
<--- SIP read from UDP:10.0.3.1:46041 --->
INVITE sip:500990990@10.0.3.183;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.3.1:46041;branch=z9hG4bK-d8754z-0f2ab55600832a10-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:tester1@10.0.3.1:46041;transport=UDP>
To: <sip:500990990@10.0.3.183;transport=UDP>
From: <sip:tester1@10.0.3.183;transport=UDP>;tag=564b4966
Call-ID: YjY2MjBkMDUwMzM3MWVjZmQ3NmRhNWMwOTNkYWFhM2I.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Allow-Events: presence, kpml
Content-Length: 319

v=0
o=Zoiper_user 0 0 IN IP4 10.0.3.1
s=Zoiper_session
c=IN IP4 10.0.3.1
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 15 lines) ---
Sending to 10.0.3.1:46041 (NAT)
Using INVITE request as basis request - YjY2MjBkMDUwMzM3MWVjZmQ3NmRhNWMwOTNkYWFhM2I.
Found peer 'tester1' for 'tester1' from 10.0.3.1:46041

<--- Reliably Transmitting (NAT) to 10.0.3.1:46041 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.3.1:46041;branch=z9hG4bK-d8754z-0f2ab55600832a10-1---d8754z-;received=10.0.3.1;rport=46041
From: <sip:tester1@10.0.3.183;transport=UDP>;tag=564b4966
To: <sip:500990990@10.0.3.183;transport=UDP>;tag=as69ae263d
Call-ID: YjY2MjBkMDUwMzM3MWVjZmQ3NmRhNWMwOTNkYWFhM2I.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="042ed8ad"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'YjY2MjBkMDUwMzM3MWVjZmQ3NmRhNWMwOTNkYWFhM2I.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:10.0.3.1:46041 --->
ACK sip:500990990@10.0.3.183;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.3.1:46041;branch=z9hG4bK-d8754z-0f2ab55600832a10-1---d8754z-;rport
Max-Forwards: 70
To: <sip:500990990@10.0.3.183;transport=UDP>;tag=as69ae263d
From: <sip:tester1@10.0.3.183;transport=UDP>;tag=564b4966
Call-ID: YjY2MjBkMDUwMzM3MWVjZmQ3NmRhNWMwOTNkYWFhM2I.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:10.0.3.1:46041 --->
INVITE sip:500990990@10.0.3.183;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.3.1:46041;branch=z9hG4bK-d8754z-792cb5fa4b1ae9b4-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:tester1@10.0.3.1:46041;transport=UDP>
To: <sip:500990990@10.0.3.183;transport=UDP>
From: <sip:tester1@10.0.3.183;transport=UDP>;tag=564b4966
Call-ID: YjY2MjBkMDUwMzM3MWVjZmQ3NmRhNWMwOTNkYWFhM2I.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Authorization: Digest username="tester1",realm="asterisk",nonce="042ed8ad",uri="sip:500990990@10.0.3.183;transport=UDP",response="af98ce8f8a39917e2a9ca3912f5f7f5d",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 319

v=0
o=Zoiper_user 0 0 IN IP4 10.0.3.1
s=Zoiper_session
c=IN IP4 10.0.3.1
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 15 lines) ---
Sending to 10.0.3.1:46041 (NAT)
Using INVITE request as basis request - YjY2MjBkMDUwMzM3MWVjZmQ3NmRhNWMwOTNkYWFhM2I.
Found peer 'tester1' for 'tester1' from 10.0.3.1:46041
  == Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.3.1:8000
Looking for 500990990 in inc (domain 10.0.3.183)
list_route: hop: <sip:tester1@10.0.3.1:46041;transport=UDP>

<--- Transmitting (NAT) to 10.0.3.1:46041 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.3.1:46041;branch=z9hG4bK-d8754z-792cb5fa4b1ae9b4-1---d8754z-;received=10.0.3.1;rport=46041
From: <sip:tester1@10.0.3.183;transport=UDP>;tag=564b4966
To: <sip:500990990@10.0.3.183;transport=UDP>
Call-ID: YjY2MjBkMDUwMzM3MWVjZmQ3NmRhNWMwOTNkYWFhM2I.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:500990990@10.0.3.183:5060>
Content-Length: 0


<------------>
    -- Executing [500990990@inc:1] NoOp("SIP/tester1-00000002", "Polaczenie przychodzace 500990990,0508381197") in new stack
    -- Executing [500990990@inc:2] Set("SIP/tester1-00000002", "GROUP(pstn)=inc") in new stack
    -- Executing [500990990@inc:3] GotoIf("SIP/tester1-00000002", "0?40:4") in new stack
    -- Goto (inc,500990990,4)
    -- Executing [500990990@inc:4] Set("SIP/tester1-00000002", "RANDX=30") in new stack
    -- Executing [500990990@inc:5] Set("SIP/tester1-00000002", "LOOPX=0") in new stack
    -- Executing [500990990@inc:6] Set("SIP/tester1-00000002", "i=0") in new stack
    -- Executing [500990990@inc:7] Set("SIP/tester1-00000002", "i=1") in new stack
    -- Executing [500990990@inc:8] Set("SIP/tester1-00000002", "j=0") in new stack
    -- Executing [500990990@inc:9] While("SIP/tester1-00000002", "1") in new stack
    -- Executing [500990990@inc:10] Dial("SIP/tester1-00000002", "SIP/500990990@sip1,,,") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.10.10.180:5060:
INVITE sip:500990990@10.10.10.180 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.183:5060;branch=z9hG4bK6eab0d48;rport
Max-Forwards: 70
From: "tester1" <sip:0508381197@10.10.10.183>;tag=as3e54850f
To: <sip:500990990@10.10.10.180>
Contact: <sip:0508381197@10.10.10.183:5060>
Call-ID: 544307980ae1dcd5090b3ab80b8a1da1@10.10.10.183:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.4.1
Date: Sun, 29 May 2011 21:30:01 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 226301983 226301983 IN IP4 10.10.10.183
s=Asterisk PBX 1.8.4.1
c=IN IP4 10.10.10.183
t=0 0
m=audio 12528 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 500990990@sip1

<--- SIP read from UDP:10.10.10.180:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.10.10.183:5060;branch=z9hG4bK6eab0d48;rport=5060
From: "tester1" <sip:0508381197@10.10.10.183>;tag=as3e54850f
To: <sip:500990990@10.10.10.180>
Call-ID: 544307980ae1dcd5090b3ab80b8a1da1@10.10.10.183:5060
CSeq: 102 INVITE
Server: Conference-eX v.0.9
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:10.10.10.180:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.183:5060;branch=z9hG4bK6eab0d48;rport=5060
Record-Route: <sip:10.10.10.180;lr;ftag=as3e54850f;did=b1b.12884885>
From: "tester1" <sip:0508381197@10.10.10.183>;tag=as3e54850f
To: <sip:500990990@10.10.10.180>;tag=116609171
Call-ID: 544307980ae1dcd5090b3ab80b8a1da1@10.10.10.183:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:500990990@10.10.10.182>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 22706 22706 IN IP4 10.10.10.182
s=session
c=IN IP4 10.10.10.182
t=0 0
m=audio 11824 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
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 telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.10.182:11824
list_route: hop: <sip:10.10.10.180;lr;ftag=as3e54850f;did=b1b.12884885>
set_destination: Parsing <sip:10.10.10.180;lr;ftag=as3e54850f;did=b1b.12884885> for address/port to send to
set_destination: set destination to 10.10.10.180:5060
Transmitting (NAT) to 10.10.10.180:5060:
ACK sip:500990990@10.10.10.182 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.183:5060;branch=z9hG4bK09f0c31f;rport
Route: <sip:10.10.10.180;lr;ftag=as3e54850f;did=b1b.12884885>
Max-Forwards: 70
From: "tester1" <sip:0508381197@10.10.10.183>;tag=as3e54850f
To: <sip:500990990@10.10.10.180>;tag=116609171
Contact: <sip:0508381197@10.10.10.183:5060>
Call-ID: 544307980ae1dcd5090b3ab80b8a1da1@10.10.10.183:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.4.1
Content-Length: 0


---
    -- SIP/sip1-00000003 answered SIP/tester1-00000002
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 10.0.3.1:46041 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.3.1:46041;branch=z9hG4bK-d8754z-792cb5fa4b1ae9b4-1---d8754z-;received=10.0.3.1;rport=46041
From: <sip:tester1@10.0.3.183;transport=UDP>;tag=564b4966
To: <sip:500990990@10.0.3.183;transport=UDP>;tag=as31924b27
Call-ID: YjY2MjBkMDUwMzM3MWVjZmQ3NmRhNWMwOTNkYWFhM2I.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:500990990@10.0.3.183:5060>
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 995164251 995164251 IN IP4 10.0.3.183
s=Asterisk PBX 1.8.4.1
c=IN IP4 10.0.3.183
t=0 0
m=audio 10546 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Locally bridging SIP/tester1-00000002 and SIP/sip1-00000003

<--- SIP read from UDP:10.0.3.1:46041 --->
ACK sip:500990990@10.0.3.183:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.1:46041;branch=z9hG4bK-d8754z-6b283e84bd0d3822-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:tester1@10.0.3.1:46041;transport=UDP>
To: <sip:500990990@10.0.3.183;transport=UDP>;tag=as31924b27
From: <sip:tester1@10.0.3.183;transport=UDP>;tag=564b4966
Call-ID: YjY2MjBkMDUwMzM3MWVjZmQ3NmRhNWMwOTNkYWFhM2I.
CSeq: 2 ACK
User-Agent: Zoiper rev.11137
Authorization: Digest username="tester1",realm="asterisk",nonce="042ed8ad",uri="sip:500990990@10.0.3.183;transport=UDP",response="af98ce8f8a39917e2a9ca3912f5f7f5d",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:10.0.0.239:62435 --->
NOTIFY sip:10.0.3.183 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.141:5060;branch=z9hG4bK-a0ea78d7
From: "tester1" <sip:tester1@10.0.3.183>;tag=cc0ea42734b4646o2
To: <sip:10.0.3.183>
Call-ID: e13969a3-b334defa@192.168.10.141
CSeq: 9741 NOTIFY
Max-Forwards: 70
Contact: "tester1" <sip:tester1@192.168.10.141:5060>
Event: keep-alive
User-Agent: Linksys/SPA942-6.1.3(a)_FCN
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- Transmitting (NAT) to 10.0.0.239:62435 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.141:5060;branch=z9hG4bK-a0ea78d7;received=10.0.0.239;rport=62435
From: "tester1" <sip:tester1@10.0.3.183>;tag=cc0ea42734b4646o2
To: <sip:10.0.3.183>;tag=as2faaf00d
Call-ID: e13969a3-b334defa@192.168.10.141
CSeq: 9741 NOTIFY
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'e13969a3-b334defa@192.168.10.141' in 32000 ms (Method: NOTIFY)

<--- SIP read from UDP:10.10.10.180:5060 --->
INVITE sip:0508381197@10.10.10.183:5060 SIP/2.0
Record-Route: <sip:10.10.10.180;lr;ftag=as1de6e3fc;did=83e.3d1b6133>
Via: SIP/2.0/UDP 10.10.10.180;branch=z9hG4bK80d9.dacfd6c.0
Via: SIP/2.0/UDP 10.10.10.181:5060;branch=z9hG4bK2f14b203;rport=5060
From: "conf=500990990_491552" <sip:conf=500990990_491552@10.10.10.181>;tag=as1de6e3fc
To: <sip:0508381197@10.10.10.180>;tag=889150249
Contact: <sip:conf=500990990_491552@10.10.10.181>
Call-ID: 7342c8a8180b5fec228912600d34edd7@10.10.10.181
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 29 May 2011 21:30:10 GMT
Replaces: 544307980ae1dcd5090b3ab80b8a1da1@10.10.10.183:5060;to-tag=116609171;from-tag=as3e54850f
Require: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
X-Session-ID: 544307980ae1dcd5090b3ab80b8a1da1@10.10.10.183:5060
Content-Type: application/sdp
Content-Length: 264
X-Refer-Contact: <sip:conf=500990990_491552@10.10.10.181?Replaces=7342c8a8180b5fec228912600d34edd7@10.10.10.181%3bto-tag%3d889150249%3bfrom-tag%3das1de6e3fc>

v=0
o=root 27231 27231 IN IP4 10.10.10.181
s=session
c=IN IP4 10.10.10.181
t=0 0
m=audio 12016 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (20 headers 13 lines) ---

<--- Reliably Transmitting (NAT) to 10.10.10.180:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.10.10.180;branch=z9hG4bK80d9.dacfd6c.0;received=10.10.10.180;rport=5060
Via: SIP/2.0/UDP 10.10.10.181:5060;branch=z9hG4bK2f14b203;rport=5060
From: "conf=500990990_491552" <sip:conf=500990990_491552@10.10.10.181>;tag=as1de6e3fc
To: <sip:0508381197@10.10.10.180>;tag=889150249
Call-ID: 7342c8a8180b5fec228912600d34edd7@10.10.10.181
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:10.10.10.180:5060 --->
ACK sip:0508381197@10.10.10.183:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.180;branch=z9hG4bK80d9.dacfd6c.0
From: "conf=500990990_491552" <sip:conf=500990990_491552@10.10.10.181>;tag=as1de6e3fc
To: <sip:0508381197@10.10.10.180>;tag=889150249
Call-ID: 7342c8a8180b5fec228912600d34edd7@10.10.10.181
CSeq: 102 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Retransmitting #1 (NAT) to 10.10.10.180:5060:
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.10.10.180;branch=z9hG4bK80d9.dacfd6c.0;received=10.10.10.180;rport=5060
Via: SIP/2.0/UDP 10.10.10.181:5060;branch=z9hG4bK2f14b203;rport=5060
From: "conf=500990990_491552" <sip:conf=500990990_491552@10.10.10.181>;tag=as1de6e3fc
To: <sip:0508381197@10.10.10.180>;tag=889150249
Call-ID: 7342c8a8180b5fec228912600d34edd7@10.10.10.181
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.10.10.180:5060 --->
ACK sip:0508381197@10.10.10.183:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.180;branch=z9hG4bK80d9.dacfd6c.0
From: "conf=500990990_491552" <sip:conf=500990990_491552@10.10.10.181>;tag=as1de6e3fc
To: <sip:0508381197@10.10.10.180>;tag=889150249
Call-ID: 7342c8a8180b5fec228912600d34edd7@10.10.10.181
CSeq: 102 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Retransmitting #2 (NAT) to 10.10.10.180:5060:
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.10.10.180;branch=z9hG4bK80d9.dacfd6c.0;received=10.10.10.180;rport=5060
Via: SIP/2.0/UDP 10.10.10.181:5060;branch=z9hG4bK2f14b203;rport=5060
From: "conf=500990990_491552" <sip:conf=500990990_491552@10.10.10.181>;tag=as1de6e3fc
To: <sip:0508381197@10.10.10.180>;tag=889150249
Call-ID: 7342c8a8180b5fec228912600d34edd7@10.10.10.181
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.10.10.180:5060 --->
ACK sip:0508381197@10.10.10.183:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.180;branch=z9hG4bK80d9.dacfd6c.0
From: "conf=500990990_491552" <sip:conf=500990990_491552@10.10.10.181>;tag=as1de6e3fc
To: <sip:0508381197@10.10.10.180>;tag=889150249
Call-ID: 7342c8a8180b5fec228912600d34edd7@10.10.10.181
CSeq: 102 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Retransmitting #3 (NAT) to 10.10.10.180:5060:
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.10.10.180;branch=z9hG4bK80d9.dacfd6c.0;received=10.10.10.180;rport=5060
Via: SIP/2.0/UDP 10.10.10.181:5060;branch=z9hG4bK2f14b203;rport=5060
From: "conf=500990990_491552" <sip:conf=500990990_491552@10.10.10.181>;tag=as1de6e3fc
To: <sip:0508381197@10.10.10.180>;tag=889150249
Call-ID: 7342c8a8180b5fec228912600d34edd7@10.10.10.181
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.10.10.180:5060 --->
ACK sip:0508381197@10.10.10.183:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.180;branch=z9hG4bK80d9.dacfd6c.0
From: "conf=500990990_491552" <sip:conf=500990990_491552@10.10.10.181>;tag=as1de6e3fc
To: <sip:0508381197@10.10.10.180>;tag=889150249
Call-ID: 7342c8a8180b5fec228912600d34edd7@10.10.10.181
CSeq: 102 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
confgx00*CLI>