J'ai fais comme tu m'a dis, j'ai mis allowguest = no dans general
puis tapez les commandes ...
J'espère que c'est les logs attendu ...

<--- 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=z9hG4bKc2be2d2a;received =192.168.1.158
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a5620017ccbc8736-cc8044ca
To: <sip:5@192.168.1.254>;tag=as31595924
Call-ID: a40cc394-a5620011-2b015b06-6de2c5da@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 2 17:44:40] NOTICE[12769]: chan_sip.c:22718 handle_request_invite: Call from 'client-3' (192.168.1.158:52726) to extension '5' rejected because extension not found in context 'users'.
Scheduling destruction of SIP dialog 'a40cc394-a5620011-2b015b06-6de2c5da@192.168.1.158' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.158:51110 --->
ACK sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKc2be2d2a
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a5620017ccbc8736-cc8044ca
To: <sip:5@192.168.1.254>;tag=as31595924
Call-ID: a40cc394-a5620011-2b015b06-6de2c5da@192.168.1.158
Max-Forwards: 70
Date: Tue, 02 Jul 2013 15:44:37 GMT
CSeq: 102 ACK
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'a40cc394-a5620011-2b015b06-6de2c5da@192.168.1.158' Method: ACK