[RESOLU] [IPPI] Appel entrant

**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

**nyko77** · 29/06/2011, 08h04

J'ai retiré le /NUM_DE_TEL, nouveau bug:

Code:

<------------->
--- (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

Envoyé par jean

et de mettre tes commandes dans ce contexte

??? j'ai pas comprit ???

**nyko77** · 30/06/2011, 16h41

???

**jean** · 30/06/2011, 17h10

=> je vois pas le pbm dans le traces, le ignoring est plutot du à une retransmission, est ce systematique ?

=> le /NUM_TEL
il faut je pense qu'il y ait une extension [NUM_TEL] dans le dialplan pour que cela marche...

Code:

; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;       register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
;

**jean** · 30/06/2011, 22h51

plus précisemment, dans le contexte de reception des appels, tu dois avoir l'extension /NUM_TEL
exten => NUM_TEL,....

**nyko77** · 01/07/2011, 09h34

Envoyé par jean

=> je vois pas le pbm dans le traces, le ignoring est plutot du à une retransmission, est ce systematique ?

Oui

Pour le NUM_TEL je l'écrit de quelle manière: +331******** ou 00331******** ou 331******** ou 01******** ???

**nyko77** · 01/07/2011, 10h02

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) ---

Discussion: [RESOLU] [IPPI] Appel entrant

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