[RESOLU] Problème dial plan asterisk ligne free

[LeRenard]

Tentative d'appels:

201 vers 0950XXXXXX = Fail to etablished call avec X-Lite

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] ChanIsAvail("SIP/201-0000000e", "SIP/90950XXXXXX@free-0950XXXXXX") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:2] Dial("SIP/201-0000000e", "SIP/90950XXXXXX@free-0950XXXXXX),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)
-- Executing [90950XXXXXX@appel_interne_X:3] Congestion("SIP/201-0000000e", "") in new stack
== Spawn extension (appel_interne_X, 90950XXXXXX, 3) exited non-zero on 'SIP/201-0000000e'
SRVVOIP1*CLI>

0950XXXXXX vers 201 = bearercapability noauth avec ZoIPer

Rien dans la CLI

sip debug sur sip peer de free:

Malheureusement, je ne peux faire de sip debug sur le sip peer de free car en faisant un "sip show peers", je me suis rendu compte tout en sachant qu'il y a seulement les extensions 201 et 0950XXXXXX d'activer sur softphone, que le peer de free n'étant pas présent. J'ai donc obtenu ce résultat qui me parait très louche étant donné qu'il n'y a pas l'extension du numéro free...

SRVVOIP1*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
205 (Unspecified) D 5060 UNKNOWN
204 (Unspecified) D 5060 UNKNOWN
203 (Unspecified) D 5060 UNKNOWN
202 (Unspecified) D 5060 UNKNOWN
201/201 10.X.X.X D 8076 OK (19 ms)
200/200 (Unspecified) D 0 UNKNOWN
6 sip peers [Monitored: 1 online, 5 offline Unmonitored: 0 online, 0 offline]
SRVVOIP1*CLI>

Je pense que ca viens de là, qu'en pensez-vous ?
Si oui, comment faire pour résoudre cette erreur ? Merci d'avance.

[Reaper]

vos pouvez utiliser "sip set debug ip freephonie.net"

[LeRenard]

vos pouvez utiliser "sip set debug ip freephonie.net"

Oui, ca me met ce résultat qui tourne en boucle sans arrêt:

<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>

[Reaper]

Avez vous essayé d’appeler ?

[LeRenard]

Avez vous essayé d’appeler ?

Oui mais non...

0950XXXXXX vers 202 = L'indicatif que vous avez demandé n'est pas utiliser...
RIEN dans la CLI

202 vers 0950XXXXXX = Le serveur est hors d'atteinte
(voir résultat 2 messages plus loin pour CLI...)

[Reaper]

Je ne vois pas l'invite qui pars vers le free.

[LeRenard]

Bonjour, désolé du petit retard
Voilà le résultat:

<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
INVITE sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-7d5eff759e47110d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:202@10.X.X.X:61399;rinstance=340261802ca46cef >
To: <sip:90950XXXXXX@WORKGROUP:5060>
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.19920.0
Content-Length: 407

v=0
o=3cxVCE 158336970 98167440 IN IP4 10.X.X.X
s=3cxVCE Audio Call
c=IN IP4 10.X.X.X
t=0 0
m=audio 40036 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40034 RTP/AVP 34
c=IN IP4 10.X.X.X
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;
a=sendrecv

<------------->
--- (13 headers 18 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 10.X.X.X : 61399 (no NAT)
Using INVITE request as basis request - Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
Found user '202' for '202'

<--- Reliably Transmitting (no NAT) to 10.X.X.X:61399 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-7d5eff759e47110d-1---d8754z-;received=10.X.X.X;rport=61399
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as2c365984
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 1 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="098e1d89"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
ACK sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-7d5eff759e47110d-1---d8754z-;rport
Max-Forwards: 70
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as2c365984
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
INVITE sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:202@10.X.X.X:61399;rinstance=340261802ca46cef >
To: <sip:90950XXXXXX@WORKGROUP:5060>
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.19920.0
Authorization: Digest username="202",realm="asterisk",nonce="098e1d89",u ri="sip:90950XXXXXX@WORKGROUP:5060",response="217b 28fe239c388230418edca15ba4f9",algorithm=MD5
Content-Length: 407

v=0
o=3cxVCE 158336970 98167440 IN IP4 10.X.X.X
s=3cxVCE Audio Call
c=IN IP4 10.X.X.X
t=0 0
m=audio 40036 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40034 RTP/AVP 34
c=IN IP4 10.X.X.X
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;
a=sendrecv

<------------->
--- (14 headers 18 lines) ---
Sending to 10.X.X.X : 61399 (no NAT)
Using INVITE request as basis request - Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
Found user '202' for '202'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found video description format H263 for ID 34
Capabilities: us - 0x4 (ulaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.X.X.X:40036
Looking for 90950XXXXXX in appel_interne_X (domain WORKGROUP)
list_route: hop: <sip:202@10.X.X.X:61399;rinstance=340261802ca46cef >
SRVVOIP1*CLI>
<--- Transmitting (no NAT) to 10.X.X.X:61399 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;received=10.X.X.X;rport=61399
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
To: <sip:90950XXXXXX@WORKGROUP:5060>
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:90950XXXXXX@10.X.X.X>
Content-Length: 0


<------------>
-- Executing [90950XXXXXX@appel_interne_X:1] ChanIsAvail("SIP/202-0000000b", "SIP/90950XXXXXX@free-0950XXXXXX") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Really destroying SIP dialog '3d93105b788f6f256b4ddcdb1e4defce@10.X.X.X' Method: INVITE
-- Executing [90950XXXXXX@appel_interne_X:2] Dial("SIP/202-0000000b", "SIP/90950XXXXXX@free-0950XXXXXX),30,r") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Really destroying SIP dialog '2870e9af28c7940d24f94dc775dcf1dd@10.X.X.X' Method: INVITE
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [90950XXXXXX@appel_interne_X:3] Congestion("SIP/202-0000000b", "") in new stack

<--- Reliably Transmitting (no NAT) to 10.X.X.X:61399 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;received=10.X.X.X;rport=61399
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as7c874b55
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 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
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 20


<------------>
== Spawn extension (appel_interne_X, 90950XXXXXX, 3) exited non-zero on 'SIP/202-0000000b'
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
ACK sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;rport
Max-Forwards: 70
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as7c874b55
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.' Method: ACK
SRVVOIP1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[SRVVOIP1.localdomain asterisk]#