Voila le résultat, je me connecte en ssh avec putty c'est le max que je puisse extraire:

--- (14 headers 0 lines) ---
Sending to 192.168.1.41:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.1.41:5060:
OPTIONS sip:5503@192.168.1.41:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK3adf2a63
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.220>;tag=as7ac5f647
To: <sip:5503@192.168.1.41:5060;transport=udp>
Contact: <sip:asterisk@192.168.1.220:5060>
Call-ID: 12cf549f6f453b9b134b5da37a7db21d@192.168.1.220:506 0
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.6.0
Date: Tue, 15 Nov 2011 19:04:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.1.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.41;branch=z9hG4bK843c192f58aaf3a64;recei ved=192.168.1.41
From: <sip:5503@192.168.1.220:5060>;tag=09aa42ea0c
To: <sip:5503@192.168.1.220:5060>;tag=as042c4e18
Call-ID: bf0fa8af85b3dda3
CSeq: 100984 REGISTER
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: <sip:5503@192.168.1.41:5060;transport=udp>;expires =120
Date: Tue, 15 Nov 2011 19:04:36 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'bf0fa8af85b3dda3' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK3adf2a63
From: "asterisk" <sip:asterisk@192.168.1.220>;tag=as7ac5f647
To: <sip:5503@192.168.1.41:5060;transport=udp>;tag=362 1404576
Call-ID: 12cf549f6f453b9b134b5da37a7db21d@192.168.1.220:506 0
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 55i/3.2.0.1011
Supported: path
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '12cf549f6f453b9b134b5da37a7db21d@192.168.1.220:50 60' Method: OPTIONS

<--- SIP read from UDP:192.168.1.11:45688 --->
BYE sip:5520@192.168.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45688;branch=z9hG4bK-d8754z-d464ac1d05087289-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:5501@192.168.1.11:45688;transport=udp>
To: <sip:5520@192.168.1.220>;tag=as1606e0bb
From: "5501"<sip:5501@192.168.1.220>;tag=ce578432
Call-ID: MmQxNGNiMGEwYjFiZDM2MjQ3MzViNjA3ZDM3MTEwMGY.
CSeq: 3 BYE
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="5501",realm="asterisk",nonce="61f12305", uri="sip:5520@192.168.1.220:5060",response="184bcc 94cc53ebec83743cb0860427d7",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.11:45688 (no NAT)
Scheduling destruction of SIP dialog 'MmQxNGNiMGEwYjFiZDM2MjQ3MzViNjA3ZDM3MTEwMGY.' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.1.11:45688 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:45688;branch=z9hG4bK-d8754z-d464ac1d05087289-1---d8754z-;received=192.168.1.11;rport=45688
From: "5501"<sip:5501@192.168.1.220>;tag=ce578432
To: <sip:5520@192.168.1.220>;tag=as1606e0bb
Call-ID: MmQxNGNiMGEwYjFiZDM2MjQ3MzViNjA3ZDM3MTEwMGY.
CSeq: 3 BYE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
set_destination: set destination to 192.168.1.151:5060
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.1.151:5060:
INVITE sip:5520@192.168.1.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK1f095ee8
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
CSeq: 105 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 203

v=0
o=root 993097821 993097824 IN IP4 192.168.1.220
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.1.220
t=0 0
m=audio 7872 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.151:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK1f095ee8
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
CSeq: 105 INVITE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.151:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK1f095ee8
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
CSeq: 105 INVITE
Contact: <sip:5520@192.168.1.151:5060>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Type: application/sdp
Content-Length: 179

v=0
o=userX 20000001 20000001 IN IP4 192.168.1.151
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.1.151
t=0 0
m=audio 10500 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
<------------->
--- (10 headers 8 lines) ---
set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
set_destination: set destination to 192.168.1.151:5060
Transmitting (no NAT) to 192.168.1.151:5060:
ACK sip:5520@192.168.1.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK49f26b0c
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
CSeq: 105 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---
Scheduling destruction of SIP dialog '219adec357d78ce34c2273335129e215@192.168.1.220:50 60' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:5520@192.168.1.151:5060> for address/port to send to
set_destination: set destination to 192.168.1.151:5060
Reliably Transmitting (no NAT) to 192.168.1.151:5060:
BYE sip:5520@192.168.1.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK33cc2e46
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
CSeq: 106 BYE
User-Agent: Asterisk PBX 1.8.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
== Spawn extension (phones, 5520, 2) exited non-zero on 'SIP/5501-00000019'
set_destination: Parsing <sip:5503@192.168.1.41:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.1.41:5060
Reliably Transmitting (no NAT) to 192.168.1.41:5060:
NOTIFY sip:5503@192.168.1.41:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK73d30871
Max-Forwards: 70
From: <sip:5501@192.168.1.220:5060>;tag=as77c8c620
To: <sip:5503@192.168.1.220:5060>;tag=560a44aa4f
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 4baa7bb7f47d8d4c
Seq: 164 NOTIFY
User-Agent: Asterisk PBX 1.8.6.0
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 209

<?xml version="1.0"?><dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="62" state="full" entity="sip:5501@192.168.1.220:5060"><dialog id="5501"><state>terminated</state>
</dialog>
</dialog-info>

---
== Extension Changed 5501[subscriptions] new state Idle for Notify User 5503

<--- SIP read from UDP:192.168.1.151:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK33cc2e46
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
CSeq: 106 BYE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.151:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK33cc2e46
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as516d31b0
To: <sip:5520@192.168.1.151:5060>;tag=393068593
Call-ID: 219adec357d78ce34c2273335129e215@192.168.1.220:506 0
CSeq: 106 BYE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '219adec357d78ce34c2273335129e215@192.168.1.220:50 60' Method: INVITE

<--- SIP read from UDP:192.168.1.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK73d30871
From: <sip:5501@192.168.1.220:5060>;tag=as77c8c620
To: <sip:5503@192.168.1.220:5060>;tag=560a44aa4f
Call-ID: 4baa7bb7f47d8d4c
CSeq: 164 NOTIFY
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Server: Aastra 55i/3.2.0.1011
Supported: path
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
serveur*CLI> sip set debug off
SIP Debugging Disabled


C'est un peu obscure pour moi, si ça vous parle n’hésitez pas a m'éclairer!!!!
Merci