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Erreur 480 côté OVH lors des appels sortants

A
Von
AlekF
Erstellungsdatum 2024-05-15 07:55:35 (edited on 2024-11-18 11:06:03) in Forfaits VoIP

A tous, bonjour,

**Configuration concernée :**
Offre : OVH SIP Découverte.
Infra : serveur en LAN derrière un pare-feu ouvert en publique
Config serveur : serveur VOIP tournant sous Asterisk 21, avec module PJSIP 2.14.1.
Config pare-feu : redirection port TCP/UDP 5060, redirection de la plage UPD 10000-50000
Téléphone VOIP : software Zoiper

**Problématique :**

L'enregistrement SIP fonctionne.
La communication entre les zoiper en local fonctionne.
Les appels sortants ne fonctionnent pas. Le serveur OVH me retourne un code 480.

**Contenu du pjsip.conf :**

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=10.0.1.0/24 ///// --> celui-ci est utilisé pour les téléphones ////
local_net=10.0.0.0/24
external_media_address=IP_PUBLIQUE
external_signaling_address=IP_PUBLIQUE

[ovh]
type=endpoint
transport=transport-udp
context=ovh-sip-entrant
allow=!all,gsm
outbound_auth=ovh
from_domain=sbc6.fr.sip.ovh
from_user=01********
callerid=01********
dtmf_mode=rfc4733
contact_user=01********
direct_media=no
force_rport=yes
rewrite_contact=yes
rtp_symmetric=yes
send_pai=yes
send_rpid=yes
aors=ovh
connected_line_method=update
direct_media_method=update

[ovh]
type=auth
auth_type=userpass
password=lemodepassecorrect
username=00331********
realm=sbc6.fr.sip.ovh


[ovh]
type=registration
outbound_auth=ovh
server_uri=sip:sbc6.fr.sip.ovh
client_uri=sip:01********@sbc6.fr.sip.ovh
contact_user=01********
retry_interval=60
expiration=120
line=yes
endpoint=ovh

[ovh]
type=aor
contact=sip:sbc6.fr.sip.ovh:5060
max_contacts=2

[ovh]
type=identify
endpoint=ovh
match=sbc6.fr.sip.ovh





[101]
type=endpoint
context=local
disallow=all
callerid=Mon Nom perso <01********>
allow=gsm
auth=101
aors=101
direct_media=no
force_rport=yes
rewrite_contact=yes
rtp_symmetric=yes
send_pai=yes
send_rpid=yes
geoloc_outgoing_call_profile=blabla

[101]
type=auth
auth_type=userpass
password=******
username=101

[101]
type=aor
max_contacts=1

[102]
type=endpoint
context=local
disallow=all
allow=gsm
auth=102
aors=102

[102]
type=auth
auth_type=userpass
password=******
username=102

[102]
type=aor
max_contacts=1

**Contenus des logs pjsip :**

LOG DE PJSIP DE LA SEQUENCE D ENREGISTREMENT (RETOUR CODE 200, DONC RAS) :

PJSIP Logging enabled
<--- Transmitting SIP request (591 bytes) to UDP:91.121.129.134:5060 --->
REGISTER sip:sbc6.fr.sip.ovh SIP/2.0
Via: SIP/2.0/UDP IP_PUBLIQUE:5060;rport;branch=z9hG4bKPj92c14be1-71f6-4db2-9db5-05822b352b11
From: ;tag=f6e2d86b-7800-42a4-93be-aa11a3cdfb14
To:
Call-ID: 583e609f-6b77-43ba-8cc7-1e8aa221f64b
CSeq: 8499 REGISTER
Contact:
Expires: 120
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 21.2.0
Content-Length: 0


<--- Received SIP response (512 bytes) from UDP:91.121.129.134:5060 --->
SIP/2.0 200 OK
Call-ID: 583e609f-6b77-43ba-8cc7-1e8aa221f64b
Contact: ;expires=120
CSeq: 8499 REGISTER
From: ;tag=f6e2d86b-7800-42a4-93be-aa11a3cdfb14
To: ;tag=00-29273-724e5dfb-427845453
Via: SIP/2.0/UDP IP_PUBLIQUE:5060;received=IP_PUBLIQUE;rport=5060;branch=z9hG4bKPj92c14be1-71f6-4db2-9db5-05822b352b11
P-Associated-URI:
Content-Length: 0


\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\
\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\
\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\
\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\


LOG LORS D UN APPEL SORTANT (retour code 480 côté OVH, donc ÉCHEC) :


<--- Received SIP request (1014 bytes) from UDP:IP_TEL_LOCAL:62432 --->
INVITE sip:0140158461@IP_SERVEUR_LOCAL:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP IP_TEL_LOCAL:62432;branch=z9hG4bK-524287-1---71e74167ae1d6ebf;rport
Max-Forwards: 70
Contact:
To:
From: ;tag=1eae5e1f
Call-ID: Yk15TrHCEQ32FIyrnhL02w..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.4 v2.10.20.4_1
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 333

v=0
o=Z 0 179414133 IN IP4 IP_TEL_LOCAL
s=Z
c=IN IP4 IP_TEL_LOCAL
t=0 0
m=audio 63802 RTP/AVP 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP response (490 bytes) to UDP:IP_TEL_LOCAL:62432 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP IP_TEL_LOCAL:62432;rport=62432;received=IP_TEL_LOCAL;branch=z9hG4bK-524287-1---71e74167ae1d6ebf
Call-ID: Yk15TrHCEQ32FIyrnhL02w..
From: ;tag=1eae5e1f
To: ;tag=z9hG4bK-524287-1---71e74167ae1d6ebf
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1715735900/9681e974dbe42903d0eb1a927ab6d8dc",opaque="57eae2be7254f383",algorithm=MD5,qop="auth"
Server: Asterisk PBX 21.2.0
Content-Length: 0


<--- Received SIP request (354 bytes) from UDP:IP_TEL_LOCAL:62432 --->
ACK sip:0140158461@IP_SERVEUR_LOCAL:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP IP_TEL_LOCAL:62432;branch=z9hG4bK-524287-1---71e74167ae1d6ebf;rport
Max-Forwards: 70
To: ;tag=z9hG4bK-524287-1---71e74167ae1d6ebf
From: ;tag=1eae5e1f
Call-ID: Yk15TrHCEQ32FIyrnhL02w..
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1317 bytes) from UDP:IP_TEL_LOCAL:62432 --->
INVITE sip:0140158461@IP_SERVEUR_LOCAL:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP IP_TEL_LOCAL:62432;branch=z9hG4bK-524287-1---6f1a551d34ace053;rport
Max-Forwards: 70
Contact:
To:
From: ;tag=1eae5e1f
Call-ID: Yk15TrHCEQ32FIyrnhL02w..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.4 v2.10.20.4_1
Authorization: Digest username="101",realm="asterisk",nonce="1715735900/9681e974dbe42903d0eb1a927ab6d8dc",uri="sip:0140158461@IP_SERVEUR_LOCAL:5060;transport=UDP",response="0576023278c7d52ddc535b57f9f623ec",cnonce="66ef14ae7c61737358d3ebca4ac0c5d1",nc=00000001,qop=auth,algorithm=MD5,opaque="57eae2be7254f383"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 333

v=0
o=Z 0 179414133 IN IP4 IP_TEL_LOCAL
s=Z
c=IN IP4 IP_TEL_LOCAL
t=0 0
m=audio 63802 RTP/AVP 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP response (298 bytes) to UDP:IP_TEL_LOCAL:62432 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP_TEL_LOCAL:62432;rport=62432;received=IP_TEL_LOCAL;branch=z9hG4bK-524287-1---6f1a551d34ace053
Call-ID: Yk15TrHCEQ32FIyrnhL02w..
From: ;tag=1eae5e1f
To:
CSeq: 2 INVITE
Server: Asterisk PBX 21.2.0
Content-Length: 0


-- Executing [0140158461@local:1] Dial("PJSIP/101-00000000", "PJSIP/0140158461@ovh") in new stack
-- Called PJSIP/0140158461@ovh
<--- Transmitting SIP request (1112 bytes) to UDP:91.121.129.134:5060 --->
INVITE sip:0140158461@sbc6.fr.sip.ovh:5060 SIP/2.0
Via: SIP/2.0/UDP IP_PUBLIQUE:5060;rport;branch=z9hG4bKPjd259e72d-f5f9-473f-be87-3325a938fb57
From: ;tag=b68d1b7a-d7e5-4a38-a79c-91ee9b0f60b3
To:
Contact:
Call-ID: e6b1abd3-0a31-472b-8b6c-104964c287e3
CSeq: 8930 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "Mon Nom perso"
Remote-Party-ID: "Mon Nom perso" ;party=calling;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 21.2.0
Content-Type: application/sdp
Content-Length: 236

v=0
o=- 491366604 491366604 IN IP4 192.168.1.205
s=Asterisk
c=IN IP4 192.168.1.205
t=0 0
m=audio 17886 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:300
a=sendrecv

<--- Received SIP response (354 bytes) from UDP:91.121.129.134:5060 --->
SIP/2.0 100 Trying
Call-ID: e6b1abd3-0a31-472b-8b6c-104964c287e3
CSeq: 8930 INVITE
From: ;tag=b68d1b7a-d7e5-4a38-a79c-91ee9b0f60b3
To:
Via: SIP/2.0/UDP IP_PUBLIQUE:5060;received=IP_PUBLIQUE;rport=5060;branch=z9hG4bKPjd259e72d-f5f9-473f-be87-3325a938fb57
Content-Length: 0


<--- Received SIP response (800 bytes) from UDP:91.121.129.134:5060 --->
SIP/2.0 407 authentication required
Call-ID: e6b1abd3-0a31-472b-8b6c-104964c287e3
Contact:
CSeq: 8930 INVITE
From: ;tag=b68d1b7a-d7e5-4a38-a79c-91ee9b0f60b3
Proxy-Authenticate: Digest realm="sbc6.fr.sip.ovh",nonce="03f3ad2742aac5c5373baff466f192c9",opaque="03e8271e3926bee",stale=false,algorithm=MD5
Record-Route: ;session=359170
To: ;tag=00-08187-03f3b444-331ef1e15
Via: SIP/2.0/UDP IP_PUBLIQUE:5060;received=IP_PUBLIQUE;rport=5060;branch=z9hG4bKPjd259e72d-f5f9-473f-be87-3325a938fb57
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.76 (gw_sip)
Content-Length: 0


<--- Transmitting SIP request (434 bytes) to UDP:91.121.129.134:5060 --->
ACK sip:0140158461@sbc6.fr.sip.ovh:5060 SIP/2.0
Via: SIP/2.0/UDP IP_PUBLIQUE:5060;rport;branch=z9hG4bKPjd259e72d-f5f9-473f-be87-3325a938fb57
From: ;tag=b68d1b7a-d7e5-4a38-a79c-91ee9b0f60b3
To: ;tag=00-08187-03f3b444-331ef1e15
Call-ID: e6b1abd3-0a31-472b-8b6c-104964c287e3
CSeq: 8930 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 21.2.0
Content-Length: 0


<--- Transmitting SIP request (1362 bytes) to UDP:91.121.129.134:5060 --->
INVITE sip:0140158461@sbc6.fr.sip.ovh:5060 SIP/2.0
Via: SIP/2.0/UDP IP_PUBLIQUE:5060;rport;branch=z9hG4bKPj83ab788a-033e-4f45-ac63-58cf13e2becc
From: ;tag=b68d1b7a-d7e5-4a38-a79c-91ee9b0f60b3
To:
Contact:
Call-ID: e6b1abd3-0a31-472b-8b6c-104964c287e3
CSeq: 8931 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 21.2.0
Proxy-Authorization: Digest username="00331********", realm="sbc6.fr.sip.ovh", nonce="03f3ad2742aac5c5373baff466f192c9", uri="sip:0140158461@sbc6.fr.sip.ovh:5060", response="bb1a0f1a31be27274217ac21b48c3197", algorithm=MD5, opaque="03e8271e3926bee"
P-Asserted-Identity: "Mon Nom perso"
Remote-Party-ID: "Mon Nom perso" ;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 236

v=0
o=- 491366604 491366604 IN IP4 192.168.1.205
s=Asterisk
c=IN IP4 192.168.1.205
t=0 0
m=audio 17886 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:300
a=sendrecv

<--- Received SIP response (354 bytes) from UDP:91.121.129.134:5060 --->
SIP/2.0 100 Trying
Call-ID: e6b1abd3-0a31-472b-8b6c-104964c287e3
CSeq: 8931 INVITE
From: ;tag=b68d1b7a-d7e5-4a38-a79c-91ee9b0f60b3
To:
Via: SIP/2.0/UDP IP_PUBLIQUE:5060;received=IP_PUBLIQUE;rport=5060;branch=z9hG4bKPj83ab788a-033e-4f45-ac63-58cf13e2becc
Content-Length: 0


<--- Received SIP response (660 bytes) from UDP:91.121.129.134:5060 --->
SIP/2.0 480 Temporarily Not Available
Call-ID: e6b1abd3-0a31-472b-8b6c-104964c287e3
Contact:
CSeq: 8931 INVITE
From: ;tag=b68d1b7a-d7e5-4a38-a79c-91ee9b0f60b3
Record-Route: ;session=359171
To: ;tag=00-08189-03f3b44b-31764ec44
Via: SIP/2.0/UDP IP_PUBLIQUE:5060;received=IP_PUBLIQUE;rport=5060;branch=z9hG4bKPj83ab788a-033e-4f45-ac63-58cf13e2becc
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Reason: q.850;cause=41
Server: Cirpack/v4.76 (gw_sip)
Content-Length: 0


<--- Transmitting SIP request (434 bytes) to UDP:91.121.129.134:5060 --->
ACK sip:0140158461@sbc6.fr.sip.ovh:5060 SIP/2.0
Via: SIP/2.0/UDP IP_PUBLIQUE:5060;rport;branch=z9hG4bKPj83ab788a-033e-4f45-ac63-58cf13e2becc
From: ;tag=b68d1b7a-d7e5-4a38-a79c-91ee9b0f60b3
To: ;tag=00-08189-03f3b44b-31764ec44
Call-ID: e6b1abd3-0a31-472b-8b6c-104964c287e3
CSeq: 8931 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 21.2.0
Content-Length: 0


== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'PJSIP/101-00000000' status is 'CHANUNAVAIL'
<--- Transmitting SIP response (503 bytes) to UDP:IP_TEL_LOCAL:62432 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP IP_TEL_LOCAL:62432;rport=62432;received=IP_TEL_LOCAL;branch=z9hG4bK-524287-1---6f1a551d34ace053
Call-ID: Yk15TrHCEQ32FIyrnhL02w..
From: ;tag=1eae5e1f
To: ;tag=427a8503-95ab-44e8-bff5-5082d2c0d9be
CSeq: 2 INVITE
Server: Asterisk PBX 21.2.0
Reason: Q.850;cause=34
P-Asserted-Identity:
Remote-Party-ID: ;party=called;privacy=off;screen=no
Content-Length: 0


<--- Received SIP request (406 bytes) from UDP:91.121.129.134:5060 --->
OPTIONS sip:01********@IP_PUBLIQUE:5060;line=qokvxfy SIP/2.0
Call-ID: 00-04447-728c8147-26e1e7cb0@91.121.129.134
Contact:
CSeq: 1 OPTIONS
From: ;tag=00-04447-728c8146-38d266834
Max-Forwards: 70
To:
Via: SIP/2.0/UDP 91.121.129.134:5060;rport;branch=z9hG4bK-DMGH-4bb4ff04-74bb3c92
Content-Length: 0


<--- Transmitting SIP response (889 bytes) to UDP:91.121.129.134:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 91.121.129.134:5060;rport=5060;received=91.121.129.134;branch=z9hG4bK-DMGH-4bb4ff04-74bb3c92
Call-ID: 00-04447-728c8147-26e1e7cb0@91.121.129.134
From: ;tag=00-04447-728c8146-38d266834
To: ;tag=z9hG4bK-DMGH-4bb4ff04-74bb3c92
CSeq: 1 OPTIONS
Accept: application/sdp, application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/pidf+xml, application/simple-message-summary, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 21.2.0
Content-Length: 0


<--- Received SIP request (355 bytes) from UDP:IP_TEL_LOCAL:62432 --->
ACK sip:0140158461@IP_SERVEUR_LOCAL:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP IP_TEL_LOCAL:62432;branch=z9hG4bK-524287-1---6f1a551d34ace053;rport
Max-Forwards: 70
To: ;tag=427a8503-95ab-44e8-bff5-5082d2c0d9be
From: ;tag=1eae5e1f
Call-ID: Yk15TrHCEQ32FIyrnhL02w..
CSeq: 2 ACK
Content-Length: 0
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2 Antworten ( Latest reply on 2024-11-18 11:06:10 Von
AlekF
)
R

Salut AlekF,

Aurais-tu trouvé une solution ?
Je passe mon ancien serveur asterisk 13 vers 20 et j'ai des problèmes avec la config SIP > PJSIP.
Tout fonctionne maintenant sauf les appels extérieur.
Hilfreich (0)
A

Salut Raphael,

Je te prie de m'excuser pour ma réponse tardive. A sein de la config et du contexte expliqués plus haut, j'ai finalement trouvé une solution qui consistait à :

1- correctement configurer les paramètres de transport UDP et de gestion des packets derrière un NAT (car il y a ré-écriture de l'IP par le routeur et tout un tas d'opérations annexes blablabla qui perturbe asterisk)
2- configurer l'encodage sur ULAW (selon tous les tests que j'ai réalisé sur mon infra, c'était le point le PLUS BLOQUANT pour une ligne SIP OVH)

Un bémol : Bien que le transport UDP de la ligne OVH fonctionne derrière le NAT, je ne peux pas utiliser les téléphones clients derrière le NAT (cad via IP publique du routeur) car la réécriture des paquets pose problème et n'étant pas très bien calé j'ai opté pour la "simplicité" : un VPN sur lequel les clients doivent se connecter préalablement à la connexion SIP, de façon, la connexion en local est transparente pour les téléphones. Autre bonne raison, ce n'est pas sécurisé de faire transiter du SIP en clair sur un réseau public haha. A ce titre, Wireguard fait très bien l'affaire, j'ai très très peu de pertes en terme de latence.

Voici mes fichiers de conf, fonctionnels à ce jour :

https://cloud.submarines.eu/s/JLMFbteFHYkSW3A

Bien à toi.
Alexandre I.
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