<--- SIP read from UDP://192.168.200.60:5060 --->
INVITE sip:40@192.168.200.240 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.60:5060;branch=z9hG4bKc37e30da9e916e67c
Max-Forwards: 70
From: "" <sip:41@192.168.200.240>;tag=f2f312b3d7
To: "40" <sip:40@192.168.200.240>
Call-ID: dedf6720455b343d
CSeq: 21616 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: <sip:41@192.168.200.60:5060;transport=udp>
Session-Expires: 90
Supported: timer, 100rel, replaces
User-Agent: Aastra 57i/2.2.1.25
Content-Type: application/sdp
Content-Length: 261

v=0
o=MxSIP 0 0 IN IP4 192.168.200.60
s=SIP Call
c=IN IP4 192.168.200.60
t=0 0
m=audio 3000 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 13 lines) ---
Sending to 192.168.200.60 : 5060 (NAT)
Using INVITE request as basis request - dedf6720455b343d
Found user '41' for '41'

<--- Reliably Transmitting (no NAT) to 192.168.200.60:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.200.60:5060;branch=z9hG4bKc37e30da9e916e67c;received=192.168.200.60
From: "" <sip:41@192.168.200.240>;tag=f2f312b3d7
To: "40" <sip:40@192.168.200.240>;tag=as3f111eb5
Call-ID: dedf6720455b343d
CSeq: 21616 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="314b57c6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'dedf6720455b343d' in 32000 ms (Method: INVITE)
asterisk1*CLI> 
<--- SIP read from UDP://192.168.200.60:5060 --->
ACK sip:40@192.168.200.240 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.60:5060;branch=z9hG4bKc37e30da9e916e67c
Max-Forwards: 70
From: "" <sip:41@192.168.200.240>;tag=f2f312b3d7
To: "40" <sip:40@192.168.200.240>;tag=as3f111eb5
Call-ID: dedf6720455b343d
CSeq: 21616 ACK
User-Agent: Aastra 57i/2.2.1.25
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
asterisk1*CLI> 
<--- SIP read from UDP://192.168.200.60:5060 --->
INVITE sip:40@192.168.200.240 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.60:5060;branch=z9hG4bK17cf9af5cadad5e69
Max-Forwards: 70
From: "" <sip:41@192.168.200.240>;tag=f2f312b3d7
To: "40" <sip:40@192.168.200.240>
Call-ID: dedf6720455b343d
CSeq: 21617 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Authorization: Digest username="41",realm="asterisk",nonce="314b57c6",uri="sip:40@192.168.200.240",response="f0c2df803a4b8b7fbf21b4eb6416adf9",algorithm=MD5
Contact: <sip:41@192.168.200.60:5060;transport=udp>
Session-Expires: 90
Supported: timer, 100rel, replaces
User-Agent: Aastra 57i/2.2.1.25
Content-Type: application/sdp
Content-Length: 261

v=0
o=MxSIP 0 0 IN IP4 192.168.200.60
s=SIP Call
c=IN IP4 192.168.200.60
t=0 0
m=audio 3000 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (16 headers 13 lines) ---
Sending to 192.168.200.60 : 5060 (no NAT)
Using INVITE request as basis request - dedf6720455b343d
Found user '41' for '41'
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.200.60:3000
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x1000 (g722), peer - audio=0x1008 (alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1000 (g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.200.60:3000
Looking for 40 in from-internal (domain 192.168.200.240)
list_route: hop: <sip:41@192.168.200.60:5060;transport=udp>

<--- Transmitting (no NAT) to 192.168.200.60:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.60:5060;branch=z9hG4bK17cf9af5cadad5e69;received=192.168.200.60
From: "" <sip:41@192.168.200.240>;tag=f2f312b3d7
To: "40" <sip:40@192.168.200.240>
Call-ID: dedf6720455b343d
CSeq: 21617 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:40@192.168.200.240>
Content-Length: 0


<------------>
 recordingcheck,20080709-191455,1215623695.0: Inbound recording not enabled
 dialparties.agi: Starting New Dialparties.agi
 dialparties.agi: Caller ID name is 'Poste 41' number is '41'
 dialparties.agi: Methodology of ring is  'none'
asterisk1*CLI> 
<--- Transmitting (no NAT) to 192.168.200.60:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.200.60:5060;branch=z9hG4bK17cf9af5cadad5e69;received=192.168.200.60
From: "" <sip:41@192.168.200.240>;tag=f2f312b3d7
To: "40" <sip:40@192.168.200.240>;tag=as6ebb1358
Call-ID: dedf6720455b343d
CSeq: 21617 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:40@192.168.200.240>
Content-Length: 0


<------------>
Audio is at 192.168.200.240 port 12246
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.200.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.60:5060;branch=z9hG4bK17cf9af5cadad5e69;received=192.168.200.60
From: "" <sip:41@192.168.200.240>;tag=f2f312b3d7
To: "40" <sip:40@192.168.200.240>;tag=as6ebb1358
Call-ID: dedf6720455b343d
CSeq: 21617 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:40@192.168.200.240>
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 356385367 356385367 IN IP4 192.168.200.240
s=Asterisk PBX 1.6.0-beta9
c=IN IP4 192.168.200.240
t=0 0
m=audio 12246 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
asterisk1*CLI> 
<--- SIP read from UDP://192.168.200.60:5060 --->
ACK sip:40@192.168.200.240 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.60:5060;branch=z9hG4bKcc41865107d91fc30
Max-Forwards: 70
From: "" <sip:41@192.168.200.240>;tag=f2f312b3d7
To: "40" <sip:40@192.168.200.240>;tag=as6ebb1358
Call-ID: dedf6720455b343d
CSeq: 21617 ACK
Authorization: Digest username="41",realm="asterisk",nonce="314b57c6",uri="sip:40@192.168.200.240",response="702533543fa81c27ae1f710a73a38bf3",algorithm=MD5
User-Agent: Aastra 57i/2.2.1.25
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog 'dedf6720455b343d' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:41@192.168.200.60:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.200.60, port 5060
Reliably Transmitting (no NAT) to 192.168.200.60:5060:
BYE sip:41@192.168.200.60:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.200.240:5060;branch=z9hG4bK559315f5;rport
Max-Forwards: 70
From: "40" <sip:40@192.168.200.240>;tag=as6ebb1358
To: "" <sip:41@192.168.200.240>;tag=f2f312b3d7
Call-ID: dedf6720455b343d
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.0-beta9
Content-Length: 0


---
asterisk1*CLI> 
<--- SIP read from UDP://192.168.200.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.240:5060;branch=z9hG4bK559315f5;rport=5060;received=192.168.200.240
From: "40" <sip:40@192.168.200.240>;tag=as6ebb1358
To: "" <sip:41@192.168.200.240>;tag=f2f312b3d7
Call-ID: dedf6720455b343d
CSeq: 102 BYE
Server: Aastra 57i/2.2.1.25
Content-Length: 0