Bonjour,

J'ai déconnecté tous les postes sauf le SPA 3102 qui visiblement ne pollue pas trop le log, et voici tout ce que j'ai pu copier

From: <sip:5520@192.168.1.220>;tag=as76046a24
To: "5501"<sip:5501@192.168.1.220>;tag=e6894cea
Contact: <sip:5520@192.168.1.220:5060>
Call-ID: YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.6.0
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=z9hG4bK36ef104f
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 104 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: 202

v=0
o=root 752446342 752446344 IN IP4 192.168.1.11
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.1.11
t=0 0
m=audio 59984 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=z9hG4bK36ef104f
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 104 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=z9hG4bK36ef104f
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 104 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=z9hG4bK7ee8a187
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:506 0
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.11:45996 --->
BYE sip:5520@192.168.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45996;branch=z9hG4bK-d8754z-2173cd01d15af29a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:5501@192.168.1.11:45996;transport=udp>
To: <sip:5520@192.168.1.220>;tag=as76046a24
From: "5501"<sip:5501@192.168.1.220>;tag=e6894cea
Call-ID: YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.
CSeq: 3 BYE
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="5501",realm="asterisk",nonce="75510cc6", uri="sip:5520@192.168.1.220:5060",response="3e6e4c 61dbb314c9ae372b80e3f83b20",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.11:45996 (no NAT)
Scheduling destruction of SIP dialog 'YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.1.11:45996 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:45996;branch=z9hG4bK-d8754z-2173cd01d15af29a-1---d8754z-;received=192.168.1.11;rport=45996
From: "5501"<sip:5501@192.168.1.220>;tag=e6894cea
To: <sip:5520@192.168.1.220>;tag=as76046a24
Call-ID: YTgzNzgwOWZjYmIyOWEwMmUzZjFjMjg3MTRmODcxMDg.
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=z9hG4bK12711fef
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@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: 204

v=0
o=root 752446342 752446345 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 11108 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=z9hG4bK12711fef
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@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=z9hG4bK12711fef
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@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=z9hG4bK5ab0f8d3
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Contact: <sip:5501@192.168.1.220:5060>
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@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 '498087fa6ab8d6634d5f1b8a42215b76@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=z9hG4bK19c8feec
Max-Forwards: 70
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@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/5501-0000002c'
== Extension Changed 5501[subscriptions] new state Idle for Notify User 5503 (queued)

<--- 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=z9hG4bK19c8feec
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@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=z9hG4bK19c8feec
From: "PC Greg" <sip:5501@192.168.1.220>;tag=as7e4a5478
To: <sip:5520@192.168.1.151:5060>;tag=1292519073
Call-ID: 498087fa6ab8d6634d5f1b8a42215b76@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 '498087fa6ab8d6634d5f1b8a42215b76@192.168.1.220:50 60' Method: INVITE