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Pages : 1

#1 Le 24/08/2015, à 15:59

kboo

[Résolu] asterisk: connection ok,mais impossible de passer un appel!

bonjour,
j'ai suivit le tutoriel https://doc.ubuntu-fr.org/asterisk sur mon vps ovh
aucun problème de compilation ni d'installation, j'ai créer 3 utilisateurs, un sur un téléphone, un sur mon ordi et un autre sur une tablette. Tous se connectent sans problème en UDP. mais impossible de faire le moindre appel!!

J'ai mis ça à la fin du fichier /etc/asterisk/extensions.conf

[work]
exten => _6XXX,1,Dial(SIP/${EXTERN},20)
exten => _6XXX,2,Hangup()

/etc/asterisk/users.conf

[general]
hasvoicemail = yes
hassip = yes
hasiax = yes
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
nat = yes

[template](!)                   ; Nom du template (ici template)
type=friend                     ; Type d'objet SIP (friend = utilisateur)
host=dynamic                    ; Vous pouvez vous connecter a ce compte SIP a partir de n’importe quelle adresse IP
dtmfmode=rfc2833                ; Mode du DTMF
disallow=all                    ; Désactiver tous les codecs
allow=ulaw                      ; Activer les codecs µlaw
context = work                  ; Contexte (exploité par le fichier extensions.conf)

[6001](template)                ; Numéro SIP et template utilisé
fullname = TheKboo              ; Nom complet de l'utilisateur (ce qui s'affichera sur le téléphone)
username = kboo                 ; Nom d'utilisateur
secret=secret                 ; Mot de passe

[6002](template)                ; Numéro SIP et template utilisé
fullname = isa              ; Nom complet de l'utilisateur (ce qui s'affichera sur le téléphone)
username = isa                 ; Nom d'utilisateur
secret=secret                 ; Mot de passe

[6003](template)                ; Numéro SIP et template utilisé
fullname = TheMac              ; Nom complet de l'utilisateur (ce qui s'affichera sur le téléphone)
username = mac                 ; Nom d'utilisateur
secret=secret                 ; Mot de passe

voici les log:

vpsXXXXX*CLI>

<--- SIP read from UDP:AA.ZZ.EE.RR:49012 --->
REGISTER sip:vpsXXXXX.ovh.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP AA.ZZ.EE.RR:49012;branch=z9hG4bK-524287-1---151e335ce82c2b23;rport
Max-Forwards: 70
Contact: <sip:6001@AA.ZZ.EE.RR:49012;rinstance=d49ff2d1f0a2d0f7;transport=UDP>
To: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>
From: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=5a171337
Call-ID: HrSj7gR7d1h1t5HvSrjl-g..
CSeq: 27 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper r32395
Authorization: Digest username="6001",realm="asterisk",nonce="743bb136",uri="sip:vpsXXXXX.ovh.net;transport=UDP",response="1788830f69540a7300d3d4b61650f87e",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to AA.ZZ.EE.RR:49012 (no NAT)
Sending to AA.ZZ.EE.RR:49012 (no NAT)

<--- Transmitting (NAT) to AA.ZZ.EE.RR:49012 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP AA.ZZ.EE.RR:49012;branch=z9hG4bK-524287-1---151e335ce82c2b23;received=AA.ZZ.EE.RR;rport=49012
From: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=5a171337
To: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=as6f1cc40c
Call-ID: HrSj7gR7d1h1t5HvSrjl-g..
CSeq: 27 REGISTER
Server: Asterisk PBX 11.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7d375fa5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'HrSj7gR7d1h1t5HvSrjl-g..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:AA.ZZ.EE.RR:49012 --->
REGISTER sip:vpsXXXXX.ovh.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP AA.ZZ.EE.RR:49012;branch=z9hG4bK-524287-1---b18af841959e44ea;rport
Max-Forwards: 70
Contact: <sip:6001@AA.ZZ.EE.RR:49012;rinstance=d49ff2d1f0a2d0f7;transport=UDP>
To: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>
From: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=5a171337
Call-ID: HrSj7gR7d1h1t5HvSrjl-g..
CSeq: 28 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper r32395
Authorization: Digest username="6001",realm="asterisk",nonce="7d375fa5",uri="sip:vpsXXXXX.ovh.net;transport=UDP",response="1fc62aafdcd8f660f6323138b06714e2",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to AA.ZZ.EE.RR:49012 (no NAT)

<--- Transmitting (NAT) to AA.ZZ.EE.RR:49012 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP AA.ZZ.EE.RR:49012;branch=z9hG4bK-524287-1---b18af841959e44ea;received=AA.ZZ.EE.RR;rport=49012
From: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=5a171337
To: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=as6f1cc40c
Call-ID: HrSj7gR7d1h1t5HvSrjl-g..
CSeq: 28 REGISTER
Server: Asterisk PBX 11.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:6001@AA.ZZ.EE.RR:49012;rinstance=d49ff2d1f0a2d0f7;transport=UDP>;expires=60
Date: Mon, 24 Aug 2015 13:49:47 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '255dfa034e63272a350f03cf76068a88@QQ.SS.DD.FF:5060' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to AA.ZZ.EE.RR:49012:
NOTIFY sip:6001@AA.ZZ.EE.RR:49012;rinstance=d49ff2d1f0a2d0f7;transport=UDP SIP/2.0
Via: SIP/2.0/UDP QQ.SS.DD.FF:5060;branch=z9hG4bK43f33ee1;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@QQ.SS.DD.FF>;tag=as2bb54a57
To: <sip:6001@AA.ZZ.EE.RR:49012;rinstance=d49ff2d1f0a2d0f7;transport=UDP>
Contact: <sip:asterisk@QQ.SS.DD.FF:5060>
Call-ID: 255dfa034e63272a350f03cf76068a88@QQ.SS.DD.FF:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 11.19.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 93

Messages-Waiting: no
Message-Account: sip:asterisk@QQ.SS.DD.FF
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog 'HrSj7gR7d1h1t5HvSrjl-g..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:AA.ZZ.EE.RR:49012 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP QQ.SS.DD.FF:5060;branch=z9hG4bK43f33ee1;rport=5060
Contact: <sip:192.168.0.42:49012>
To: <sip:6001@AA.ZZ.EE.RR:49012;rinstance=d49ff2d1f0a2d0f7;transport=UDP>;tag=5f2f9150
From: "asterisk" <sip:asterisk@QQ.SS.DD.FF>;tag=as2bb54a57
Call-ID: 255dfa034e63272a350f03cf76068a88@QQ.SS.DD.FF:5060
CSeq: 102 NOTIFY
User-Agent: Zoiper r32395
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '255dfa034e63272a350f03cf76068a88@QQ.SS.DD.FF:5060' Method: NOTIFY

<--- SIP read from UDP:AA.ZZ.EE.RR:49012 --->
INVITE sip:6003@vpsXXXXX.ovh.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP AA.ZZ.EE.RR:49012;branch=z9hG4bK-524287-1---0b618baa532ee555;rport
Max-Forwards: 70
Contact: <sip:6001@AA.ZZ.EE.RR:49012;transport=UDP>
To: <sip:6003@vpsXXXXX.ovh.net;transport=UDP>
From: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=935c5e20
Call-ID: btPkszaG9DyDQ4Tj0odAVA..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper r32395
Allow-Events: presence, kpml
Content-Length: 160

v=0
o=Z 0 0 IN IP4 AA.ZZ.EE.RR
s=Z
c=IN IP4 AA.ZZ.EE.RR
t=0 0
m=audio 49934 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 9 lines) ---
Sending to AA.ZZ.EE.RR:49012 (no NAT)
Sending to AA.ZZ.EE.RR:49012 (no NAT)
Using INVITE request as basis request - btPkszaG9DyDQ4Tj0odAVA..
Found peer '6001' for '6001' from AA.ZZ.EE.RR:49012

<--- Reliably Transmitting (NAT) to AA.ZZ.EE.RR:49012 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP AA.ZZ.EE.RR:49012;branch=z9hG4bK-524287-1---0b618baa532ee555;received=AA.ZZ.EE.RR;rport=49012
From: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=935c5e20
To: <sip:6003@vpsXXXXX.ovh.net;transport=UDP>;tag=as123cc8df
Call-ID: btPkszaG9DyDQ4Tj0odAVA..
CSeq: 1 INVITE
Server: Asterisk PBX 11.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="150e2bc5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'btPkszaG9DyDQ4Tj0odAVA..' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:AA.ZZ.EE.RR:49012 --->
ACK sip:6003@vpsXXXXX.ovh.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP AA.ZZ.EE.RR:49012;branch=z9hG4bK-524287-1---0b618baa532ee555;rport
Max-Forwards: 70
To: <sip:6003@vpsXXXXX.ovh.net;transport=UDP>;tag=as123cc8df
From: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=935c5e20
Call-ID: btPkszaG9DyDQ4Tj0odAVA..
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:AA.ZZ.EE.RR:49012 --->
INVITE sip:6003@vpsXXXXX.ovh.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP AA.ZZ.EE.RR:49012;branch=z9hG4bK-524287-1---db2ef10d5e377367;rport
Max-Forwards: 70
Contact: <sip:6001@AA.ZZ.EE.RR:49012;transport=UDP>
To: <sip:6003@vpsXXXXX.ovh.net;transport=UDP>
From: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=935c5e20
Call-ID: btPkszaG9DyDQ4Tj0odAVA..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper r32395
Authorization: Digest username="6001",realm="asterisk",nonce="150e2bc5",uri="sip:6003@vpsXXXXX.ovh.net;transport=UDP",response="011837b0f1af7f0ff20cf2ddcbacc79b",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 160

v=0
o=Z 0 0 IN IP4 AA.ZZ.EE.RR
s=Z
c=IN IP4 AA.ZZ.EE.RR
t=0 0
m=audio 49934 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (15 headers 9 lines) ---
Sending to AA.ZZ.EE.RR:49012 (NAT)
Using INVITE request as basis request - btPkszaG9DyDQ4Tj0odAVA..
Found peer '6001' for '6001' from AA.ZZ.EE.RR:49012
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port AA.ZZ.EE.RR:49934
Looking for 6003 in work (domain vpsXXXXX.ovh.net)
list_route: hop: <sip:6001@AA.ZZ.EE.RR:49012;transport=UDP>

<--- Transmitting (NAT) to AA.ZZ.EE.RR:49012 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP AA.ZZ.EE.RR:49012;branch=z9hG4bK-524287-1---db2ef10d5e377367;received=AA.ZZ.EE.RR;rport=49012
From: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=935c5e20
To: <sip:6003@vpsXXXXX.ovh.net;transport=UDP>
Call-ID: btPkszaG9DyDQ4Tj0odAVA..
CSeq: 2 INVITE
Server: Asterisk PBX 11.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:6003@QQ.SS.DD.FF:5060>
Content-Length: 0


<------------>
    -- Executing [6003@work:1] Dial("SIP/6001-00000010", "SIP/,20") in new stack
[Aug 24 15:49:48] WARNING[30798][C-00000010]: app_dial.c:2330 dial_exec_full: Dial argument takes format (technology/resource)
  == Spawn extension (work, 6003, 1) exited non-zero on 'SIP/6001-00000010'
Scheduling destruction of SIP dialog 'btPkszaG9DyDQ4Tj0odAVA..' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to AA.ZZ.EE.RR:49012 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP AA.ZZ.EE.RR:49012;branch=z9hG4bK-524287-1---db2ef10d5e377367;received=AA.ZZ.EE.RR;rport=49012
From: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=935c5e20
To: <sip:6003@vpsXXXXX.ovh.net;transport=UDP>;tag=as08c9dfc5
Call-ID: btPkszaG9DyDQ4Tj0odAVA..
CSeq: 2 INVITE
Server: Asterisk PBX 11.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0


<------------>

<--- SIP read from UDP:AA.ZZ.EE.RR:49012 --->
ACK sip:6003@vpsXXXXX.ovh.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP AA.ZZ.EE.RR:49012;branch=z9hG4bK-524287-1---db2ef10d5e377367;rport
Max-Forwards: 70
To: <sip:6003@vpsXXXXX.ovh.net;transport=UDP>;tag=as08c9dfc5
From: "6001"<sip:6001@vpsXXXXX.ovh.net;transport=UDP>;tag=935c5e20
Call-ID: btPkszaG9DyDQ4Tj0odAVA..
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

Dernière modification par kboo (Le 24/08/2015, à 23:02)

Hors ligne

#2 Le 24/08/2015, à 16:55

sinbad83

Re : [Résolu] asterisk: connection ok,mais impossible de passer un appel!

Bonjour,
cela fait longtemps que je n'utilise plus Asterisk, mais je te conseille de plutôt passer à FreePBX qui est un outil de configuration graphique d'Asterisk. Comme c'est une distribution, pour l'utiliser sur Ubuntu, il suffit de la virtualiser.

Dernière modification par sinbad83 (Le 24/08/2015, à 16:59)

La connaissance n'est pas une denrée rare, il faut la partager avec les autres.
Linux registered #484707
Site: www.coursinforev.org/doku.php
Desktop AMD Ryzen 5-3600, RAM 16GB, Ubuntu 20.10,   HP Pavillon G6 Ubuntu 20.10 et Ten, Serveur Ubuntu 18.04

Hors ligne

#3 Le 24/08/2015, à 17:28

tiramiseb

Re : [Résolu] asterisk: connection ok,mais impossible de passer un appel!

Salut,

sinbad83: avant de conseiller un autre outil, demande si cela correspondrait au besoin ! Je n'aime pas cette manie qu'on a parfois à conseiller autre chose, juste parce qu'on ne connaît pas (ou plus) le produit utilisé. Surtout que là le problème est trivial.

---------

kboo a écrit :
exten => _6XXX,1,Dial(SIP/${EXTERN},20)

Tu as mis un "R" en trop, ici. Tu fais appel à la variable ${EXTEN}, qui contient l'extension appelée...

Dernière modification par tiramiseb (Le 24/08/2015, à 17:29)

Sébastien Maccagnoni - https://www.maccagnoni.eu - https://www.domotego.com

Hors ligne

#4 Le 24/08/2015, à 23:01

kboo

Re : [Résolu] asterisk: connection ok,mais impossible de passer un appel!

merci tiramiseb, c'était bien ça!!!

Hors ligne

Pages : 1