[RESOLU] [IPPI] Appel entrant

[nyko77]

J'ai retiré le /NUM_DE_TEL, nouveau bug:

<------------->
--- (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 - 318196c5321426a359e34d466b8a1fc0@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=z9hG4bKffa.88fe2ed6.0;received=213.215.45.230
Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK152bcced;rport=5060
From: "331********" <sip:331********@ippi.fr>;tag=as4acb879b
To: <sip:01********@ippi.fr>;tag=as5bcdcfc8
Call-ID: 318196c5321426a359e34d466b8a1fc0@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="5b0301d7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '318196c5321426a359e34d466b8a1fc0@ippi.fr' in 6400 ms (Method: INVITE)

<--- 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=z9hG4bKffa.88fe2ed6.1
Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK152bcced;rport=5060
From: "331********" <sip:331********@ippi.fr>;tag=as4acb879b
To: <sip:01********@ippi.fr>
Contact: <sip:331********@213.215.45.252>
Call-ID: 318196c5321426a359e34d466b8a1fc0@ippi.fr
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 12
Date: Wed, 29 Jun 2011 06:14:11 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 11904 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) ---
Ignoring this INVITE request

et de mettre tes commandes dans ce contexte

??? j'ai pas comprit ???