<--- SIP read from UDP://82.37.110.107:58566 --->
INVITE sip:30691111@IP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:58566;branch=z9hG4bK-d8754z-7536775afc7b2860-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1111@82.37.110.107:58566>
To: "30691111"<sip:30691111@domain;tag=as62b4c520
From: "1111"<sip:1111@domain>;tag=8d55ce56
Call-ID: ZTJhNWNjOGY0ODYwOTA1NjcwNDE5ZGVjYjhmNTZiYzg.
CSeq: 3 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="1111",realm="asterisk",nonce="3a1dab2d",uri="sip:30691111@ip",response="4be09150e30822b4dbb394b2bc7c0977",algorithm=MD5
Content-Length: 581

v=0
o=- 0 3 IN IP4 192.168.0.101
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.101
t=0 0
m=audio 49718 RTP/AVP 107 0 8 101
a=alt:1 2 : Dy1lGy3b bzU5feP0 192.168.213.174 49718
a=alt:2 1 : AYgByBza uEffXEot 192.168.0.101 49718
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
m=video 48868 RTP/AVP 115 34
a=alt:1 2 : iL65WCSO zass7MmM 192.168.213.174 48868
a=alt:2 1 : t9whmZEt D77EXiRR 192.168.0.101 48868
a=fmtp:115 QCIF=1;CIF=1;I=1;J=1;T=1
a=fmtp:34 QCIF=1;CIF=1
a=rtpmap:115 H263-1998/90000
a=rtpmap:34 H263/90000
a=sendrecv

<------------->
--- (13 headers 20 lines) ---
Sending to 82.37.110.107 : 58566 (NAT)
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found RTP video format 115
Found RTP video format 34
Peer audio RTP is at port 192.168.0.101:49718
Found unknown media description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Found video description format H263-1998 for ID 115
Found video description format H263 for ID 34
Capabilities: us - 0x38000a (gsm|alaw|h263|h263p|h264), peer - audio=0xc (ulaw|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x180008 (alaw|h263|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.101:49718
Peer video RTP is at port 192.168.0.101:48868

<--- Transmitting (NAT) to 82.37.110.107:58566 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.101:58566;branch=z9hG4bK-d8754z-7536775afc7b2860-1---d8754z-;received=82.37.110.107;rport=58566
From: "1111"<sip:1111@domain;tag=8d55ce56
To: "30691111"<sip:30691111@domain;tag=as62b4c520
Call-ID: ZTJhNWNjOGY0ODYwOTA1NjcwNDE5ZGVjYjhmNTZiYzg.
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:30691111@IP>
Content-Length: 0

<------------>
Audio is at IP port 17726
Video is at IP port 12172
Adding codec 0x8 (alaw) to SDP
Adding video codec 0x80000 (h263) to SDP
Adding video codec 0x100000 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 82.37.110.107:58566 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.101:58566;branch=z9hG4bK-d8754z-7536775afc7b2860-1---d8754z-;received=82.37.110.107;rport=58566
From: "1111"<sip:1111@domain;tag=8d55ce56
To: "30691111"<sip:30691111@domain;tag=as62b4c520
Call-ID: ZTJhNWNjOGY0ODYwOTA1NjcwNDE5ZGVjYjhmNTZiYzg.
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:30691111@IP>
Content-Type: application/sdp
Content-Length: 369

v=0
o=root 1788008387 1788008388 IN IP4 IP
s=Asterisk PBX 1.6.1.1
c=IN IP4 IP
b=CT:384
t=0 0
m=audio 17726 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34 115
a=rtpmap:34 H263/90000
a=rtpmap:115 h263-1998/90000
a=sendrecv

<------------>
asterisk1-uk*CLI>
<--- SIP read from UDP://82.37.110.107:58566 --->
ACK sip:30691111@IP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:58566;branch=z9hG4bK-d8754z-127ac2131a251873-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1111@82.37.110.107:58566>
To: "30691111"<sip:30691111@domain;tag=as62b4c520
From: "1111"<sip:1111@domain;tag=8d55ce56
Call-ID: ZTJhNWNjOGY0ODYwOTA1NjcwNDE5ZGVjYjhmNTZiYzg.
CSeq: 3 ACK
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username="1111",realm="asterisk",nonce="3a1dab2d",uri="sip:30691111@IP",response="4be09150e30822b4dbb394b2bc7c0977",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
asterisk1-uk*CLI>
<--- SIP read from UDP://82.37.110.107:58566 --->
BYE sip:30691111@IP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:58566;branch=z9hG4bK-d8754z-79774b531325106e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1111@82.37.110.107:58566>
To: "30691111"<sip:30691111@domain;tag=as62b4c520
From: "1111"<sip:1111@domain;tag=8d55ce56
Call-ID: ZTJhNWNjOGY0ODYwOTA1NjcwNDE5ZGVjYjhmNTZiYzg.
CSeq: 4 BYE
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username="1111",realm="asterisk",nonce="3a1dab2d",uri="sip:30691111@IP",response="093a78e21659d88ede6faf854050f2df",algorithm=MD5
Reason: SIP;description="User Hung Up"
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 82.37.110.107 : 58566 (NAT)
asterisk1-uk*CLI>
<--- Transmitting (NAT) to 82.37.110.107:58566 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.101:58566;branch=z9hG4bK-d8754z-79774b531325106e-1---d8754z-;received=82.37.110.107;rport=58566
From: "1111"<sip:1111@domain;tag=8d55ce56
To: "30691111"<sip:30691111@domain;tag=as62b4c520
Call-ID: ZTJhNWNjOGY0ODYwOTA1NjcwNDE5ZGVjYjhmNTZiYzg.
CSeq: 4 BYE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '0382376c7f31d7721e3b4b2015fbcddd@IP' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:30691111@82.37.110.107:38604> for address/port to send to
set_destination: set destination to 82.37.110.107, port 38604
Reliably Transmitting (NAT) to 82.37.110.107:38604:
BYE sip:30691111@82.37.110.107:38604 SIP/2.0
Via: SIP/2.0/UDP IP:5060;branch=z9hG4bK179dc884;rport
Max-Forwards: 70
From: "1111" <sip:1111@IP>;tag=as5672544c
To: <sip:30691111@82.37.110.107:38604>;tag=d1853d9a
Call-ID: 0382376c7f31d7721e3b4b2015fbcddd@IP
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.1.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
asterisk1-uk*CLI>
<--- SIP read from UDP://82.37.110.107:38604 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP:5060;branch=z9hG4bK179dc884;rport=5060
Contact: <sip:30691111@82.37.110.107:38604>
To: <sip:30691111@82.37.110.107:38604>;tag=d1853d9a
From: "1111"<sip:1111@IP>;tag=as5672544c
Call-ID: 0382376c7f31d7721e3b4b2015fbcddd@IP
CSeq: 103 BYE
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0

<------------->