rendered paste bodyvngt*CLI>
<-- SIP read from 202.134.16.202:10578:
--- (0 headers 0 lines) Nat keepalive ---
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
INVITE sip:178@203.162.5.166 SIP/2.0
To: <sip:178@203.162.5.166>
From: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
Via: SIP/2.0/UDP 192.168.3.250:10578;branch=z9hG4bK-d87543-757223881-1--d87543-;rport
Call-ID: 1424354d86767571
CSeq: 1 INVITE
Contact: <sip:1111@192.168.3.250:10578>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 3006o stamp 17551
Content-Length: 238
v=0
o=- 137467970 137468009 IN IP4 192.168.3.250
s=eyeBeam
c=IN IP4 192.168.3.250
t=0 0
m=audio 10574 RTP/AVP 18 101
a=alt:1 1 : E5178AB0 0000009A 192.168.3.250 10574
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (12 headers 10 lines)---
Using INVITE request as basis request - 1424354d86767571
Sending to 192.168.3.250 : 10578 (NAT)
Found peer '1111'
Reliably Transmitting (NAT) to 202.134.16.202:10578:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.3.250:10578;branch=z9hG4bK-d87543-757223881-1--d87543-;received=202.134.16.202;rport=10578
From: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
To: <sip:178@203.162.5.166>;tag=as04103ecc
Call-ID: 1424354d86767571
CSeq: 1 INVITE
User-Agent: VSIP Application Server
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:178@203.162.5.166>
Proxy-Authenticate: Digest realm="", nonce="5c306b30"
Content-Length: 0
---
Scheduling destruction of call '1424354d86767571' in 15000 ms
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
ACK sip:178@203.162.5.166 SIP/2.0
To: <sip:178@203.162.5.166>;tag=as04103ecc
From: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
Via: SIP/2.0/UDP 192.168.3.250:10578;branch=z9hG4bK-d87543-757223881-1--d87543-;rport
Call-ID: 1424354d86767571
CSeq: 1 ACK
Content-Length: 0
--- (7 headers 0 lines)---
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
INVITE sip:178@203.162.5.166 SIP/2.0
To: <sip:178@203.162.5.166>
From: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
Via: SIP/2.0/UDP 192.168.3.250:10578;branch=z9hG4bK-d87543-10906149-1--d87543-;rport
Call-ID: 1424354d86767571
CSeq: 2 INVITE
Contact: <sip:1111@192.168.3.250:10578>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="1111",realm="",nonce="5c306b30",uri="sip:178@203.162.5.166",response="7fa2a25d5307998663c42c990a681e5a",algorithm=MD5
User-Agent: eyeBeam release 3006o stamp 17551
Content-Length: 238
v=0
o=- 137467970 137468009 IN IP4 192.168.3.250
s=eyeBeam
c=IN IP4 192.168.3.250
t=0 0
m=audio 10574 RTP/AVP 18 101
a=alt:1 1 : E5178AB0 0000009A 192.168.3.250 10574
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (13 headers 10 lines)---
Using INVITE request as basis request - 1424354d86767571
Sending to 192.168.3.250 : 10578 (NAT)
Found peer '1111'
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.3.250:10574
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 178 in default (domain 203.162.5.166)
list_route: hop: <sip:1111@192.168.3.250:10578>
Transmitting (NAT) to 202.134.16.202:10578:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.250:10578;branch=z9hG4bK-d87543-10906149-1--d87543-;received=202.134.16.202;rport=10578
From: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
To: <sip:178@203.162.5.166>
Call-ID: 1424354d86767571
CSeq: 2 INVITE
User-Agent: VSIP Application Server
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:178@203.162.5.166>
Content-Length: 0
---
-- Executing Answer("SIP/1111-b4ad", "") in new stack
We're at 203.162.5.166 port 21220
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 202.134.16.202:10578:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.250:10578;branch=z9hG4bK-d87543-10906149-1--d87543-;received=202.134.16.202;rport=10578
From: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
To: <sip:178@203.162.5.166>;tag=as2583541e
Call-ID: 1424354d86767571
CSeq: 2 INVITE
User-Agent: VSIP Application Server
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:178@203.162.5.166>
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 15894 15894 IN IP4 203.162.5.166
s=session
c=IN IP4 203.162.5.166
t=0 0
m=audio 21220 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Executing Dial("SIP/1111-b4ad", "SIP/84903416002@ITSP1") in new stack
We're at 203.162.5.166 port 29936
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (NAT) to 199.37.112.68:5060:
INVITE sip:84903416002@199.37.112.68 SIP/2.0
Via: SIP/2.0/UDP 203.162.5.166:5060;branch=z9hG4bK39fbfc5e;rport
From: "Thai Duy Hoa" <sip:1111@203.162.5.166>;tag=as64d59521
To: <sip:84903416002@199.37.112.68>
Contact: <sip:1111@203.162.5.166>
Call-ID: 2631d51b30915257741a28e51a85c40f@203.162.5.166
CSeq: 102 INVITE
User-Agent: VSIP Application Server
Max-Forwards: 70
Date: Mon, 26 Jun 2006 04:31:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 15894 15894 IN IP4 203.162.5.166
s=session
c=IN IP4 203.162.5.166
t=0 0
m=audio 29936 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 84903416002@ITSP1
Retransmitting #1 (NAT) to 202.134.16.202:10578:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.250:10578;branch=z9hG4bK-d87543-10906149-1--d87543-;received=202.134.16.202;rport=10578
From: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
To: <sip:178@203.162.5.166>;tag=as2583541e
Call-ID: 1424354d86767571
CSeq: 2 INVITE
User-Agent: VSIP Application Server
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:178@203.162.5.166>
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 15894 15894 IN IP4 203.162.5.166
s=session
c=IN IP4 203.162.5.166
t=0 0
m=audio 21220 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
ACK sip:178@203.162.5.166 SIP/2.0
To: <sip:178@203.162.5.166>;tag=as2583541e
From: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
Via: SIP/2.0/UDP 192.168.3.250:10578;branch=z9hG4bK-d87543-47594082-1--d87543-;rport
Call-ID: 1424354d86767571
CSeq: 2 ACK
Contact: <sip:1111@192.168.3.250:10578>
Max-Forwards: 70
Proxy-Authorization: Digest username="1111",realm="",nonce="5c306b30",uri="sip:178@203.162.5.166",response="7fa2a25d5307998663c42c990a681e5a",algorithm=MD5
User-Agent: eyeBeam release 3006o stamp 17551
Content-Length: 0
--- (11 headers 0 lines)---
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
ACK sip:178@203.162.5.166 SIP/2.0
To: <sip:178@203.162.5.166>;tag=as2583541e
From: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
Via: SIP/2.0/UDP 192.168.3.250:10578;branch=z9hG4bK-d87543-47594082-1--d87543-;rport
Call-ID: 1424354d86767571
CSeq: 2 ACK
Contact: <sip:1111@192.168.3.250:10578>
Max-Forwards: 70
Proxy-Authorization: Digest username="1111",realm="",nonce="5c306b30",uri="sip:178@203.162.5.166",response="7fa2a25d5307998663c42c990a681e5a",algorithm=MD5
User-Agent: eyeBeam release 3006o stamp 17551
Content-Length: 0
--- (11 headers 0 lines)---
vngt*CLI>
<-- SIP read from 199.37.112.68:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.162.5.166:5060;branch=z9hG4bK39fbfc5e;rport
From: "Thai Duy Hoa" <sip:1111@203.162.5.166>;tag=as64d59521
To: <sip:84903416002@199.37.112.68>;tag=5da465b352802f109e53000000000000--695
Date: Mon, 26 Jun 2006 04:31:18 GMT
Call-ID: 2631d51b30915257741a28e51a85c40f@203.162.5.166
CSeq: 102 INVITE
User-Agent: VSIP Application Server
Content-Length: 0
--- (9 headers 0 lines)---
vngt*CLI>
<-- SIP read from 199.37.112.68:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 203.162.5.166:5060;branch=z9hG4bK39fbfc5e;rport
From: "Thai Duy Hoa" <sip:1111@203.162.5.166>;tag=as64d59521
To: <sip:84903416002@199.37.112.68>;tag=5da465b352802f109e53000000000000--695
Date: Mon, 26 Jun 2006 04:31:18 GMT
Call-ID: 2631d51b30915257741a28e51a85c40f@203.162.5.166
CSeq: 102 INVITE
Contact: "Proxy" <sip:84903416002@199.37.112.68:5060;transport=UDP>
Content-Length: 243
Content-Type: application/sdp
vngt*CLI>
v=0
o=voizBridge 21839 7169 IN IP4 207.116.52.4
s=SIP Media Capabilities
c=IN IP4 207.116.52.4
t=0 0
m=audio 7842 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (10 headers 11 lines)---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 207.116.52.4:7842
Found description format G729
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
-- SIP/ITSP1-364d is making progress passing it to SIP/1111-b4ad
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
--- (0 headers 0 lines) Nat keepalive ---
vngt*CLI>
<-- SIP read from 199.37.112.68:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.162.5.166:5060;branch=z9hG4bK39fbfc5e;rport
From: "Thai Duy Hoa" <sip:1111@203.162.5.166>;tag=as64d59521
To: <sip:84903416002@199.37.112.68>;tag=5da465b352802f109e53000000000000--695
Date: Mon, 26 Jun 2006 04:31:18 GMT
Call-ID: 2631d51b30915257741a28e51a85c40f@203.162.5.166
CSeq: 102 INVITE
Contact: "Proxy" <sip:84903416002@199.37.112.68:5060;transport=UDP>
Content-Length: 243
Content-Type: application/sdp
v=0
o=voizBridge 21839 7169 IN IP4 207.116.52.4
s=SIP Media Capabilities
c=IN IP4 207.116.52.4
t=0 0
m=audio 7842 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (10 headers 11 lines)---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 207.116.52.4:7842
Found description format G729
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:84903416002@199.37.112.68:5060;transport=UDP>
set_destination: Parsing <sip:84903416002@199.37.112.68:5060;transport=UDP> for address/port to send to
set_destination: set destination to 199.37.112.68, port 5060
Transmitting (NAT) to 199.37.112.68:5060:
ACK sip:84903416002@199.37.112.68:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 203.162.5.166:5060;branch=z9hG4bK44aad2fa;rport
From: "Thai Duy Hoa" <sip:1111@203.162.5.166>;tag=as64d59521
To: <sip:84903416002@199.37.112.68>;tag=5da465b352802f109e53000000000000--695
Contact: <sip:1111@203.162.5.166>
Call-ID: 2631d51b30915257741a28e51a85c40f@203.162.5.166
CSeq: 102 ACK
User-Agent: VSIP Application Server
Max-Forwards: 70
Content-Length: 0
---
-- SIP/ITSP1-364d answered SIP/1111-b4ad
-- Attempting native bridge of SIP/1111-b4ad and SIP/ITSP1-364d
set_destination: Parsing <sip:1111@192.168.3.250:10578> for address/port to send to
set_destination: set destination to 192.168.3.250, port 10578
We're at 203.162.5.166 port 21220
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (NAT) to 202.134.16.202:10578:
INVITE sip:1111@192.168.3.250:10578 SIP/2.0
Via: SIP/2.0/UDP 203.162.5.166:5060;branch=z9hG4bK1d34c7cc;rport
From: <sip:178@203.162.5.166>;tag=as2583541e
To: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
Contact: <sip:178@203.162.5.166>
Call-ID: 1424354d86767571
CSeq: 102 INVITE
User-Agent: VSIP Application Server
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 15894 15895 IN IP4 207.116.52.4
s=session
c=IN IP4 207.116.52.4
t=0 0
m=audio 7842 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
set_destination: Parsing <sip:84903416002@199.37.112.68:5060;transport=UDP> for address/port to send to
set_destination: set destination to 199.37.112.68, port 5060
We're at 203.162.5.166 port 29936
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (NAT) to 199.37.112.68:5060:
INVITE sip:84903416002@199.37.112.68:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 203.162.5.166:5060;branch=z9hG4bK6b7aed1f;rport
From: "Thai Duy Hoa" <sip:1111@203.162.5.166>;tag=as64d59521
To: <sip:84903416002@199.37.112.68>;tag=5da465b352802f109e53000000000000--695
Contact: <sip:1111@203.162.5.166>
Call-ID: 2631d51b30915257741a28e51a85c40f@203.162.5.166
CSeq: 103 INVITE
User-Agent: VSIP Application Server
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 243
v=0
o=root 15894 15895 IN IP4 202.134.16.202
s=session
c=IN IP4 202.134.16.202
t=0 0
m=audio 10574 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
SIP/2.0 200 OK
To: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
From: <sip:178@203.162.5.166>;tag=as2583541e
Via: SIP/2.0/UDP 203.162.5.166:5060;branch=z9hG4bK1d34c7cc;rport=5060;received=203.162.5.166
Call-ID: 1424354d86767571
CSeq: 102 INVITE
Contact: <sip:1111@192.168.3.250:10578>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 238
v=0
o=- 137467970 137468009 IN IP4 192.168.3.250
s=eyeBeam
c=IN IP4 192.168.3.250
t=0 0
m=audio 10574 RTP/AVP 18 101
a=alt:1 1 : E5178AB0 0000009A 192.168.3.250 10574
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (10 headers 10 lines)---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.3.250:10574
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:1111@192.168.3.250:10578>
set_destination: Parsing <sip:1111@192.168.3.250:10578> for address/port to send to
set_destination: set destination to 192.168.3.250, port 10578
Transmitting (NAT) to 202.134.16.202:10578:
ACK sip:1111@192.168.3.250:10578 SIP/2.0
Via: SIP/2.0/UDP 203.162.5.166:5060;branch=z9hG4bK4d4f1faf;rport
From: <sip:178@203.162.5.166>;tag=as2583541e
To: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
Contact: <sip:178@203.162.5.166>
Call-ID: 1424354d86767571
CSeq: 102 ACK
User-Agent: VSIP Application Server
Max-Forwards: 70
Content-Length: 0
---
vngt*CLI>
<-- SIP read from 199.37.112.68:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.162.5.166:5060;branch=z9hG4bK6b7aed1f;rport
From: "Thai Duy Hoa" <sip:1111@203.162.5.166>;tag=as64d59521
To: <sip:84903416002@199.37.112.68>;tag=5da465b352802f109e53000000000000--695
Call-ID: 2631d51b30915257741a28e51a85c40f@203.162.5.166
CSeq: 103 INVITE
User-Agent: VSIP Application Server
Content-Length: 0
X-asterisk-info: SIP re-invite (RTP bridge)
--- (9 headers 0 lines)---
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
--- (0 headers 0 lines) Nat keepalive ---
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
--- (0 headers 0 lines) Nat keepalive ---
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
BYE sip:178@203.162.5.166 SIP/2.0
To: <sip:178@203.162.5.166>;tag=as2583541e
From: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
Via: SIP/2.0/UDP 192.168.3.250:10578;branch=z9hG4bK-d87543-98783836-1--d87543-;rport
Call-ID: 1424354d86767571
CSeq: 3 BYE
Contact: <sip:1111@192.168.3.250:10578>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Proxy-Authorization: Digest username="1111",realm="",nonce="5c306b30",uri="sip:178@203.162.5.166",response="da5d2c096f6f9d39cbaed484e4b7a7fb",algorithm=MD5
User-Agent: eyeBeam release 3006o stamp 17551
Content-Length: 0
--- (12 headers 0 lines)---
Sending to 192.168.3.250 : 10578 (NAT)
Transmitting (NAT) to 202.134.16.202:10578:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.250:10578;branch=z9hG4bK-d87543-98783836-1--d87543-;received=202.134.16.202;rport=10578
From: Hoa Thai<sip:1111@203.162.5.166>;tag=2b45fe3f
To: <sip:178@203.162.5.166>;tag=as2583541e
Call-ID: 1424354d86767571
CSeq: 3 BYE
User-Agent: VSIP Application Server
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:178@203.162.5.166>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
Destroying call '1424354d86767571'
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
--- (0 headers 0 lines) Nat keepalive ---
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
vngt*CLI>
<-- SIP read from 202.134.16.202:10578:
--- (0 headers 0 lines) Nat keepalive ---
vngt*CLI>
<-- SIP read from 199.37.112.68:5060:
BYE sip:1111@203.162.5.166;transport=UDP SIP/2.0
To: "Thai Duy Hoa" <sip:1111@203.162.5.166>;tag=as64d59521
From: <sip:84903416002@199.37.112.68>;tag=5da465b352802f109e53000000000000--695
Via: SIP/2.0/UDP 199.37.112.68:5060
Call-ID: 2631d51b30915257741a28e51a85c40f@203.162.5.166
CSeq: 104 BYE
Content-Length: 0
Max-Forwards: 70
--- (8 headers 0 lines)---
Sending to 199.37.112.68 : 5060 (NAT)
Transmitting (NAT) to 199.37.112.68:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 199.37.112.68:5060;received=199.37.112.68
From: <sip:84903416002@199.37.112.68>;tag=5da465b352802f109e53000000000000--695
To: "Thai Duy Hoa" <sip:1111@203.162.5.166>;tag=as64d59521
Call-ID: 2631d51b30915257741a28e51a85c40f@203.162.5.166
CSeq: 104 BYE
User-Agent: VSIP Application Server
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1111@203.162.5.166>
Content-Length: 0
---