problème appel entrant configuration ovh

**kalidev** · 22/03/2011, 09h51

Envoyé par Reaper

La trace SIP ne montre rien. Il te manque la trace de CLI pendant l'appel entrant, colle la ici.
J'ai supprimé mon dernier message puisque tu as déjà le renvoie vers numéro correct.

Kalivev1 et 2 est connecté ?

Bonjour,

oui, les deux téléphone kalidev 1 et 2 sont connectés. Dans la CLI je met la verbose à 10 mais je n'ai rien qui s'affiche lors d'un appel entrant, comme si l'appel n'atteignait pas mon serveur.

Code:

prodserver*CLI> sip show registry
Host                           dnsmgr Username       Refresh State                Reg.Time
sip.ovh.net:5060               N      0033535541xx      105 Registered           Tue, 22 Mar 2011 09:01:34
sip.ovh.net:5060               N      0033535541yy       105 Registered           Tue, 22 Mar 2011 09:02:37

Code:

prodserver*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
forfait-ovh/0033535541xxx  91.121.129.17        N      5060     OK (61 ms)
forfait-ovh2/003353554yyy 91.121.129.17        N      5060     OK (60 ms)
kalidev1/kalidev1          192.168.1.20     D   N      5060     Unmonitored
kalidev2/kalidev2          192.168.1.20     D   N      5060     Unmonitored
4 sip peers [Monitored: 2 online, 0 offline Unmonitored: 2 online, 0 offline]

Pour un appel sortant par contre j'ai bien les messages :

Code:

Verbosity was 5 and is now 10
  == Using SIP RTP CoS mark 5
    -- Executing [06731705xx@appel-sortant:1] Dial("SIP/kalidev1-0000000a", "SIP/06731705xx@forfait-ovh") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 06731705xx@forfait-ovh
    -- SIP/forfait-ovh-0000000b is ringing
    -- SIP/forfait-ovh-0000000b is making progress passing it to SIP/kalidev1-0000000a
  == Spawn extension (appel-sortant, 06731705xx, 1) exited non-zero on 'SIP/kalidev1-0000000a'

Merci de ton aide

**Reaper** · 22/03/2011, 10h55

Alors active le sip debug sur le téléphone concerné, et sur tes fournisseurs, appelle et colle tout sur pastebin, ou dans un fichier et l'attache ici.

**kalidev** · 22/03/2011, 11h10

Envoyé par Reaper

Alors active le sip debug sur le téléphone concerné, et sur ton fournisseurs, appelle et colle tout sur pastebin, ou dans un fichier et l'attache ici.

Il ne se passe rien avec le sip set debug peer kalidev1 et même sip set debug on.

Aucune trace

**Reaper** · 22/03/2011, 11h21

et sur ton fournisseur

Colle tout ici je te dis.

**kalidev** · 22/03/2011, 11h33

voila ce que j'ai avec sip debug sur le fournisseur et le téléphone concerné :

Code:

<--- SIP read from UDP:192.168.1.20:5060 --->
REGISTER sip:192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK7661e6afeb5feb59aa739d1eb44f4d7;rport
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1694489133
To: "kalidev1" <sip:kalidev1@192.168.1.10>
Call-ID: 2652774740@192_168_1_20
CSeq: 10999 REGISTER
Contact: <sip:kalidev1@192.168.1.20:5060>
Max-Forwards: 70
User-Agent: A580 IP021920000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.1.20 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK7661e6afeb5feb59aa739d1eb44f4d7;received=192.168.1.20;rport=5060
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1694489133
To: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=as79e3afc6
Call-ID: 2652774740@192_168_1_20
CSeq: 10999 REGISTER
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="62adcae5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2652774740@192_168_1_20' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.20:5060 --->
REGISTER sip:192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK6cb7fd43a5a990958942e8acefbc9068;rport
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1694489133
To: "kalidev1" <sip:kalidev1@192.168.1.10>
Call-ID: 2652774740@192_168_1_20
CSeq: 11000 REGISTER
Contact: <sip:kalidev1@192.168.1.20:5060>
Authorization: Digest username="kalidev1", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.10", nonce="62adcae5", response="5454a87e3f25dee5e45d95aaacd0e2cc"
Max-Forwards: 70
User-Agent: A580 IP021920000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.1.20 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK6cb7fd43a5a990958942e8acefbc9068;received=192.168.1.20;rport=5060
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1694489133
To: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=as79e3afc6
Call-ID: 2652774740@192_168_1_20
CSeq: 11000 REGISTER
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 180
Contact: <sip:kalidev1@192.168.1.20:5060>;expires=180
Date: Tue, 22 Mar 2011 09:44:48 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2652774740@192_168_1_20' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.20:5060 --->
REGISTER sip:192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK3e8ec846d6facef21f0a2dd424d323d9;rport
From: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=260232850
To: "kalidev 2" <sip:kalidev2@192.168.1.10>
Call-ID: 1219423529@192_168_1_20
CSeq: 13700 REGISTER
Contact: <sip:kalidev2@192.168.1.20:5060>
Max-Forwards: 70
User-Agent: A580 IP021920000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.1.20 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK3e8ec846d6facef21f0a2dd424d323d9;received=192.168.1.20;rport=5060
From: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=260232850
To: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=as7b678c2d
Call-ID: 1219423529@192_168_1_20
CSeq: 13700 REGISTER
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="15f2f0d7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1219423529@192_168_1_20' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.20:5060 --->
REGISTER sip:192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK74c89957c85cc131509a64262982de0e;rport
From: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=260232850
To: "kalidev 2" <sip:kalidev2@192.168.1.10>
Call-ID: 1219423529@192_168_1_20
CSeq: 13701 REGISTER
Contact: <sip:kalidev2@192.168.1.20:5060>
Authorization: Digest username="kalidev2", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.10", nonce="15f2f0d7", response="6a14cf5a06890be3d48d57d27ac466d6"
Max-Forwards: 70
User-Agent: A580 IP021920000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.1.20 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK74c89957c85cc131509a64262982de0e;received=192.168.1.20;rport=5060
From: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=260232850
To: "kalidev 2" <sip:kalidev2@192.168.1.10>;tag=as7b678c2d
Call-ID: 1219423529@192_168_1_20
CSeq: 13701 REGISTER
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 180
Contact: <sip:kalidev2@192.168.1.20:5060>;expires=180
Date: Tue, 22 Mar 2011 09:44:48 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1219423529@192_168_1_20' in 32000 ms (Method: REGISTER)
prodserver*CLI>

**kalidev** · 22/03/2011, 11h35

voila ce que j'ai si je passe un appel :

Code:

<--- Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f
Call-ID: 3641468165@192_168_1_20
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:06731705xx@192.168.1.10>
Content-Length: 0


<------------>
Audio is at 192.168.1.10 port 13190
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f
Call-ID: 3641468165@192_168_1_20
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:06731705xx@192.168.1.10>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 152311073 152311073 IN IP4 192.168.1.10
s=Asterisk PBX 1.6.2.9-2
c=IN IP4 192.168.1.10
t=0 0
m=audio 13190 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:192.168.1.20:5060 --->
CANCEL sip:06731705xx@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;rport
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone
Call-ID: 3641468165@192_168_1_20
CSeq: 3 CANCEL
Contact: <sip:kalidev1@192.168.1.20:5060>
Authorization: Digest username="kalidev1", realm="asterisk", algorithm=MD5, uri="sip:06731705xx@192.168.1.10", nonce="17667a62", response="41d4678f083049f066476f80d4b380e8"
Max-Forwards: 70
User-Agent: A580 IP021920000000
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.20 : 5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f
Call-ID: 3641468165@192_168_1_20
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


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

<--- Transmitting (no NAT) to 192.168.1.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;received=192.168.1.20;rport=5060
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f
Call-ID: 3641468165@192_168_1_20
CSeq: 3 CANCEL
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.20:5060 --->
ACK sip:06731705xx@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK5ec10b8b137f72489268ba8050973fae;rport
From: "kalidev1" <sip:kalidev1@192.168.1.10>;tag=1446470539
To: <sip:06731705xx@192.168.1.10>;user=phone;tag=as390c2a3f
Call-ID: 3641468165@192_168_1_20
CSeq: 3 ACK
Contact: <sip:kalidev1@192.168.1.20:5060>
Authorization: Digest username="kalidev1", realm="asterisk", algorithm=MD5, uri="sip:06731705xx@192.168.1.10", nonce="17667a62", response="fdea256982ac7f340489f95dfba9f231"
Max-Forwards: 70
User-Agent: A580 IP021920000000
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '3641468165@192_168_1_20' Method: ACK
[Mar 22 10:47:21] NOTICE[8131]: chan_sip.c:11528 sip_reregister:    -- Re-registration for  0033535541287@sip.ovh.net
[Mar 22 10:47:21] NOTICE[8131]: chan_sip.c:18270 handle_response_register: Outbound Registration: Expiry for sip.ovh.net is 120 sec (Scheduling reregistration in 105 s)
prodserver*CLI>

**Reaper** · 22/03/2011, 11h41

Bonjour, la trace est incomplète, colle les traces sur pastebin (again)
Et chaque fois il faut spécifier si c'est un appel sortant ou entrant.

En tout cas on vois que c'est le 192.168.1.20 qui termine le communication.

Je vous demande les traces complètes de l'appel qui ne fonctionne pas pour la dernière fois.

Discussion: problème appel entrant configuration ovh

Outils de la discussion

Rechercher dans la discussion

Affichage

Vue hybride

Les tags pour cette discussion

Règles de messages