[RESOLU] [IPPI] Appel entrant

**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) ---

**nyko77** · 01/07/2011, 10h12

Je remet mon code actuelle:

sip.conf:

Code:

[general]
defaultexpirey=1800
dtmfmode=auto
qualify=yes
register => UTILISATEUR:MOT_DE_PASSE@ippi.fr/NUM_TEL
;NUM_TEL ecrit 331********

externip=sediad.homelinux.org
externrefresh=60
localnet=192.168.1.0/255.255.255.0

[ippi]
type=friend
host=ippi.fr
username=UTILISATEUR
secret=MOT_DE_PASSE
fromuser=UTILISTATEUR
fromdomain=ippi.fr
nat=yes
canreinvite=no
insecure=very
context=fromippi

[sediad1]
type=friend
secret=MOT_DE_PASSE
host=dynamic
context=home
nat=yes

[sediad2]
type=friend
secret=MOT_DE_PASSE
host=dynamic
context=home
nat=yes

extensions.conf;

Code:

[fromippi]
exten => _+.,1,Verbose(1, *** International Formati : 00${EXTEN:1})
exten => s,n,Dial(SIP/sediad1)

[home]
exten => 001,1,Dial(SIP/sediad1)
exten => 002,1,Dial(SIP/sediad2)
exten => _X.,1,Dial(SIP/ippi/${EXTEN})

**vincent10** · 01/07/2011, 12h54

Bonjour,

Voici comment j'ai configuré chez moi pour ippi.

Code:

[ippi_incoming] ; configuration des appels entrants depuis ippi
type=peer
host=ippi.fr
context=context

Code:

[ippi_appel-sortant-ippi] ; configuration des appels sortants par ippi
type=peer
host=ippi.fr
username=username
secret=mdp
fromuser=user
fromdomain=ippi.fr
nat=yes
insecure=port,invite

Voilà chez moi tout fonctionne.

**nyko77** · 03/07/2011, 21h31

Merci vincent10

j'ai avancé d'un pas, je ne pence pas être loin de la vérité:
Lorsque j'appelle j'ai l'erreur suivante:

Code:

  == Using SIP RTP CoS mark 5
[Jul  2 20:25:47] NOTICE[873]: chan_sip.c:20039 handle_request_invite: Call from 'ippi_incoming' to extension '331********' rejected because extension not found.

Donc, j'ai donc retirer l’extension dans "register", sa me donne donc:

Code:

register => UTILISATEUR:MOT_DE_PASSE@ippi.fr

j'ai donc l'erreur suivante:

Code:

  == Using SIP RTP CoS mark 5
[Jul  3 21:18:59] NOTICE[995]: chan_sip.c:20039 handle_request_invite: Call from 'ippi_incoming' to extension 's' rejected because extension not found.

**vincent10** · 04/07/2011, 07h43

Bonjour,

Pouvez vous mettre votre fichier extensions.conf.

**nyko77** · 04/07/2011, 08h02

Mon extensions.conf :

Code:

[fromippi]
exten => s,1,Dial(SIP/sediad1)

[home]
exten => 001,1,Dial(SIP/sediad1)
exten => 002,1,Dial(SIP/sediad2)
exten => _0.,1,Dial(SIP/ippi_appel-sortant-ippi/${EXTEN})

**nyko77** · 04/07/2011, 09h39

Sa me donne :

Code:

 == Using SIP RTP CoS mark 5
[Jul  4 09:45:56] NOTICE[892]: chan_sip.c:20039 handle_request_invite: Call from 'ippi_incoming' to extension '01********' rejected because extension not found.

sip.conf :

Code:

[general]
defaultexpirey=1800
dtmfmode=auto
qualify=yes
register => UTILISATEUR:MOT_DE_PASSE@ippi.fr/01********

externip=sediad.homelinux.org
externrefresh=60
localnet=192.168.1.0/255.255.255.0

[ippi_incoming] ; configuration des appels entrants depuis ippi
type=peer
host=ippi.fr
context=context

[ippi_appel-sortant-ippi] ; configuration des appels sortants par ippi
type=peer
host=ippi.fr
username=UTILISATEUR
secret=MOT_DE_PASSE
fromuser=UTILISATEUR
fromdomain=ippi.fr
nat=yes
insecure=port,invite

[sediad1]
type=friend
secret=mot_de_passe
host=dynamic
context=home
nat=yes

[sediad2]
type=friend
secret=mot_de_passe
host=dynamic
context=home
nat=yes

extensions.conf :

Code:

[fromippi]
exten => 01********,1,Ringing(1)
exten => 01********,n,Wait(3)
exten => 01********,n,Answer
exten => 01********,n,Dial(SIP/sediad1)

[home]
exten => 001,1,Dial(SIP/sediad1)
exten => 002,1,Dial(SIP/sediad2)
exten => _00.,1,Dial(SIP/ippi_appel-sortant-ippi/${EXTEN})

**vincent10** · 04/07/2011, 10h10

Dans [ippi_incoming]
le context est bien fromippi?

**nyko77** · 04/07/2011, 10h31

Envoyé par vincent10

Dans [ippi_incoming]
le context est bien fromippi?

OUPS...

Sa fonctionne nikel!!!

Merci beaucoup

Discussion: [RESOLU] [IPPI] Appel entrant

Outils de la discussion

Rechercher dans la discussion

Affichage

Vue hybride

Les tags pour cette discussion

Règles de messages