suite des logs
Peer audio RTP is at port 192.168.1.158:23956
Looking for 5 in users (domain 192.168.1.254)
<--- Reliably Transmitting (no NAT) to 192.168.1.158:5061 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKfc3dd1e1;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: 102 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[Jul 3 08:52:31] NOTICE[14513]: chan_sip.c:22718 handle_request_invite: Call from 'client-3' (192.168.1.158:50296) to extension '5' rejected because extension not found in context 'users'.
Scheduling destruction of SIP dialog 'a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.1.158:52349 --->
ACK 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>;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: 102 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'OTlkMGE1ODgyM2FjNzYyNWIzMTFkN2QzOGE3ZjIyZTU.' Method: REGISTER
Really destroying SIP dialog '2db2795f36d6bb4733362aad3d972fe6@192.168.1.254:50 60' Method: BYE
<--- SIP read from UDP:192.168.1.150:5060 --->
<------------->
Really destroying SIP dialog 'a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158' Method: ACK
<--- SIP read from UDP:192.168.1.150:5060 --->
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<--- SIP read from UDP:192.168.1.150:5060 --->
<------------->
<--- SIP read from UDP:192.168.1.150:5060 --->
<------------->
<--- SIP read from UDP:192.168.1.150:5060 --->
<------------->
lol*CLI>