suite des logs

set_destination: Parsing <sip:client-1@192.168.1.150:5060> for address/port to send to
set_destination: set destination to 192.168.1.150:5060
Transmitting (NAT) to 192.168.1.150:5060:
ACK sip:client-1@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK767cf41d;rport
Max-Forwards: 70
From: <sip:557@192.168.1.254:5060>;tag=as16824010
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
Contact: <sip:557@192.168.1.254:5060>
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Content-Length: 0


---
== Spawn extension (users, 557, 1) exited non-zero on 'SIP/client-1-00000000'
Scheduling destruction of SIP dialog 'YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:client-1@192.168.1.150:5060> for address/port to send to
set_destination: set destination to 192.168.1.150:5060
Reliably Transmitting (NAT) to 192.168.1.150:5060:
BYE sip:client-1@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK42bf98f8;rport
Max-Forwards: 70
From: <sip:557@192.168.1.254:5060>;tag=as16824010
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Proxy-Authorization: Digest username="client-1", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.254", nonce="", response="323d061907b0821b1decc513156e9f67"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK42bf98f8;rport=50 60
Contact: <sip:client-1@192.168.1.150:5060>
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
From: <sip:557@192.168.1.254:5060>;tag=as16824010
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 104 BYE
User-Agent: X-Lite release 5.0.0 stamp 67284
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.' Method: ACK

<--- SIP read from UDP:192.168.1.150:5060 --->


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

<--- SIP read from UDP:192.168.1.158:50296 --->
INVITE sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK386bb6e8
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:20 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7942G/9.3.1
Contact: <sip:557@192.168.1.158:5061;transport=udp>
Expires: 180
Accept: application/sdp
llow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTE R,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 354
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 12876 0 IN IP4 192.168.1.158
s=SIP Call
t=0 0
m=audio 23956 RTP/AVP 0 8 18 102 116 101
c=IN IP4 192.168.1.158
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (18 headers 16 lines) ---
Sending to 192.168.1.158:50296 (NAT)
Using INVITE request as basis request - a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Found peer 'client-3' for 'client-3' from 192.168.1.158:50296

<--- Reliably Transmitting (no NAT) to 192.168.1.158:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK386bb6e8;received =192.168.1.158
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>;tag=as7134625c
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02702734"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.158:53122 --->
ACK sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK386bb6e8
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>;tag=as7134625c
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:20 GMT
CSeq: 101 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.158:50296 --->
INVITE sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKfc3dd1e1
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:20 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7942G/9.3.1
Contact: <sip:557@192.168.1.158:5061;transport=udp>
Authorization: Digest username="client-3",realm="asterisk",uri="sip:5@192.168.1.254;user= phone",response="61db6f5bf1b4ac464547bc3903ead800" ,nonce="02702734",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTE R,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 354
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 12876 0 IN IP4 192.168.1.158
s=SIP Call
t=0 0
m=audio 23956 RTP/AVP 0 8 18 102 116 101
c=IN IP4 192.168.1.158
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (19 headers 16 lines) ---
Sending to 192.168.1.158:5061 (no NAT)
Using INVITE request as basis request - a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Found peer 'client-3' for 'client-3' from 192.168.1.158:50296
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 102
Found RTP audio format 116
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 G729 for ID 18
Found audio description format L16 for ID 102
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x850c (ulaw|alaw|g729|ilbc|slin16)/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|)