[RESOLU] Problème dial plan asterisk ligne free

[Reaper]

Pouvez vous me donner un acess root sur votre machine ?
J'ai pas le temps de deviner, ça doit être quelque chose très simple.
Ou sinon vous pouvez effacer tout, créer un peer Free, et un compte de téléphone, et essayer appeler par ce peer. Je vous invite de re-faire votre installation, de trouver un tutoriel sip free (des tonnes sur internet) et le suivre

[LeRenard]

Pouvez vous me donner un acess root sur votre machine ?
J'ai pas le temps de deviner, ça doit être quelque chose très simple.
Ou sinon vous pouvez effacer tout, créer un peer Free, et un compte de téléphone, et essayer appeler par ce peer. Je vous invite de re-faire votre installation, de trouver un tutoriel sip free (des tonnes sur internet) et le suivre

Euh, non, ca ne vas pas être possible ! LOL
Merci quand même de ton aide Reaper !

[LeRenard]

Bonjour !


Après quelques recherches et modifications personnelles, j'ai bien avancé !
En effet, j'ai réussi à faire fonctionner certaines communications (voir ci-dessous).
Cependant, il me reste un petit problème au niveau du numéro free...comme vous pouvez le voir ci-dessous également:

0950XXXXXX vers 201 = OK ! =)
0950XXXXXX vers 202 = OK ! =)
0950XXXXXX vers 0950XXXXXX = FAIL ! :(

201 vers 202 = OK ! =)
201 vers 201 = OK ! =)
201 vers 0950XXXXXX = FAIL ! :(

202 vers 201 = OK ! =)
202 vers 202 = OK ! =)
202 vers 0950XXXXXX = FAIL ! :(



Voici le résultat dans la CLI:

SRVVOIP1*CLI> sip set debug peer 0950XXXXXX
SIP Debugging Enabled for IP: 212.27.52.5:5060
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 82.X.X.X:5060;branch=z9hG4bK1f55abcb;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@DOMAIN>;tag=as78adad8c
To: <sip:freephonie.net>
Contact: <sip:Unknown@82.X.X.X>
Call-ID: 1bf6409a4e27c9cd078a9d5f5722552a@DOMAIN
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Tue, 20 Sep 2011 13:46:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 501 Not Implemented Yet
Call-ID: 1bf6409a4e27c9cd078a9d5f5722552a@DOMAIN
CSeq: 102 OPTIONS
From: "Unknown" <sip:Unknown@DOMAIN>;tag=as78adad8c
To: <sip:freephonie.net>;tag=00-31101-1ed40e68-735604160
Via: SIP/2.0/UDP 82.X.X.X:5060;received=82.X.X.X;rport=1024;branch= z9hG4bK1f55abcb
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '1bf6409a4e27c9cd078a9d5f5722552a@DOMAIN' Method: OPTIONS
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] Ringing("SIP/202-00000059", "") in new stack
-- Executing [90950XXXXXX@appel_interne_X:2] Wait("SIP/202-00000059", "") in new stack
-- Executing [90950XXXXXX@appel_interne_X:3] Answer("SIP/202-00000059", "") in new stack
-- Executing [90950XXXXXX@appel_interne_X:4] Dial("SIP/202-00000059", "SIP/0950XXXXXX@freephonie.net),30,r") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Everyone is busy/congested at this time (1:0/0/1)
SRVVOIP1*CLI>

Le problème est bien au niveau de:
== Everyone is busy/congested at this time (1:0/0/1)

Avez-vous une idée ?! Merci d'avance.

[LeRenard]

Pour information:

J'ai fais un test en appelant le numéro free 0950XXXXXX via mon téléphone portable 06XXXXXXXX. Voici le résultat que la CLI m'a sorti:

SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
INVITE sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 23528-QA-102f2397-369c648f4@freephonie.net
Contact: <sip:172.X.X.X:5062>
Content-Type: application/sdp
CSeq: 264484018 INVITE
From: "0674420004" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=235 28-XD-102f2398-54dd3b156
Max-Forwards: 27
Record-Route: <sip:C=on;t=UBFVF@212.27.52.5:5060;lr>
To: <sip:0950XXXXXX@172.X.X.X;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-UBFV-0e04cd9b-1b10a8d8
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.42q (gw_sip)
Content-Length: 184

v=0
o=cp10 131659623199 131659623199 IN IP4 172.X.X.X
s=SIP Call
c=IN IP4 212.27.52.129
t=0 0
m=audio 31122 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 10 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 212.27.52.5 : 5060 (NAT)
Using INVITE request as basis request - 23528-QA-102f2397-369c648f4@freephonie.net
No user '06XXXXXXXX' in SIP users list
Found peer '0950XXXXXX-in' for '06XXXXXXXX' from 212.27.52.5:5060
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x4 (ulaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
SRVVOIP1*CLI>
<--- Reliably Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-UBFV-0e04cd9b-1b10a8d8;received=212.27.52.5
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=235 28-XD-102f2398-54dd3b156
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as737167 53
Call-ID: 23528-QA-102f2397-369c648f4@freephonie.net
CSeq: 264484018 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '23528-QA-102f2397-369c648f4@freephonie.net' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
ACK sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 23528-QA-102f2397-369c648f4@freephonie.net
CSeq: 264484018 ACK
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=235 28-XD-102f2398-54dd3b156
Max-Forwards: 27
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as737167 53
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-UBFV-0e04cd9b-1b10a8d8
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '23528-QA-102f2397-369c648f4@freephonie.net' Method: ACK
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>

[Reaper]

Found peer '0950XXXXXX-in' for '06XXXXXXXX' from 212.27.52.5:5060
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x4 (ulaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
SRVVOIP1*CLI>
<--- Reliably Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 488 Not acceptable here

Pour le peer 0950XXXXXX-in il faut authoriser le codec alaw
[0950XXXXXX-in]
allow=alaw

[LeRenard]

Ok, c'est fait

SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
INVITE sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
Contact: <sip:172.X.X.X:5062>
Content-Type: application/sdp
CSeq: 264728047 INVITE
From: "06XXXXXXXX" <sip:06XXXXXX@freephonie.net;user=phone>;tag=302 86-NB-1032fe7d-0553aa8e1
Max-Forwards: 27
Record-Route: <sip:C=on;t=EEZIZ@212.27.52.5:5060;lr>
To: <sip:0950XXXXXX@172.X.X.X;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-EEZI-0e094372-21092597
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.42q (gw_sip)
Content-Length: 184

v=0
o=cp10 131660620697 131660620697 IN IP4 172.X.X.X
s=SIP Call
c=IN IP4 212.27.52.130
t=0 0
m=audio 37620 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 10 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 212.27.52.5 : 5060 (NAT)
Using INVITE request as basis request - 30286-IQ-1032fe7c-7986080b4@freephonie.net
No user '06XXXXXXXX' in SIP users list
Found peer '0950XXXXXX-out' for '06XXXXXXXX' from 212.27.52.5:5060
SRVVOIP1*CLI>
<--- Reliably Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-EEZI-0e094372-21092597;received=212.27.52.5
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=302 86-NB-1032fe7d-0553aa8e1
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as5652bb 63
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
CSeq: 264728047 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="219a361b"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '30286-IQ-1032fe7c-7986080b4@freephonie.net' in 6400 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
ACK sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
CSeq: 264728047 ACK
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=302 86-NB-1032fe7d-0553aa8e1
Max-Forwards: 27
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as5652bb 63
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-EEZI-0e094372-21092597
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
INVITE sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Authorization: Digest username="anonymous",realm="asterisk",nonce="219a3 61b",uri="sip:0950XXXXXX@82.X.X.X:5060",response=" 86c534089edcb328ba1165c683e70308",algorithm=MD5,op aque=""
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
Contact: <sip:172.X.X.X:5062>
Content-Type: application/sdp
CSeq: 264728049 INVITE
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=302 86-NB-1032fe7d-0553aa8e1
Max-Forwards: 26
Record-Route: <sip:C=on;t=RQXKY@212.27.52.5:5060;lr>
To: <sip:0950XXXXXX@172.X.X.X;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RQXK-0e094374-32cd0b3e
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.42q (gw_sip)
Content-Length: 184

v=0
o=cp10 131660620699 131660620699 IN IP4 172.X.X.X
s=SIP Call
c=IN IP4 212.27.52.129
t=0 0
m=audio 36718 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv

<------------->
--- (14 headers 10 lines) ---
Sending to 212.27.52.5 : 5060 (NAT)
Using INVITE request as basis request - 30286-IQ-1032fe7c-7986080b4@freephonie.net
No user '06XXXXXXXX' in SIP users list
Found peer '0950XXXXXX-out' for '06XXXXXXXX' from 212.27.52.5:5060

<--- Reliably Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RQXK-0e094374-32cd0b3e;received=212.27.52.5
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=302 86-NB-1032fe7d-0553aa8e1
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as5652bb 63
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
CSeq: 264728049 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '30286-IQ-1032fe7c-7986080b4@freephonie.net' in 6400 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
ACK sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
CSeq: 264728049 ACK
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=302 86-NB-1032fe7d-0553aa8e1
Max-Forwards: 26
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as5652bb 63
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RQXK-0e094374-32cd0b3e
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>

[Reaper]

Il faut ajouter unsecure=invite pour le matched peer, mais vous avez un souci la, au début l'appel est identifié avec 0950XXXXXX-in, après sous 0950XXXXXX-out vous avez deux peers ou vous avez changé le nom de 0950XXXXXX-in sur 0950XXXXXX-out ? je vous conseille d'avoir seulement un peer 0950XXXXXX qui a le même nom que le numéro de téléphone.

[LeRenard]

J'ai rectifier et assembler le tout car je voulais tester séparément ce que ca donnerait... mais toujours le même problème.
Désolé de t'embêter Reaper

[Reaper]

Et bien encore une fois poste les traces.

[LeRenard]

Voici les traces:

SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
INVITE sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 03899-OL-104abd3b-1767bae27@freephonie.net
Contact: <sip:172.X.X.X:5062>
Content-Type: application/sdp
CSeq: 266246805 INVITE
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=038 99-BR-104abd3c-7d5952bc2
Max-Forwards: 27
Record-Route: <sip:C=on;t=CNFBF@212.27.52.5:5060;lr>
To: <sip:0950XXXXXX@172.X.X.X;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-CNFB-0e2c7845-2a7d002a
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.42q (gw_sip)
Content-Length: 184

v=0
o=cp10 131667697362 131667697362 IN IP4 172.X.X.X
s=SIP Call
c=IN IP4 212.27.52.129
t=0 0
m=audio 36848 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 10 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 212.27.52.5 : 5060 (NAT)
Using INVITE request as basis request - 03899-OL-104abd3b-1767bae27@freephonie.net
No user '06XXXXXXXX' in SIP users list
Found peer '0950XXXXXX' for '06XXXXXXXX' from 212.27.52.5:5060
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.27.52.129:36848
Looking for 0950XXXXXX in appel_interne_X (domain 82.X.X.X)
SRVVOIP1*CLI>
<--- Reliably Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-CNFB-0e2c7845-2a7d002a;received=212.27.52.5
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=038 99-BR-104abd3c-7d5952bc2
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as2921bb 00
Call-ID: 03899-OL-104abd3b-1767bae27@freephonie.net
CSeq: 266246805 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '03899-OL-104abd3b-1767bae27@freephonie.net' in 6400 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
ACK sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 03899-OL-104abd3b-1767bae27@freephonie.net
CSeq: 266246805 ACK
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=038 99-BR-104abd3c-7d5952bc2
Max-Forwards: 27
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as2921bb 00
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-CNFB-0e2c7845-2a7d002a
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '03899-OL-104abd3b-1767bae27@freephonie.net' Method: ACK
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>

Je te joins également les fichiers sip.con et extensions.conf pour que tu puisses voir les modifications et m'apporter peut-être une solution


sip.conf

[general]

Binaddr=0.0.0.0
Bindport=5060
Disallow=all
Allow=ulaw
Allow=alaw
Context=appel_interne_X
Dtmfmode=rfc2833
Allowoverlap=yes
Tos_sip=cs3
Tos_audio=ef
Tos=lowdelay
srvlookup=yes
language=fr
register => 0950XXXXXX:passwordsipfree@freephonie.net/0950XXXXXX
defaultexpirey=1800
fromdomain=freephonie.net
externip = 82.X.X.X
localnet=10.X.X.X
nat=yes


[0950513080]

type=friend
disallow=all
allow=alaw
host=freephonie.net
context=appel_interne_X
language=fr
insecure=invite
defaultuser=0950XXXXXX
secret=passwordsipfree
callerid="freebox" <0950XXXXXX>
nat=yes
canreinvite=no
dtmfmode=inband
videosupport=no
restrictcid=no
amaflags=default
defaultexpirey=1800
qualify=yes
fromuser=0950XXXXXX
fromdomain=freephonie.net


[202]

Defaultuser=202
Secret=3456
callerid="beber" <202>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
dtmf=inband
Mailbox=202

extensions.conf

[appel_interne_X]

Include=>salle_de_conference
Include=>local_voicemail
Include=>horloge_parlante
Include=>auto_attendant
Include=>parkedcalls
Include=>client
Include=>general


[general]

static=yes
writeprotect=no
autofallthrough=no
clearglobalvars=no
priorityjumping=no


[globals]

GENERAL=SIP/200&SIP/201&SIP/202&SIP/0950XXXXXX
CONSOLE=Console/dsp
SIPINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1


[client]

exten => _2XX,1,Answer()
exten => _2XX,2,Dial(SIP/${EXTEN},20,tr)
exten => _2XX,3,Voicemail(2${EXTEN}@default)
exten => _2XX,4,Hangup()


; MESSAGERIE
exten => _3XX,1,Answer()
exten => _3XX,2,Wait(1)
exten => _3XX,3,VoiceMailMain(${EXTEN}@default)
exten => _3XX,4,Hangup()


; APPELS ENTRANTS

exten => s,1,Answer
exten => s,2,Dial(SIP/${EXTEN}@0950XXXXXX),10,tm)
exten => s,3,VoiceMail(888)
exten => s,4,Hangup()


; APPELS SORTANTS

exten => _90[1-6]XXXXXXXX,1,Ringing()
exten => _90[1-6]XXXXXXXX,2,Wait()
exten => _90[1-6]XXXXXXXX,3,Answer()
exten => _90[1-6]XXXXXXXX,4,Dial(SIP/${EXTEN:1}@0950XXXXXX),30,r)


; LOOPBACK DE TEST

exten => 99,1,Answer()
exten => 99,2,SetLanguage(fr)
exten => 99,3,Echo()
exten => 99,4,Playback(vm-goodbye)
exten => 99,5,Hangup()


[local_voicemail]

Exten=>888,1,Answer()
Exten=>888,2,VoiceMailMain()
Exten=>888,3,Hangup()


[horloge_parlante]

Exten=>3344,1,Answer()
Exten=>3344,2,SayUnixTime(,Europe/Paris,AdBY kM)
Exten=>3344,3,Hangup()


[auto_attendant]

Exten=>2233,1,Read(digit,/var/lib/asterisk/sounds/en/hello-word,1)
Exten=>2233,2,Gotoif($["${digit}" = "1"]?appel_interne_X,3344,1)
Exten=>2233,3,Gotoif($["${digit}" = "2"]?appel_interne_X,900,1)
Exten=>2233,4,Gotoif($["${digit}" = "3"]?appel_interne_X,888,1)
Exten=>i,1,Goto(2233,1)


[macro-appel_interne_X] :

Exten=>s,1,Answer()
Exten=>s,2,Dial(${ARG1},20,Ttr)
Exten=>s,3,VoiceMail(${ARG1}@voicemail)
Exten=>s,4,Playback(vm-goodbye)
Exten=>s,5,Hangup()