[RESOLU] Problème dial plan asterisk ligne free

**LeRenard** · 15/09/2011, 09h56

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]#

**LeRenard** · 15/09/2011, 15h48

On me dit dans l'oreillette (dixit, un modérateur d'un forum...) que la clé du problème est:

== Everyone is busy/congested at this time (1:0/0/1)

D'après ce qu'on m'a dit ci-dessus, j'en conclus que le problème viendrais du fichier extensions.conf et plus particulièrement sur cette partie:

; APPELS SORTANTS

exten => _9.,1,ChanIsAvail(SIP/${EXTEN}@free-0950XXXXXX)
exten => _9.,2,Dial(SIP/${EXTEN}@free-0950XXXXXX),30,r)
exten => _9.,3,Congestion

exten => _9.,104,Dial(Zap/1/${EXTEN},30,r)
exten => _9.,105,Congestion

exten => _9.,206,SetLanguage(fr)
exten => _9.,207,Wait(1)
exten => _9.,208,Playback(all-circuits-busy-now)
exten => _9.,209,Hangup

Qu'en pensez-vous ?
Que dois-je faire pour y remédier si c'est vraiment le problème ?

Merci d'avance !

**Reaper** · 15/09/2011, 18h08

exten => _9.,2,Dial(SIP/${EXTEN:1}@free-0950XXXXXX),30,r)

Il faut retirer le 9 devant. Et je ne vois pas l'invite qui va vers FREE, ilf faut Activer le debug dessus.

**LeRenard** · 16/09/2011, 15h19

Merci pour ta réponse

Malheureusement, ca n'a rien changer !

J'ai activer le debug, comme tu me l'as dit en faisant:

sip set debug ip freephonie.net

Mais comme je te l'ai déjà dit, je ne voyais pas le peer de la FREE quand je faisais un "sip show peers".
CEPENDANT, après quelques modification, j'ai réussi à obtenir le peer FREE !

SRVVOIP1*CLI> sip show registry
Host Username Refresh State Reg.Time
freephonie.net:5060 0950XXXXXX 1785 Registered Fri, 16 Sep 2011 14:47:29
1 SIP registrations.

SRVVOIP1*CLI> sip set debug ip freephonie.net
SIP Debugging Enabled for IP: 212.27.52.5

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
202/202 10.X.X.X D 58068 OK (118 ms)
201 (Unspecified) D 5060 UNKNOWN
200 (Unspecified) D 5060 UNKNOWN
0950XXXXXX/0950XXXXXX 212.27.52.5 N 5060 Unmonitored
7 sip peers [Monitored: 1 online, 5 offline Unmonitored: 1 online, 0 offline]

SRVVOIP1*CLI> sip set debug peer 0950XXXXXX
SIP Debugging Enabled for IP: 212.27.52.5:5060

SRVVOIP1*CLI> sip reload
Reloading SIP
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 212.27.52.5:5060:
REGISTER sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.X.X:5060;branch=z9hG4bK477a7044;rport
Max-Forwards: 70
From: <sip:0950XXXXXX@freephonie.net>;tag=as0e96a2b7
To: <sip:0950XXXXXX@freephonie.net>
Call-ID: 34d7d78f0fc14276729f15f351e4e41b@DOMAINE
CSeq: 108 REGISTER
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Authorization: Digest username="0950XXXXXX", realm="freephonie.net", algorithm=MD5, uri="sip:freephonie.net", nonce="0fa90c373f73b149039e93a2676ec438", response="7b4f9518e37de25966f0bfc07630f684", opaque="0fa8463130c7edc"
Expires: 120
Contact: <sip:0950XXXXXX@192.168.X.X>
Event: registration
Content-Length: 0

---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 100 Trying
Call-ID: 34d7d78f0fc14276729f15f351e4e41b@DOMAINE
CSeq: 108 REGISTER
From: <sip:0950XXXXXX@freephonie.net>;tag=as0e96a2b7
To: <sip:0950XXXXXX@freephonie.net>
Via: SIP/2.0/UDP 192.168.X.X:5060;received=82.X.X.X;rport=1024;bran ch=z9hG4bK477a7044
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 423 Interval Too Brief
Call-ID: 34d7d78f0fc14276729f15f351e4e41b@DOMAINE
CSeq: 108 REGISTER
From: <sip:0950XXXXXX@freephonie.net>;tag=as0e96a2b7
To: <sip:0950XXXXXX@freephonie.net>;tag=00-30996-1d90330f-06b98d490
Via: SIP/2.0/UDP 192.168.X.X:5060;received=82.X.X.X;rport=1024;bran ch=z9hG4bK477a7044
Min-Expires: 1800
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 212.27.52.5:5060:
REGISTER sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.X.X:5060;branch=z9hG4bK38eedc76;rport
Max-Forwards: 70
From: <sip:0950XXXXXX@freephonie.net>;tag=as15abc832
To: <sip:0950XXXXXX@freephonie.net>
Call-ID: 34d7d78f0fc14276729f15f351e4e41b@DOMAINE
CSeq: 109 REGISTER
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Authorization: Digest username="0950XXXXXX", realm="freephonie.net", algorithm=MD5, uri="sip:freephonie.net", nonce="0fa90c373f73b149039e93a2676ec438", response="7b4f9518e37de25966f0bfc07630f684", opaque="0fa8463130c7edc"
Expires: 1800
Contact: <sip:0950XXXXXX@192.168.X.X>
Event: registration
Content-Length: 0

---
Really destroying SIP dialog '34d7d78f0fc14276729f15f351e4e41b@DOMAINE' Method: REGISTER
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 100 Trying
Call-ID: 34d7d78f0fc14276729f15f351e4e41b@DOMAINE
CSeq: 109 REGISTER
From: <sip:0950XXXXXX@freephonie.net>;tag=as15abc832
To: <sip:0950XXXXXX@freephonie.net>
Via: SIP/2.0/UDP 192.168.X.X:5060;received=82.X.X.X;rport=1024;bran ch=z9hG4bK38eedc76
Content-Length: 0

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

<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 401 Nonce has changed
Call-ID: 34d7d78f0fc14276729f15f351e4e41b@DOMAINE
CSeq: 109 REGISTER
From: <sip:0950XXXXXX@freephonie.net>;tag=as15abc832
To: <sip:0950XXXXXX@freephonie.net>;tag=00-08120-0fa95045-2da3a7837
Via: SIP/2.0/UDP 192.168.X.X:5060;received=82.X.X.X;rport=1024;bran ch=z9hG4bK38eedc76
WWW-Authenticate: Digest realm="freephonie.net",nonce="0fa94f1f4e0994753214 657774812b96",opaque="0fa8463130c7edc",stale=true, algorithm=MD5
Server: Cirpack/v4.42q (gw_sip)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Responding to challenge, registration to domain/host name freephonie.net
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 212.27.52.5:5060:
REGISTER sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.X.X:5060;branch=z9hG4bK39ac43fe;rport
Max-Forwards: 70
From: <sip:0950XXXXXX@freephonie.net>;tag=as49ae63ee
To: <sip:0950XXXXXX@freephonie.net>
Call-ID: 34d7d78f0fc14276729f15f351e4e41b@DOMAINE
CSeq: 110 REGISTER
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Authorization: Digest username="0950XXXXXX", realm="freephonie.net", algorithm=MD5, uri="sip:freephonie.net", nonce="0fa94f1f4e0994753214657774812b96", response="c9ed97fca8892eb17bd93f25f9b43185", opaque="0fa8463130c7edc"
Expires: 1800
Contact: <sip:0950XXXXXX@192.168.X.X>
Event: registration
Content-Length: 0

---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 100 Trying
Call-ID: 34d7d78f0fc14276729f15f351e4e41b@DOMAINE
CSeq: 110 REGISTER
From: <sip:0950XXXXXX@freephonie.net>;tag=as49ae63ee
To: <sip:0950XXXXXX@freephonie.net>
Via: SIP/2.0/UDP 192.168.X.X:5060;received=82.X.X.X;rport=1024;bran ch=z9hG4bK39ac43fe
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 200 OK
Call-ID: 34d7d78f0fc14276729f15f351e4e41b@DOMAINE
Contact: <sip:0950XXXXXX@192.168.X.X>;expires=1800
CSeq: 110 REGISTER
From: <sip:0950XXXXXX@freephonie.net>;tag=as49ae63ee
To: <sip:0950XXXXXX@freephonie.net>;tag=00-08120-0fa95047-303823dc4
Via: SIP/2.0/UDP 192.168.X.X:5060;received=82.X.X.X;rport=1024;bran ch=z9hG4bK39ac43fe
P-Associated-URI: <sip:0950XXXXXX@freephonie.net>
Server: Cirpack/v4.42q (gw_sip)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '34d7d78f0fc14276729f15f351e4e41b@DOMAINE' in 32000 ms (Method: REGISTER)
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>

Quand je passe un appel de 0950XXXXXX à 202, voici le résultat:

7 sip peers [Monitored: 1 online, 5 offline Unmonitored: 1 online, 0 offline]
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] ChanIsAvail("SIP/202-00000000", "SIP/0950XXXXXX@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/202-00000000", "SIP/0950513080@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/202-00000000", "") in new stack
== Spawn extension (appel_interne_X, 90950XXXXXX, 3) exited non-zero on 'SIP/202-00000000'
SRVVOIP1*CLI>

Quand je passe un appel de 202 à 0950XXXXXX, voici le résultat:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] ChanIsAvail("SIP/202-00000001", "SIP/0950XXXXXX@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/202-00000001", "SIP/0950XXXXXX@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/202-00000001", "") in new stack
== Spawn extension (appel_interne_X, 90950XXXXXX, 3) exited non-zero on 'SIP/202-00000001'
SRVVOIP1*CLI>

Nous nous approchons du BUT !

**Reaper** · 16/09/2011, 15h37

defaultexpirey=1800 pour le peer

**LeRenard** · 16/09/2011, 16h02

Envoyé par Reaper

defaultexpirey=1800 pour le peer

C'est fait mais toujours rien, même chose

**Reaper** · 16/09/2011, 16h09

sip show registry vous donne quoi ?

**LeRenard** · 16/09/2011, 16h16

Envoyé par Reaper

sip show registry vous donne quoi ?

La même chose:

SRVVOIP1*CLI> sip show registry
Host Username Refresh State Reg.Time
freephonie.net:5060 0950XXXXXX 1785 Registered Fri, 16 Sep 2011 16:24:13
1 SIP registrations.

EDIT: Après avoir mis la variable "qualify=3000" dans [0950XXXXXX], quand je fais à nouveau un sip show peers, le status du peer FREE est OK

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
202/202 10.X.X.X D 54202 OK (116 ms)
201 (Unspecified) D 5060 UNKNOWN
200 (Unspecified) D 5060 UNKNOWN
0950XXXXXX/0950XXXXXX 212.27.52.5 N 5060 OK (29 ms)
7 sip peers [Monitored: 2 online, 5 offline Unmonitored: 0 online, 0 offline]

**Reaper** · 16/09/2011, 16h30

Si vous êtes en publique ça doit fonctionner, sinon en privé il faut activer le nat=yes pour le peer.
Je ne vois pas l'invite qui part vers free, je vous conseille de vérifier votre plan de numérotation

**LeRenard** · 16/09/2011, 16h37

Envoyé par Reaper

Si vous êtes en publique ça doit fonctionner, sinon en privé il faut activer le nat=yes pour le peer.
Je ne vois pas l'invite qui part vers free, je vous conseille de vérifier votre plan de numérotation

J'ai édit mon précédent post, je sais pas si ca peut aider...

Concernant la variable "nat=yes", c'est déjà fait car oui, je suis en privé !
J'avais poster mes fichiers de config dans le premier post

Peux-tu me dire ce qui ne va pas dans le plan de numérotation ? Merci d'avance.

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