f9b5b8e05675e-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=6e69bc6d
Call-ID: ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.
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=- 8 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 11496 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 - ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.
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-077f9b5b8e05675e-1---d8754z-;received=192.168.1.5;rport=12592
From: <sip:201@192.168.1.111>;tag=6e69bc6d
To: "299"<sip:299@192.168.1.111>;tag=as4ed500c0
Call-ID: ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.
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="16186e99"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.' in 6400 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-077f9b5b8e05675e-1---d8754z-;rport
To: "299"<sip:299@192.168.1.111>;tag=as4ed500c0
From: <sip:201@192.168.1.111>;tag=6e69bc6d
Call-ID: ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.
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-c4508f0a8e38c61d-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=6e69bc6d
Call-ID: ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.
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="16186e99",uri="sip:299@192.168.1.111",response="256a20feb08f11d90e98782449f9a273",algorithm=MD5
Content-Length: 214

v=0
o=- 8 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 11496 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 - ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.
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 192.168.1.5:11496
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-c4508f0a8e38c61d-1---d8754z-;received=192.168.1.5;rport=12592
From: <sip:201@192.168.1.111>;tag=6e69bc6d
To: "299"<sip:299@192.168.1.111>
Call-ID: ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.
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/201-00000011", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/201-00000011", "AMPUSER=201") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/201-00000011", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/201-00000011", "1?Set(REALCALLERIDNUM=201)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/201-00000011", "AMPUSER=201") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/201-00000011", "AMPUSERCIDNAME=201") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/201-00000011", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/201-00000011", "AMPUSERCID=201") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/201-00000011", "CALLERID(all)="201" <201>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/201-00000011", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/201-00000011", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/201-00000011", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/201-00000011", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/201-00000011", "Using CallerID "201" <201>") in new stack
    -- Executing [299@from-internal:2] Answer("SIP/201-00000011", "") in new stack
Audio is at 192.168.1.111 port 10272
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-c4508f0a8e38c61d-1---d8754z-;received=192.168.1.5;rport=12592
From: <sip:201@192.168.1.111>;tag=6e69bc6d
To: "299"<sip:299@192.168.1.111>;tag=as198d95c0
Call-ID: ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.
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: 262

v=0
o=root 1493762953 1493762953 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 10272 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/201-00000011", "__BLKVM_OVERRIDE=BLKVM/299/SIP/201-00000011") in new stack
    -- Executing [299@from-internal:4] Set("SIP/201-00000011", "__BLKVM_BASE=299") in new stack
    -- Executing [299@from-internal:5] Set("SIP/201-00000011", "DB(BLKVM/299/SIP/201-00000011)=TRUE") in new stack
    -- Executing [299@from-internal:6] ExecIf("SIP/201-00000011", "1?Set(_DIAL_OPTIONS=tkM(auto-blkvm))") in new stack
    -- Executing [299@from-internal:7] Set("SIP/201-00000011", "__NODEST=299") in new stack
    -- Executing [299@from-internal:8] Set("SIP/201-00000011", "MONITOR_FILENAME=/var/spool/asterisk/monitor/q299-20100819-145113-1282222273.26") in new stack
    -- Executing [299@from-internal:9] Queue("SIP/201-00000011", "299,t,,") in new stack
    -- Started music on hold, class 'default', on SIP/201-00000011
  == 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 10196
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=z9hG4bK789fa776;rport
Max-Forwards: 70
From: "201" <sip:201@192.168.1.111>;tag=as1371d7f3
To: <sip:202@192.168.1.52:5060;line=budunjf2>
Contact: <sip:201@192.168.1.111>
Call-ID: 2a97df6e7d59ce356a28f019123f1214@192.168.1.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Thu, 19 Aug 2010 12:51:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 715063718 715063718 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 10196 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

---
  == 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 10044
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:
INVITE sip:203@192.168.1.213:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK70d70222;rport
Max-Forwards: 70
From: "201" <sip:201@192.168.1.111>;tag=as405f3769
To: <sip:203@192.168.1.213:5060;transport=udp>
Contact: <sip:201@192.168.1.111>
Call-ID: 53923e6f0bb1583f320936cd7bc17346@192.168.1.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Thu, 19 Aug 2010 12:51:13 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 1769870696 1769870696 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 10044 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

---

<--- 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-b01ce51dd405f85a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:201@192.168.1.5:12592>
To: "299"<sip:299@192.168.1.111>;tag=as198d95c0
From: <sip:201@192.168.1.111>;tag=6e69bc6d
Call-ID: ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.
CSeq: 2 ACK
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username="201",realm="asterisk",nonce="16186e99",uri="sip:299@192.168.1.111",response="256a20feb08f11d90e98782449f9a273",algorithm=MD5
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---

<--- 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=z9hG4bK70d70222;rport
From: "201" <sip:201@192.168.1.111>;tag=as405f3769
To: <sip:203@192.168.1.213:5060;transport=udp>
Call-ID: 53923e6f0bb1583f320936cd7bc17346@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=z9hG4bK70d70222;rport
From: "201" <sip:201@192.168.1.111>;tag=as405f3769
To: <sip:203@192.168.1.213:5060;transport=udp>;tag=a655a055aa502950
Call-ID: 53923e6f0bb1583f320936cd7bc17346@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/203-00000013 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=z9hG4bK789fa776;rport=5060
From: "201" <sip:201@192.168.1.111>;tag=as1371d7f3
To: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=f56fcamen3
Call-ID: 2a97df6e7d59ce356a28f019123f1214@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/202-00000012 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=z9hG4bK789fa776;rport=5060
From: "201" <sip:201@192.168.1.111>;tag=as1371d7f3
To: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=f56fcamen3
Call-ID: 2a97df6e7d59ce356a28f019123f1214@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/202-00000012 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=z9hG4bK789fa776;rport=5060
From: "201" <sip:201@192.168.1.111>;tag=as1371d7f3
To: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=f56fcamen3
Call-ID: 2a97df6e7d59ce356a28f019123f1214@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: 265

v=0
o=root 520069819 520069820 IN IP4 192.168.1.52
s=call
c=IN IP4 192.168.1.52
t=0 0
m=audio 59168 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 192.168.1.52:59168
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=z9hG4bK2d1ebfc1;rport
Max-Forwards: 70
From: "201" <sip:201@192.168.1.111>;tag=as1371d7f3
To: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=f56fcamen3
ontact: <sip:201@192.168.1.111>
Call-ID: 2a97df6e7d59ce356a28f019123f1214@192.168.1.111
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0


---
    -- SIP/202-00000012 answered SIP/201-00000011
Scheduling destruction of SIP dialog '53923e6f0bb1583f320936cd7bc17346@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=z9hG4bK70d70222;rport
Max-Forwards: 70
From: "201" <sip:201@192.168.1.111>;tag=as405f3769
To: <sip:203@192.168.1.213:5060;transport=udp>
Call-ID: 53923e6f0bb1583f320936cd7bc17346@192.168.1.111
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.11
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '53923e6f0bb1583f320936cd7bc17346@192.168.1.111' in 6400 ms (Method: INVITE)
    -- Stopped music on hold on SIP/201-00000011

<--- 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=z9hG4bK70d70222;rport
From: "201" <sip:201@192.168.1.111>;tag=as405f3769
To: <sip:203@192.168.1.213:5060;transport=udp>;tag=a655a055aa502950
Call-ID: 53923e6f0bb1583f320936cd7bc17346@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=z9hG4bK70d70222;rport
From: "201" <sip:201@192.168.1.111>;tag=as405f3769
To: <sip:203@192.168.1.213:5060;transport=udp>;tag=a655a055aa502950
Call-ID: 53923e6f0bb1583f320936cd7bc17346@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=z9hG4bK70d70222;rport
Max-Forwards: 70
From: "201" <sip:201@192.168.1.111>;tag=as405f3769
To: <sip:203@192.168.1.213:5060;transport=udp>;tag=a655a055aa502950
Contact: <sip:201@192.168.1.111>
Call-ID: 53923e6f0bb1583f320936cd7bc17346@192.168.1.111
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0


---
Really destroying SIP dialog '53923e6f0bb1583f320936cd7bc17346@192.168.1.111' Method: INVITE

<--- 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-kofoxqwjtfqz;rport
From: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=f56fcamen3
To: "201" <sip:201@192.168.1.111>;tag=as1371d7f3
Call-ID: 2a97df6e7d59ce356a28f019123f1214@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=108,Rx_Pkts=107,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=107,Tx_Pkts=107,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-kofoxqwjtfqz;received=192.168.1.52;rport=5060
From: <sip:202@192.168.1.52:5060;line=budunjf2>;tag=f56fcamen3
To: "201" <sip:201@192.168.1.111>;tag=as1371d7f3
Call-ID: 2a97df6e7d59ce356a28f019123f1214@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@from-internal:1] Macro("SIP/201-00000011", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/201-00000011", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/201-00000011", "0?skipblkvm") in new stack
    -- Executing [s@macro-hangupcall:5] NoOp("SIP/201-00000011", "Cleaning Up Block VM Flag: BLKVM/299/SIP/201-00000011") in new stack
    -- Executing [s@macro-hangupcall:6] NoOp("SIP/201-00000011", "Deleting: BLKVM/299/SIP/201-00000011 TRUE") in new stack
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/201-00000011", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/201-00000011", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/201-00000011' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-00000011'
  == Spawn extension (from-internal, 299, 9) exited non-zero on 'SIP/201-00000011'
    -- Executing [h@from-internal:1] Macro("SIP/201-00000011", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/201-00000011", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/201-00000011", "0?skipblkvm") in new stack
    -- Executing [s@macro-hangupcall:5] NoOp("SIP/201-00000011", "Cleaning Up Block VM Flag: BLKVM/299/SIP/201-00000011") in new stack
    -- Executing [s@macro-hangupcall:6] NoOp("SIP/201-00000011", "Deleting: BLKVM/299/SIP/201-00000011 ") in new stack
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/201-00000011", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/201-00000011", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/201-00000011' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-00000011'
Scheduling destruction of SIP dialog 'ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.' in 6400 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=z9hG4bK71bc43a6;rport
Max-Forwards: 70
From: "299"<sip:299@192.168.1.111>;tag=as198d95c0
To: <sip:201@192.168.1.111>;tag=6e69bc6d
Call-ID: ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.
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=z9hG4bK71bc43a6;rport=5060
Contact: <sip:201@192.168.1.5:12592>
To: <sip:201@192.168.1.111>;tag=6e69bc6d
From: "299"<sip:299@192.168.1.111>;tag=as198d95c0
Call-ID: ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.
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 'ODA5ZjEwYWY2MjM3MjAzMThjMzQxNzAxZTUzZmYzYWU.' Method: ACK
Really destroying SIP dialog '2a97df6e7d59ce356a28f019123f1214@192.168.1.111' Method: BYE