[RESOLU] [IPPI] Appel entrant

**jean** · 28/06/2011, 23h01

ok.... tu as tenté un appel entrant pendant que tu prenais ces traces ?

si oui, c'est que les paquets de IPPI n'arrivent pas à ton pc... as tu ouvert le firewall de ta box ?

**nyko77** · 28/06/2011, 23h17

Envoyé par jean

ok.... tu as tenté un appel entrant pendant que tu prenais ces traces ?

si oui, c'est que les paquets de IPPI n'arrivent pas à ton pc... as tu ouvert le firewall de ta box ?

Code:

# sudo ngrep -d eth0 port 5060 and host ippi.fr
interface: eth0 (192.168.1.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 and host ippi.fr )
#
U 213.215.45.230:5060 -> 192.168.1.100:5060
  INVITE sip:+33177*****@***.***.***.*** SIP/2.0..Record-Route: <sip:213.215.
  45.230;lr=on>..Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bKa825.1098d646.
  0..Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG
  4bK51aaea0b;rport=5060..From: "331********" <sip:331********@ippi.fr>;tag=a
  s202fa92f..To: <sip:01********@ippi.fr>..Contact: <sip:331********@213.215.
  45.252>..Call-ID: 354024bf6a5a4abf4aa75bae57ada3cb@ippi.fr..CSeq: 102 INVIT
  E..User-Agent: Asterisk PBX..Max-Forwards: 12..Date: Tue, 28 Jun 2011 21:19
  :18 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 13466 RTP/AVP 8 0 97 3 18 101..a=rtpmap:8 PCMA/8
  000..a=rtpmap:0 PCMU/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:3 GSM/8000..a=rt
  pmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..
  a=fmtp:101 0-16..a=silenceSupp:off - - - -..                               
#
U 192.168.1.100:5060 -> 213.215.45.230:5060
  SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bKa82
  5.1098d646.0;received=213.215.45.230..Via: SIP/2.0/UDP 213.215.45.252:5060;
  received=213.215.45.252;branch=z9hG4bK51aaea0b;rport=5060..From: "331********" <sip:331********@ippi.fr>;tag=as202fa92f..To: <sip:01********@ippi.fr>;
  tag=as7875e22a..Call-ID: 354024bf6a5a4abf4aa75bae57ada3cb@ippi.fr..CSeq: 10
  2 INVITE..Server: Asterisk PBX 1.6.2.5-0ubuntu1.3..Allow: INVITE, ACK, CANC
  EL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces, time
  r..WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b8c8aa
  6"..Content-Length: 0....                                                  
#
U 213.215.45.230:5060 -> 192.168.1.100:5060
  ACK sip:+331********@84.100.202.211 SIP/2.0..Via: SIP/2.0/UDP 213.215.45.23
  0;branch=z9hG4bKa825.1098d646.0..From: "331********" <sip:331********@ippi.
  fr>;tag=as202fa92f..Call-ID: 354024bf6a5a4abf4aa75bae57ada3cb@ippi.fr..To: 
  <sip:01********@ippi.fr>;tag=as7875e22a..CSeq: 102 ACK..Max-Forwards: 70..U
  ser-Agent: OpenSIPS (1.6.3-notls (i386/linux))..Content-Length: 0....

Donc les infos arrive bien au pc. Ma neufbox a le port 5060 ouvert
Mais rien de plus au niveau téléphonique....

**jean** · 28/06/2011, 23h31

ok... effectivement, ca cause avec ton asterisk, c'est déjà ça :-)

next... pousse le détail sur la CLI, doit y a avoir qque chose...
core set verbose 9
core set debug 9

ensuite, .. as tu une extension _+. dans ton contexte ? vu que les numéros arrivent avec ce format ?

**jean** · 28/06/2011, 23h34

btw.... vu que le port 5060 est ouvert.... soit ton firewall autorise exclusivment ippi.fr a te contacter (si l'@ est constante), soit il te faut imperativement fail2ban, sinon, tu vas pleurer... (fraude)

**nyko77** · 29/06/2011, 00h03

Envoyé par jean

ok... effectivement, ca cause avec ton asterisk, c'est déjà ça :-)

next... pousse le détail sur la CLI, doit y a avoir qque chose...
core set verbose 9
core set debug 9

ensuite, .. as tu une extension _+. dans ton contexte ? vu que les numéros arrivent avec ce format ?

Code:

*CLI> core set verbose 9
Verbosity was 7 and is now 9
*CLI> core set debug 9
Core debug was 0 and is now 9

Comment fait-on une extention_+. ?????

**jean** · 29/06/2011, 00h14

ben... comme les autres....

exten => _+.,1,Verbose(1, *** International Formati : 00${EXTEN:1})
exten => _+.,n,Goto(00${EXTEN:1},1)

(pour par exemple remplacer le + par 00 mais ce n'est peut etre pas ce que tu recherches)

**nyko77** · 29/06/2011, 00h33

J'ai activer "sip set debug" quand j'ai appellai, il marque:

Code:

<--- 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=z9hG4bK4a2.b4561c11.0
Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK2ca377dc;rport=5060
From: "336********" <sip:336********@ippi.fr>;tag=as24278228
To: <sip:01********@ippi.fr>
Contact: <sip:336********@213.215.45.252>
Call-ID: 15594c9b4ae5bf865302279c7c0cac77@ippi.fr
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 12
Date: Tue, 28 Jun 2011 22:36:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 340
DID-info: 336********

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 18138 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 - 15594c9b4ae5bf865302279c7c0cac77@ippi.fr
Found peer 'ippi' for '336********' 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=z9hG4bK4a2.b4561c11.0;received=213.215.45.230
Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK2ca377dc;rport=5060
From: "336********" <sip:336********@ippi.fr>;tag=as24278228
To: <sip:01********@ippi.fr>;tag=as0f438622
Call-ID: 15594c9b4ae5bf865302279c7c0cac77@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="79decfb5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '15594c9b4ae5bf865302279c7c0cac77@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=z9hG4bK4a2.b4561c11.0
From: "336********" <sip:336********@ippi.fr>;tag=as24278228
Call-ID: 15594c9b4ae5bf865302279c7c0cac77@ippi.fr
To: <sip:01********@ippi.fr>;tag=as0f438622
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: OpenSIPS (1.6.3-notls (i386/linux))
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Sediad*CLI> 
<--- SIP read from UDP:192.168.1.77:61198 --->



<------------->
Reliably Transmitting (NAT) to 192.168.1.77:61198:
OPTIONS sip:tel1@192.168.1.***:61198;rinstance=54f821032d61eaf2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK12134893;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.100>;tag=as7544ff20
To: <sip:tel1@192.168.1.***:61198;rinstance=54f821032d61eaf2>
Contact: <sip:asterisk@192.168.1.100>
Call-ID: 54a2557109e378b826294e30427c4caa@192.168.1.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.3
Date: Tue, 28 Jun 2011 22:36:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Sediad*CLI> 
<--- SIP read from UDP:192.168.1.***:61198 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK12134893;rport=5060
Contact: <sip:192.168.1.***:61198>
To: <sip:tel1@192.168.1.**:61198;rinstance=54f821032d61eaf2>;tag=209c83b7
From: "asterisk"<sip:asterisk@192.168.1.100>;tag=as7544ff20
Call-ID: 54a2557109e378b826294e30427c4caa@192.168.1.100
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '54a2557109e378b826294e30427c4caa@192.168.1.100' Method: OPTIONS
Really destroying SIP dialog '15594c9b4ae5bf865302279c7c0cac77@ippi.fr' Method: ACK

Mais sans plus

**jean** · 29/06/2011, 02h33

asterisk balance un unauthorized....

next... dans le register,
register => UTILISATEUR:MOT_DE_PASSE@ippi.fr/NUM_DE_TEL

le /NUM_DE_TEL designe l'extension... essaie de virer le /num_tel, ou de voir à quoi correspond cette extension (dans le sip ou extensions.conf), et de mettre tes commandes dans ce contexte

Discussion: [RESOLU] [IPPI] Appel entrant

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