j'ai mit 336********
quand j'appelle:
Code:
Asterisk 1.6.2.5-0ubuntu1.3, Copyright (C) 1999 - 2009 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.5-0ubuntu1.3 currently running on Sediad (pid = 741)
Verbosity was 0 and is now 7
  == Using SIP RTP CoS mark 5
Sediad*CLI>
Mode debug:
Code:
SIP Debugging enabled
Sediad*CLI> 
<--- SIP read from UDP:213.215.45.230:5060 --->
INVITE sip:331********@***.***.***.*** SIP/2.0
Record-Route: <sip:213.215.45.230;lr=on>
Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4021.e4e3eba1.0
Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK4e499720;rport=5060
From: "331********" <sip:331********@ippi.fr>;tag=as259f1013
To: <sip:01********@ippi.fr>
Contact: <sip:331********@213.215.45.252>
Call-ID: 535a09dd7b98f0d231326e4613d941ab@ippi.fr
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 12
Date: Fri, 01 Jul 2011 08:11:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 340
DID-info: 331********

v=0
o=root 17453 17453 IN IP4 213.215.45.252
s=session
c=IN IP4 213.215.45.252
t=0 0
m=audio 19328 RTP/AVP 8 0 97 3 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

<------------->
--- (16 headers 15 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 213.215.45.230 : 5060 (no NAT)
Using INVITE request as basis request - 535a09dd7b98f0d231326e4613d941ab@ippi.fr
Found peer 'ippi' for '331********' from 213.215.45.230:5060

<--- Reliably Transmitting (NAT) to 213.215.45.230:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4021.e4e3eba1.0;received=213.215.45.230
Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK4e499720;rport=5060
From: "331********" <sip:331********@ippi.fr>;tag=as259f1013
To: <sip:01********@ippi.fr>;tag=as26827315
Call-ID: 535a09dd7b98f0d231326e4613d941ab@ippi.fr
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.5-0ubuntu1.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e0ad280"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '535a09dd7b98f0d231326e4613d941ab@ippi.fr' in 6400 ms (Method: INVITE)
Sediad*CLI> 
<--- SIP read from UDP:213.215.45.230:5060 --->
ACK sip:331********@***.***.***.*** SIP/2.0
Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4021.e4e3eba1.0
From: "331********" <sip:331********@ippi.fr>;tag=as259f1013
Call-ID: 535a09dd7b98f0d231326e4613d941ab@ippi.fr
To: <sip:01********@ippi.fr>;tag=as26827315
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: OpenSIPS (1.6.3-notls (i386/linux))
Content-Length: 0


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