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LeRenard

13/09/2011, 11h07

Bonjour à tous, :hello:

Je pense avoir un problème concernant la configuration mais je ne sais pas d'où viens l'erreur...

En effet, j'ai un problème concernant les appels via le numéro free dans le réseau de l'entreprise (10.X.X.X). Les softphones avec les extensions 200 (et cie) marchent nickel (ca communique entre eux) mais quand je veux appeler le numéro Free de l'entreprise (et vice versa), ca me mets "no route to destination" ou alors "Interdit" ou même "Service or option unavailable".

Voici un petit récapitulatif des liaisons téléphoniques dans un tableau:

http://img820.imageshack.us/img820/489/etatdesliaisons.jpg

J'ai cherché sur différents forum et essayer leurs solutions mais néant.
Donc, je fais appel à vous car là, je suis bloqué ! :frown:

Je vous joins les requêtes CLI ainsi que les fichiers de configurations "extensions.conf" et "sip.conf" ci-dessous pour que vous puissiez y jeter un coup d'oeil avec les données les plus sensibles masqués bien évidement. Merci d'avance de votre aide ! :ouimaitre:

Requêtes CLI

SRVVOIP1*CLI> asterisk -rvvvvvvvvvvvvvvvvvv
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] ChanIsAvail("SIP/200-00000000", "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/200-00000000", "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/200-00000000", "") in new stack
== Spawn extension (appel_interne_X, 90950XXXXXX, 3) exited non-zero on 'SIP/200-00000000'
SRVVOIP1*CLI>

Fichier sip.conf

[general]

Binaddr=0.0.0.0
Bindport=5060
Disallow=all
Allow=ulaw
Context=appel_interne_X
Dtmfmode=rfc2833
Allowoverlap=yes
Tos_sip=cs3
Tos_audio=ef
Tos=lowdelay
srvlookup=yes
language=fr
Context=free-0950XXXXXX
register => 0950XXXXXX:password@freephonie.net/0950XXXXXX

[0950513080]
; Venant du forfait telephonique (free)

type=peer
host=sip.freephonie.net
context=0950XXXXXX
language=fr
insecure=port
defaultuser=0950XXXXXX
secret=password
nat=yes
canreinvite=no
dtmfmode=auto
videosupport=no
restrictcid=no
amaflags=default
defaultexpirey=1800
qualify=3000

[200]

Defaultuser=200
Secret= password
callerid="bob" <200>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
Mailbox=200@voicemail

[201]

Defaultuser=201
Secret= password
callerid="val" <201>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
Mailbox=201@voicemail

[202]

Defaultuser=202
Secret= password
callerid="beber" <202>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
Mailbox=202@voicemail

[sets](!)
; Creation d'une section SIP (extension) pour un poste interne
; A repeter selon le nombre de postes
; Utilisation de template

Defaultuser=200
Secret=password
callerid="bob" <200>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
Mailbox=200@voicemail

[203](sets)
[204](sets)
[205](sets)

; CONFIG IP PHONE CISCO 7960

;[pstn01]
;defaultuser=pstn01
;secret=pstn01
;type=friend
;host=dynamic
;allow=all
;context=appel_interne_X

;[pstn02]
;defaultuser=pstn02
;secret=pstn02
;type=friend
;host=dynamic
;allow=all
;context=appel_interne_X

;[test_cisco]
;defaultuser=test
;secret=test
;type=friend
;host=dynamic
;allow=all
;context=appel_interne_X

Fichier extensions.conf

[appel_interne_X]

Include=>salle_de_conference
Include=>local_voicemail
Include=>horloge_parlante
Include=>auto_attendant
Include=>parkedcalls
Include=>client
Include=>globals
Include=>general
Include=>from-asterisk2
Include=>menu
Include=>menu1
Include=>menu2
Include=>menu3
Include=>free-0950XXXXXX
Include=>test

[general]

static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]

GENERAL=SIP/200&SIP/201&SIP/202&SIP/0950XXXXXX
CONSOLE=Console/dsp
SIPINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1

[client]

; APPELS INTERNES

exten => _2XX,1,Dial(SIP/${EXTEN},20,tr)
exten => _2XX,2,Voicemail(2${EXTEN:1}@default)
exten => _2XX,3,Hangup()

; MESSAGERIE

exten => _3XX,1,Answer()
exten => _3XX,2,Wait(1)
exten => _3XX,3,VoiceMailMain(${EXTEN}@default)
exten => _3XX,4,Hangup()

; APPELS SORTANTS

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

exten => _9.,104,Dial(Zap/1/${EXTEN:1},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

; LOOPBACK DE TEST

exten => 99,1,Answer()
exten => 99,2,SetLanguage(fr)
exten => 99,3,Echo()
exten => 99,4,Playback(vm-goodbye)
exten => 99,5,Hangup()

[Menu]
; standard automatique

exten => s, 1, Background(/var/msg/Menu)
exten => s, 2, WaitExten(2)
exten => s, 3, Goto(Menu,s,1)

exten => 1, 1,SayNumber(1)
exten => 1, 2, goto(Menu1,s,1)
exten => 2, 1, SayNumber(2)
exten => 2, 2, Goto(Menu2,s,1)

exten => 3, 1, SayNumber(3)
exten => 3, 2, Goto(Menu3,s,1)

exten => 9, 1, SayNumber(9)
exten => 9, 2, Hangup()

[Menu1]

exten => s, 1, Background(/var/msg/Menu1)
exten => s, 2, WaitExten(2)
exten => s, 3, Goto(Menu1,s,1)

exten => 1, 1,SayNumber(1)
exten => 1, 2, goto(local,200, 1)

exten => 2, 1, SayNumber(2)
exten => 2, 2, Goto(local,201, 1)

exten => 3, 1, SayNumber(3)
exten => 3, 2, Goto(local,202, 1)

exten => 9, 1, SayNumber(9)
exten => 9, 2, Hangup()

[Menu2]

exten => s, 1, Background(/var/msg/Menu2)
exten => s, 2, WaitExten(2)
exten => s, 3, Goto(Menu2,s,1)

exten => 1, 1, SayNumber(1)
exten => 1, 2, Goto(local,200, 2)

exten => 2, 1, SayNumber(2)
exten => 2, 2, Goto(local,201, 2)

exten => 3, 1, SayNumber(3)
exten => 3, 2, Goto(local,202, 2)

exten => 9, 1, SayNumber(9)
exten => 9, 2, Hangup()

[Menu3]

exten => s, 1, Background(/var/msg/Menu3)
exten => s, 2, WaitExten(2)
exten => s, 3, Goto(Menu3,s,1)

exten => 1, 1, SayNumber(1)
exten => 1, 2, Goto(local,211, 1)

exten => 2, 1, SayNumber(2)
exten => 2, 2, Goto(local,298, 1)

exten => 3, 1, SayNumber(3)
exten => 3, 2, Goto(local,212, 1)

exten => 9, 1, SayNumber(9)
exten => 9, 2, Hangup()

[free-0950XXXXXX]

exten => s,1,Answer

exten => s,2,JABBERSend(gtalk_account,0950XXXXXX:password@s ip.freephonie.net/0950XXXXXX
exten => s,3,Playback(/home/xxxx/voip/voix_recherche_telephone)
exten => s,4,Ringing(4)
exten => s,5,Dial(SIP/200,10,tm)
exten => s,6,Ringing(4)
exten => s,7,Dial(SIP/201,10,tm)
exten => s,8,VoiceMail(1001)
exten => s,9,Hangup(16)

; test IP Phone cisco 7960
;[test]
;exten => cisco_line1,1,Goto(200,1)
;exten => 200,1,Dial(SIP/pstn01)
;exten => 200,2,Hangup()

;exten => cisco_line2,1,Goto(201,1)
;exten => 201,1,Dial(SIP/pstn01)
;exten => 201,2,Hangup()

;exten => 202,1,Dial(SIP/test_cisco)
;exten => 202,2,Hangup()

[local_voicemail]

Exten=>888,1,Answer()
Exten=>888,2,VoiceMailMain()
Exten=>888,3,Hangup()

[horloge_parlante]

Exten=>3344,1,Answer()
Exten=>3344,2,SayUnixTime(,Europe/Paris,AdBY kM)
Exten=>3344,3,Hangup()

[auto_attendant]

Exten=>2233,1,Read(digit,/var/lib/asterisk/sounds/en/hello-word,1)
Exten=>2233,2,Gotoif($["${digit}" = "1"]?appel_interne_X,3344,1)
Exten=>2233,3,Gotoif($["${digit}" = "2"]?appel_interne_X,900,1)
Exten=>2233,4,Gotoif($["${digit}" = "3"]?appel_interne_X,888,1)
Exten=>i,1,Goto(2233,1)

[macro-appel_interne_X] :

Exten=>s,1,Answer()
Exten=>s,2,Dial(${ARG1},20,Ttr)
Exten=>s,3,VoiceMail(${ARG1}@voicemail)
Exten=>s,4,Playback(vm-goodbye)
Exten=>s,5,Hangup()

Reaper

13/09/2011, 19h17

defaultexpirey=1800 dans la section general de sip.conf.
reload la conf et montre nous output de "sip show registry"

LeRenard

14/09/2011, 09h26

Ok, c'est fait ! Voici le résultat:

SRVVOIP1*CLI> sip show registry
Host Username Refresh State Reg.Time
freephonie.net:5060 0950XXXXXX 1785 Registered Wed, 14 Sep 2011 09:31:47
1 SIP registrations.
SRVVOIP1*CLI>

Reaper

14/09/2011, 14h19

Tentative d’appel ça donne quoi ?
Faites un sip debug sur le sip peer de free pour voir.

LeRenard

14/09/2011, 14h51

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

14/09/2011, 15h08

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

LeRenard

14/09/2011, 15h54

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

14/09/2011, 16h03

Avez vous essayé d’appeler ?

LeRenard

14/09/2011, 16h20

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

14/09/2011, 16h26

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

LeRenard

15/09/2011, 09h56

Bonjour, désolé du petit retard :pt1cable:
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",uri="sip:90950XXXXXX@WORKGROUP:5060",response="217b28fe239c388230418edca15ba4f9",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 :wink:

Malheureusement, ca n'a rien changer ! :frown:
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="0fa94f1f4e0994753214657774812b96",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 ! :pt1cable::lol:

Reaper

16/09/2011, 15h37

defaultexpirey=1800 pour le peer

LeRenard

16/09/2011, 16h02

defaultexpirey=1800 pour le peer

C'est fait mais toujours rien, même chose :sweat:

Reaper

16/09/2011, 16h09

sip show registry vous donne quoi ?

LeRenard

16/09/2011, 16h16

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

LeRenard

16/09/2011, 16h37

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

Reaper

16/09/2011, 16h44

Faites un appel vers le 06XXX par le peer free, avec le debug.

LeRenard

16/09/2011, 16h54

Faites un appel vers le 06XXX par le peer free, avec le debug.

Je viens de faire le test en passant par le peer free et en activant le debug, il y a absolument rien dans la CLI !

Le softphone, lui, me dit ceci:
"Vous ne pouvez pas composer ce numéro."

Reaper

16/09/2011, 17h26

Active le verbose 20
core set verbose 20
Et passe un appel, ensuite regarde le dialplan de contexte "appel_interne_X"

dialplan show appel_interne_X

Vous avez numérote 906XX je suppose ?

LeRenard

19/09/2011, 09h31

Exactement la même chose... :frown:
Et oui, j'ai bien utilisé 906XXXXXXXX

SRVVOIP1*CLI> core set verbose 20
Verbosity was 32 and is now 20
SRVVOIP1*CLI>

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '6bd25e715269c1453dc008091a0c2421@DOMAINE' Method: OPTIONS
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90326XXXXXX@appel_interne_X:1] ChanIsAvail("SIP/202-0000000a", "SIP/0326XXXXXX@free-0950XXXXXX") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90326XXXXXX@appel_interne_X:2] Dial("SIP/202-0000000a", "SIP/0326XXXXXX@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 [90326XXXXXX@appel_interne_X:3] Congestion("SIP/202-0000000a", "") in new stack
== Spawn extension (appel_interne_X, 90326XXXXXX, 3) exited non-zero on 'SIP/202-0000000a'
SRVVOIP1*CLI>

SRVVOIP1*CLI> dialplan show appel_interne_X
[ Context 'appel_interne_X' created by 'pbx_config' ]
Include => 'salle_de_conference' [pbx_config]
Include => 'local_voicemail' [pbx_config]
Include => 'horloge_parlante' [pbx_config]
Include => 'auto_attendant' [pbx_config]
Include => 'parkedcalls' [pbx_config]
Include => 'client' [pbx_config]
Include => 'globals' [pbx_config]
Include => 'general' [pbx_config]
Include => 'from-asterisk2' [pbx_config]
Include => 'menu' [pbx_config]
Include => 'menu1' [pbx_config]
Include => 'menu2' [pbx_config]
Include => 'menu3' [pbx_config]
Include => 'free-0950XXXXXX' [pbx_config]
Include => 'test' [pbx_config]

-= 0 extensions (0 priorities) in 1 context. =-
SRVVOIP1*CLI>

Reaper

19/09/2011, 09h45

Pouvez vous retirer les lignes:

exten => _9.,1,ChanIsAvail(SIP/${EXTEN:1}@free-0950XXXXXX)
et la congestion, laisse que le dial. avec priorité 1.
Test ?

LeRenard

19/09/2011, 10h40

Rien du tout !
Je désespères :pt1cable:

Appels:

202 -> 90950XXXXXX = La destination non trouvée

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] Dial("SIP/202-00000000", "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)
-- Auto fallthrough, channel 'SIP/202-00000000' status is 'CHANUNAVAIL'
SRVVOIP1*CLI>

0950XXXXXX -> 202 = L'indicatif que vous avez demander n'est pas utilisé. Votre appel ne peut aboutir...
Rien dans la CLI

Reaper

19/09/2011, 10h48

LeRenard

19/09/2011, 11h07

Vous êtes certain que vous avez activé le debug chaque fois sur le peer de free ?
Et autre chose, vous faite un deial vers free-09xxx; il est ou votre peer free-09xxxx ?

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

Oui, en faisant "sip set debug peer 0950XXXXXX"
exten => _9.,1,Dial(SIP/${EXTEN:1}@free-0950XXXXXX),30,r) est dans le context [client]

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] Dial("SIP/202-00000004", "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)
-- Auto fallthrough, channel 'SIP/202-00000004' status is 'CHANUNAVAIL'
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.X.X:5060;branch=z9hG4bK48d723cf;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@DOMAINE>;tag=as4d70df1b
To: <sip:freephonie.net>
Contact: <sip:Unknown@192.168.X.X>
Call-ID: 5c81be7648e25bcc6a86232355416fd0@DOMAINE
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Mon, 19 Sep 2011 09:13:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 501 Not Implemented Yet
Call-ID: 5c81be7648e25bcc6a86232355416fd0@DOMAINE
CSeq: 102 OPTIONS
From: "Unknown" <sip:Unknown@DOMAINE>;tag=as4d70df1b
To: <sip:freephonie.net>;tag=00-30951-1e71b144-6d2ba8477
Via: SIP/2.0/UDP 192.168.X.X:5060;received=82.X.X.X;rport=1024;bran ch=z9hG4bK48d723cf
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '5c81be7648e25bcc6a86232355416fd0@DOMAINE' Method: OPTIONS
SRVVOIP1*CLI>

Reaper

19/09/2011, 11h41

Encore une fois

cette ligne:

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

Au lieu de:

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

Test ?

LeRenard

19/09/2011, 11h52

Ca donne la même chose...(sans le free-) :wink:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] Dial("SIP/202-00000000", "SIP/0950XXXXXX@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)
-- Auto fallthrough, channel 'SIP/202-00000000' status is 'CHANUNAVAIL'
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.X.X:5060;branch=z9hG4bK64a12463;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@DOMAINE>;tag=as7e3e65bc
To: <sip:freephonie.net>
Contact: <sip:Unknown@192.168.X.X>
Call-ID: 413b10222d56588e1a01d93a255f0608@DOMAINE
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Mon, 19 Sep 2011 09:57:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 501 Not Implemented Yet
Call-ID: 413b10222d56588e1a01d93a255f0608@DOMAINE
CSeq: 102 OPTIONS
From: "Unknown" <sip:Unknown@DOMAINE>;tag=as7e3e65bc
To: <sip:freephonie.net>;tag=00-32535-1e73fa81-3df00d4d2
Via: SIP/2.0/UDP 192.168.X.X:5060;received=82.X.X.X;rport=1024;bran ch=z9hG4bK64a12463
Content-Length: 0

Reaper

19/09/2011, 11h56

Pouvez vous me donner un acess root sur votre machine ?
J'ai pas le temps de deviner, ça doit être quelque chose très simple.
Ou sinon vous pouvez effacer tout, créer un peer Free, et un compte de téléphone, et essayer appeler par ce peer. Je vous invite de re-faire votre installation, de trouver un tutoriel sip free (des tonnes sur internet) et le suivre

LeRenard

20/09/2011, 15h55

LeRenard

20/09/2011, 15h55

Bonjour !

Après quelques recherches et modifications personnelles, j'ai bien avancé !
En effet, j'ai réussi à faire fonctionner certaines communications (voir ci-dessous).
Cependant, il me reste un petit problème au niveau du numéro free...comme vous pouvez le voir ci-dessous également:

0950XXXXXX vers 201 = OK ! =)
0950XXXXXX vers 202 = OK ! =)
0950XXXXXX vers 0950XXXXXX = FAIL ! :(

201 vers 202 = OK ! =)
201 vers 201 = OK ! =)
201 vers 0950XXXXXX = FAIL ! :(

202 vers 201 = OK ! =)
202 vers 202 = OK ! =)
202 vers 0950XXXXXX = FAIL ! :(

Voici le résultat dans la CLI:

SRVVOIP1*CLI> sip set debug peer 0950XXXXXX
SIP Debugging Enabled for IP: 212.27.52.5:5060
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 82.X.X.X:5060;branch=z9hG4bK1f55abcb;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@DOMAIN>;tag=as78adad8c
To: <sip:freephonie.net>
Contact: <sip:Unknown@82.X.X.X>
Call-ID: 1bf6409a4e27c9cd078a9d5f5722552a@DOMAIN
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Tue, 20 Sep 2011 13:46:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 501 Not Implemented Yet
Call-ID: 1bf6409a4e27c9cd078a9d5f5722552a@DOMAIN
CSeq: 102 OPTIONS
From: "Unknown" <sip:Unknown@DOMAIN>;tag=as78adad8c
To: <sip:freephonie.net>;tag=00-31101-1ed40e68-735604160
Via: SIP/2.0/UDP 82.X.X.X:5060;received=82.X.X.X;rport=1024;branch= z9hG4bK1f55abcb
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '1bf6409a4e27c9cd078a9d5f5722552a@DOMAIN' Method: OPTIONS
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] Ringing("SIP/202-00000059", "") in new stack
-- Executing [90950XXXXXX@appel_interne_X:2] Wait("SIP/202-00000059", "") in new stack
-- Executing [90950XXXXXX@appel_interne_X:3] Answer("SIP/202-00000059", "") in new stack
-- Executing [90950XXXXXX@appel_interne_X:4] Dial("SIP/202-00000059", "SIP/0950XXXXXX@freephonie.net),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)
SRVVOIP1*CLI>

Le problème est bien au niveau de:
== Everyone is busy/congested at this time (1:0/0/1)

Avez-vous une idée ?! Merci d'avance.

LeRenard

21/09/2011, 11h08

Pour information:

J'ai fais un test en appelant le numéro free 0950XXXXXX via mon téléphone portable 06XXXXXXXX. Voici le résultat que la CLI m'a sorti:

SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
INVITE sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 23528-QA-102f2397-369c648f4@freephonie.net
Contact: <sip:172.X.X.X:5062>
Content-Type: application/sdp
CSeq: 264484018 INVITE
From: "0674420004" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=23528-XD-102f2398-54dd3b156
Max-Forwards: 27
Record-Route: <sip:C=on;t=UBFVF@212.27.52.5:5060;lr>
To: <sip:0950XXXXXX@172.X.X.X;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-UBFV-0e04cd9b-1b10a8d8
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.42q (gw_sip)
Content-Length: 184

v=0
o=cp10 131659623199 131659623199 IN IP4 172.X.X.X
s=SIP Call
c=IN IP4 212.27.52.129
t=0 0
m=audio 31122 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 10 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 212.27.52.5 : 5060 (NAT)
Using INVITE request as basis request - 23528-QA-102f2397-369c648f4@freephonie.net
No user '06XXXXXXXX' in SIP users list
Found peer '0950XXXXXX-in' for '06XXXXXXXX' from 212.27.52.5:5060
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x4 (ulaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
SRVVOIP1*CLI>
<--- Reliably Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-UBFV-0e04cd9b-1b10a8d8;received=212.27.52.5
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=23528-XD-102f2398-54dd3b156
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as73716753
Call-ID: 23528-QA-102f2397-369c648f4@freephonie.net
CSeq: 264484018 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

<------------>
Scheduling destruction of SIP dialog '23528-QA-102f2397-369c648f4@freephonie.net' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
ACK sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 23528-QA-102f2397-369c648f4@freephonie.net
CSeq: 264484018 ACK
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=23528-XD-102f2398-54dd3b156
Max-Forwards: 27
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as73716753
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-UBFV-0e04cd9b-1b10a8d8
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '23528-QA-102f2397-369c648f4@freephonie.net' Method: ACK
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>

Reaper

21/09/2011, 13h31

Found peer '0950XXXXXX-in' for '06XXXXXXXX' from 212.27.52.5:5060
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x4 (ulaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
SRVVOIP1*CLI>
<--- Reliably Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 488 Not acceptable here

Pour le peer 0950XXXXXX-in il faut authoriser le codec alaw
[0950XXXXXX-in]
allow=alaw

LeRenard

21/09/2011, 14h01

Ok, c'est fait :wink:

SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
INVITE sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
Contact: <sip:172.X.X.X:5062>
Content-Type: application/sdp
CSeq: 264728047 INVITE
From: "06XXXXXXXX" <sip:06XXXXXX@freephonie.net;user=phone>;tag=30286-NB-1032fe7d-0553aa8e1
Max-Forwards: 27
Record-Route: <sip:C=on;t=EEZIZ@212.27.52.5:5060;lr>
To: <sip:0950XXXXXX@172.X.X.X;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-EEZI-0e094372-21092597
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.42q (gw_sip)
Content-Length: 184

v=0
o=cp10 131660620697 131660620697 IN IP4 172.X.X.X
s=SIP Call
c=IN IP4 212.27.52.130
t=0 0
m=audio 37620 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 10 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 212.27.52.5 : 5060 (NAT)
Using INVITE request as basis request - 30286-IQ-1032fe7c-7986080b4@freephonie.net
No user '06XXXXXXXX' in SIP users list
Found peer '0950XXXXXX-out' for '06XXXXXXXX' from 212.27.52.5:5060
SRVVOIP1*CLI>
<--- Reliably Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-EEZI-0e094372-21092597;received=212.27.52.5
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=30286-NB-1032fe7d-0553aa8e1
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as5652bb63
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
CSeq: 264728047 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="219a361b"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '30286-IQ-1032fe7c-7986080b4@freephonie.net' in 6400 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
ACK sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
CSeq: 264728047 ACK
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=30286-NB-1032fe7d-0553aa8e1
Max-Forwards: 27
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as5652bb63
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-EEZI-0e094372-21092597
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
INVITE sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Authorization: Digest username="anonymous",realm="asterisk",nonce="219a361b",uri="sip:0950XXXXXX@82.X.X.X:5060",response="86c534089edcb328ba1165c683e70308",algorithm=MD5,opaque=""
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
Contact: <sip:172.X.X.X:5062>
Content-Type: application/sdp
CSeq: 264728049 INVITE
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=30286-NB-1032fe7d-0553aa8e1
Max-Forwards: 26
Record-Route: <sip:C=on;t=RQXKY@212.27.52.5:5060;lr>
To: <sip:0950XXXXXX@172.X.X.X;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RQXK-0e094374-32cd0b3e
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.42q (gw_sip)
Content-Length: 184

v=0
o=cp10 131660620699 131660620699 IN IP4 172.X.X.X
s=SIP Call
c=IN IP4 212.27.52.129
t=0 0
m=audio 36718 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv

<------------->
--- (14 headers 10 lines) ---
Sending to 212.27.52.5 : 5060 (NAT)
Using INVITE request as basis request - 30286-IQ-1032fe7c-7986080b4@freephonie.net
No user '06XXXXXXXX' in SIP users list
Found peer '0950XXXXXX-out' for '06XXXXXXXX' from 212.27.52.5:5060

<--- Reliably Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RQXK-0e094374-32cd0b3e;received=212.27.52.5
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=30286-NB-1032fe7d-0553aa8e1
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as5652bb63
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
CSeq: 264728049 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

<------------>
Scheduling destruction of SIP dialog '30286-IQ-1032fe7c-7986080b4@freephonie.net' in 6400 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
ACK sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 30286-IQ-1032fe7c-7986080b4@freephonie.net
CSeq: 264728049 ACK
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=30286-NB-1032fe7d-0553aa8e1
Max-Forwards: 26
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as5652bb63
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RQXK-0e094374-32cd0b3e
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>

Reaper

21/09/2011, 14h07

Il faut ajouter unsecure=invite pour le matched peer, mais vous avez un souci la, au début l'appel est identifié avec 0950XXXXXX-in, après sous 0950XXXXXX-out vous avez deux peers ou vous avez changé le nom de 0950XXXXXX-in sur 0950XXXXXX-out ? je vous conseille d'avoir seulement un peer 0950XXXXXX qui a le même nom que le numéro de téléphone.

LeRenard

21/09/2011, 16h51

J'ai rectifier et assembler le tout car je voulais tester séparément ce que ca donnerait... mais toujours le même problème.
Désolé de t'embêter Reaper :frown::lol:

Reaper

21/09/2011, 18h34

Et bien encore une fois poste les traces.

LeRenard

22/09/2011, 09h45

Voici les traces:

SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
INVITE sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 03899-OL-104abd3b-1767bae27@freephonie.net
Contact: <sip:172.X.X.X:5062>
Content-Type: application/sdp
CSeq: 266246805 INVITE
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=03899-BR-104abd3c-7d5952bc2
Max-Forwards: 27
Record-Route: <sip:C=on;t=CNFBF@212.27.52.5:5060;lr>
To: <sip:0950XXXXXX@172.X.X.X;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-CNFB-0e2c7845-2a7d002a
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.42q (gw_sip)
Content-Length: 184

v=0
o=cp10 131667697362 131667697362 IN IP4 172.X.X.X
s=SIP Call
c=IN IP4 212.27.52.129
t=0 0
m=audio 36848 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 10 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 212.27.52.5 : 5060 (NAT)
Using INVITE request as basis request - 03899-OL-104abd3b-1767bae27@freephonie.net
No user '06XXXXXXXX' in SIP users list
Found peer '0950XXXXXX' for '06XXXXXXXX' from 212.27.52.5:5060
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.27.52.129:36848
Looking for 0950XXXXXX in appel_interne_X (domain 82.X.X.X)
SRVVOIP1*CLI>
<--- Reliably Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-CNFB-0e2c7845-2a7d002a;received=212.27.52.5
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=03899-BR-104abd3c-7d5952bc2
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as2921bb00
Call-ID: 03899-OL-104abd3b-1767bae27@freephonie.net
CSeq: 266246805 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

<------------>
Scheduling destruction of SIP dialog '03899-OL-104abd3b-1767bae27@freephonie.net' in 6400 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
ACK sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 03899-OL-104abd3b-1767bae27@freephonie.net
CSeq: 266246805 ACK
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=03899-BR-104abd3c-7d5952bc2
Max-Forwards: 27
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as2921bb00
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-CNFB-0e2c7845-2a7d002a
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '03899-OL-104abd3b-1767bae27@freephonie.net' Method: ACK
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>

Je te joins également les fichiers sip.con et extensions.conf pour que tu puisses voir les modifications et m'apporter peut-être une solution :)

sip.conf

[general]

Binaddr=0.0.0.0
Bindport=5060
Disallow=all
Allow=ulaw
Allow=alaw
Context=appel_interne_X
Dtmfmode=rfc2833
Allowoverlap=yes
Tos_sip=cs3
Tos_audio=ef
Tos=lowdelay
srvlookup=yes
language=fr
register => 0950XXXXXX:passwordsipfree@freephonie.net/0950XXXXXX
defaultexpirey=1800
fromdomain=freephonie.net
externip = 82.X.X.X
localnet=10.X.X.X
nat=yes

[0950513080]

type=friend
disallow=all
allow=alaw
host=freephonie.net
context=appel_interne_X
language=fr
insecure=invite
defaultuser=0950XXXXXX
secret=passwordsipfree
callerid="freebox" <0950XXXXXX>
nat=yes
canreinvite=no
dtmfmode=inband
videosupport=no
restrictcid=no
amaflags=default
defaultexpirey=1800
qualify=yes
fromuser=0950XXXXXX
fromdomain=freephonie.net

[202]

Defaultuser=202
Secret=3456
callerid="beber" <202>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
dtmf=inband
Mailbox=202

extensions.conf

[appel_interne_X]

Include=>salle_de_conference
Include=>local_voicemail
Include=>horloge_parlante
Include=>auto_attendant
Include=>parkedcalls
Include=>client
Include=>general

[general]

static=yes
writeprotect=no
autofallthrough=no
clearglobalvars=no
priorityjumping=no

[globals]

GENERAL=SIP/200&SIP/201&SIP/202&SIP/0950XXXXXX
CONSOLE=Console/dsp
SIPINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1

[client]

exten => _2XX,1,Answer()
exten => _2XX,2,Dial(SIP/${EXTEN},20,tr)
exten => _2XX,3,Voicemail(2${EXTEN}@default)
exten => _2XX,4,Hangup()

; MESSAGERIE
exten => _3XX,1,Answer()
exten => _3XX,2,Wait(1)
exten => _3XX,3,VoiceMailMain(${EXTEN}@default)
exten => _3XX,4,Hangup()

; APPELS ENTRANTS

exten => s,1,Answer
exten => s,2,Dial(SIP/${EXTEN}@0950XXXXXX),10,tm)
exten => s,3,VoiceMail(888)
exten => s,4,Hangup()

; APPELS SORTANTS

exten => _90[1-6]XXXXXXXX,1,Ringing()
exten => _90[1-6]XXXXXXXX,2,Wait()
exten => _90[1-6]XXXXXXXX,3,Answer()
exten => _90[1-6]XXXXXXXX,4,Dial(SIP/${EXTEN:1}@0950XXXXXX),30,r)

; LOOPBACK DE TEST

exten => 99,1,Answer()
exten => 99,2,SetLanguage(fr)
exten => 99,3,Echo()
exten => 99,4,Playback(vm-goodbye)
exten => 99,5,Hangup()

[local_voicemail]

Exten=>888,1,Answer()
Exten=>888,2,VoiceMailMain()
Exten=>888,3,Hangup()

[horloge_parlante]

Exten=>3344,1,Answer()
Exten=>3344,2,SayUnixTime(,Europe/Paris,AdBY kM)
Exten=>3344,3,Hangup()

[auto_attendant]

Exten=>2233,1,Read(digit,/var/lib/asterisk/sounds/en/hello-word,1)
Exten=>2233,2,Gotoif($["${digit}" = "1"]?appel_interne_X,3344,1)
Exten=>2233,3,Gotoif($["${digit}" = "2"]?appel_interne_X,900,1)
Exten=>2233,4,Gotoif($["${digit}" = "3"]?appel_interne_X,888,1)
Exten=>i,1,Goto(2233,1)

[macro-appel_interne_X] :

Exten=>s,1,Answer()
Exten=>s,2,Dial(${ARG1},20,Ttr)
Exten=>s,3,VoiceMail(${ARG1}@voicemail)
Exten=>s,4,Playback(vm-goodbye)
Exten=>s,5,Hangup()

Reaper

22/09/2011, 10h44

Votre appel arrive sut votre standard, mais asterisk ne trouve pas que faire avec, dans votre dialplan:

; APPELS ENTRANTS

; APPELS ENTRANTS

exten => s,1,Answer
exten => s,2,Dial(SIP/${EXTEN}@0950XXXXXX),10,tm)
exten => s,3,VoiceMail(888)
exten => s,4,Hangup()

Vous avez redirigé l'appel entrant en sortant ? Je ne comprends pas. Enfin bref, nous allons supposer qu il faut sonner utilisateur 202 lorsque 09 est appelé, il faut changer ces lignes sur:

; APPELS ENTRANTS

exten => 09XXXXX,1,Dial(SIP/200)

TEST ?

LeRenard

22/09/2011, 11h18

AH !! C'est bon ca ! :D
Les appels extérieurs sont bien redirigé vers l'extension 202 :)

Voici le résultat:

SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
INVITE sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 24776-MF-104cd6ec-331029352@freephonie.net
Contact: <sip:172.X.X.X:5062>
Content-Type: application/sdp
CSeq: 266378831 INVITE
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=24776-CU-104cd6ed-352a20ba5
Max-Forwards: 27
Record-Route: <sip:C=on;t=SECSA@212.27.52.5:5060;lr>
To: <sip:0950XXXXXX@172.X.X.X;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-SECS-0e2f16c4-3587a51c
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.42q (gw_sip)
Content-Length: 184

v=0
o=cp10 131668242911 131668242911 IN IP4 172.X.X.X
s=SIP Call
c=IN IP4 212.27.52.130
t=0 0
m=audio 32628 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 10 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 212.27.52.5 : 5060 (NAT)
Using INVITE request as basis request - 24776-MF-104cd6ec-331029352@freephonie.net
No user '06XXXXXXXX' in SIP users list
Found peer '0950XXXXXX' for '06XXXXXXXX' from 212.27.52.5:5060
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 212.27.52.130:32628
Looking for 0950XXXXXX in appel_interne_X (domain 82.X.X.X)
list_route: hop: <sip:C=on;t=SECSA@212.27.52.5:5060;lr>

<--- Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-SECS-0e2f16c4-3587a51c;received=212.27.52.5
Record-Route: <sip:C=on;t=SECSA@212.27.52.5:5060;lr>
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=24776-CU-104cd6ed-352a20ba5
To: <sip:0950XXXXXX@172.X.X.X;user=phone>
Call-ID: 24776-MF-104cd6ec-331029352@freephonie.net
CSeq: 266378831 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:0950XXXXXX@82.X.X.X>
Content-Length: 0

<------------>
-- Executing [0950XXXXXX@appel_interne_X:1] Dial("SIP/0950XXXXXX-00000013", "SIP/202") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 202
-- SIP/202-00000014 is ringing
SRVVOIP1*CLI>
<--- Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-SECS-0e2f16c4-3587a51c;received=212.27.52.5
Record-Route: <sip:C=on;t=SECSA@212.27.52.5:5060;lr>
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=24776-CU-104cd6ed-352a20ba5
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as6eb4b7c8
Call-ID: 24776-MF-104cd6ec-331029352@freephonie.net
CSeq: 266378831 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:0950XXXXXX@82.X.X.X>
Content-Length: 0

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

Cependant, je ne veux pas redirigé les appels entrants en sortants...
Je veux que ca sonne sur l'extension 0950XXXXXX ! Comment ?

On y est presque, merci beaucoup Reaper !! :ouimaitre:
Maintenant, il faut que je puisses contacter via l'extension 202, l'extérieur !

Quand, je fais un appel de 202 à 0326XXXXXX (numéro interne dans l'entreprise), j'obtiens ceci:

SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90326XXXXXX@appel_interne_X:1] Ringing("SIP/202-0000001c", "") in new stack
-- Executing [90326XXXXXX@appel_interne_X:2] Wait("SIP/202-0000001c", "") in new stack
-- Executing [90326XXXXXX@appel_interne_X:3] Answer("SIP/202-0000001c", "") in new stack
-- Executing [90326XXXXXX@appel_interne_X:4] Dial("SIP/202-0000001c", "SIP/0326XXXXXX@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)
SRVVOIP1*CLI>

Quand, je fais un appel de 202 à 06XXXXXXXX, j'obtiens ceci:

SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [906XXXXXXXX@appel_interne_X:1] Ringing("SIP/202-0000001e", "") in new stack
-- Executing [906XXXXXXXX@appel_interne_X:2] Wait("SIP/202-0000001e", "") in new stack
-- Executing [906XXXXXXXX@appel_interne_X:3] Answer("SIP/202-0000001e", "") in new stack
-- Executing [906XXXXXXXX@appel_interne_X:4] Dial("SIP/202-0000001e", "SIP/06XXXXXXXX@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)
SRVVOIP1*CLI>

Quand, je fais un appel de 202 à 0950XXXXXX, j'obtiens RIEN !

Reaper

22/09/2011, 14h09

LeRenard

22/09/2011, 14h25

Je vous conseille de lire le livre d'asterisk, surtout à propos de contextes, vous avez des questions de base qui sont traites dans la partie dialplan de livre. Vous pouvez commencer par lire le petit tuto sur notre site: http://www.asterisk-france.org/content.php/64-Les-contextes

Ok, je vais voir ca...

Une dernière chose,

Vous etes entrain de contacter le numéro 0950XXXXXX par 200 qui est en fait votre fournisseur et qui renvoie sur 200, je ne comprends pas le sens de test.

Le but était de faire un test et d'accéder à la messagerie free directement comme si sur un portable ou autre, on s'appelle nous même sur la même ligne et on tombe directement sur la messagerie.

Et autre moment free vous donne seulement 1 appel en même temps sur le compte.

Effectivement, j'avais complètement zappé que free = 1 seul appel en même temps...

Dans tous les cas, je tiens vraiment a te remercier Reaper !! :ouimaitre:
Tu m'as vraiment beaucoup aider et éclaircie sur certains points :)

benjborhen

22/09/2011, 16h28

LeRenard

22/09/2011, 16h40

Salut,

dans la déclaration de ton trunk SIP il faut le dissocier sur 2 parties.

Une partie en type=peer ==> émettre les appels
Une partie en type=user ==> recevoir les appels ( avec spécification du context pour la réception).

Je vois que la partie peer. ajoute la partie user et montre nous ce que tu as .

Faudrait vous mettre d'accord, je plaisantes :lol:
Reaper m'a dis de regrouper le tout :wink:

C'est pour ca que dans le contexte [0950XXXXXX], la variable type est à "friend" ce qui permet d'émettre et de recevoir.

Me trompe-je ?! :gratgrat:

PS: Ne pas regarder les premiers POST ou il y a les fichiers sip.conf et extensions.conf
Ces fichiers ont était mis à jour dans le POST N°40 en page 4

LeRenard

27/09/2011, 12h05

Bonjour à vous ! :D

Après de nombreuses modifications, j'ai réussi à mettre en place les appels comme vous pouvez le voir sur le schéma ci-dessous:

http://nsa27.casimages.com/img/2011/09/27/110927122021995142.jpg

Je vous joins la configuration des fichiers SIP.conf et EXTENSIONS.conf.

sip.conf

[general]

Binaddr=0.0.0.0
Bindport=5060
Disallow=all
Allow=ulaw
Allow=alaw
context=0950XXXXXX
Dtmfmode=rfc2833
Allowoverlap=yes
Tos_sip=cs3
Tos=lowdelay
srvlookup=yes
language=fr
register => 0950XXXXXX:password@freephonie.net/0950XXXXXX
defaultexpirey=1800
fromdomain=DOMAINE
externip = 82.X.X.X
localnet=10.200.X.X/255.255.0.0
nat=yes

[0950XXXXXX]

type=friend
disallow=all
allow=alaw
host=freephonie.net
context=client
language=fr
insecure=invite,port
defaultuser=0950XXXXXX
secret=password
callerid="freebox" <0950XXXXXX>
nat=yes
canreinvite=no
dtmfmode=inband
videosupport=no
restrictcid=no
amaflags=default
defaultexpirey=1800
qualify=yes
fromuser=0950XXXXXX
fromdomain=freephonie.net

[200]

Defaultuser=200
Secret=1234
callerid="bob" <200>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
dtmf=inband
Directmedia=yes
Mailbox=200

[201]

Defaultuser=201
Secret=2345
callerid="val" <201>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
dtmf=inband
Mailbox=201

[202]

Defaultuser=202
Secret=3456
callerid="beber" <202>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
dtmf=inband
Mailbox=202

extensions.conf

[appel_interne_X]
; declaration du dialplan "appel_interne_X"

Include=>salle_de_conference
; inclus le context dans "appel_interne_X"
Include=>local_voicemail
Include=>horloge_parlante
Include=>auto_attendant
Include=>parkedcalls
Include=>client
Include=>general
Include=>globals

[general]

static=yes
writeprotect=no
autofallthrough=no
clearglobalvars=no
priorityjumping=no

[globals]

GENERAL=SIP/2XX&SIP/0950XXXXXX
CONSOLE=Console/dsp
SIPINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1

;[free-in]
;exten => _.,1,Goto(client,200,1)

[client]

exten => _2XX,1,Answer()
exten => _2XX,2,Dial(SIP/${EXTEN},20,tr)
exten => _2XX,3,Voicemail(2${EXTEN}@default)
exten => _2XX,4,Hangup()

; MESSAGERIE
exten => _3XX,1,Answer()
exten => _3XX,2,Wait(1)
exten => _3XX,3,VoiceMailMain(${EXTEN}@default)
exten => _3XX,4,Hangup()

; APPELS ENTRANTS

exten => 0950XXXXXX,1,Dial(SIP/200)
; redirection des appels entrants en sortants vers une extension comme par exemple 200 qui fera standard

; APPELS SORTANTS

;exten => _90[1-5]XXXXXXXX,1,Ringing()
;exten => _90[1-5]XXXXXXXX,2,Wait()
;exten => _90[1-5]XXXXXXXX,3,Answer()
;exten => _90[1-5]XXXXXXXX,4,Dial(SIP/${EXTEN:1}@0950XXXXXX),30,r)
;exten => _90[1-5]XXXXXXXX,5,Congestion

; ----------------------------------------
; Pour passer un appel (en france)
exten => _00[12345]XXXXXXXX,1,Dial(SIP/0950XXXXXX/${EXTEN:1},240,T)
exten => _00[12345]XXXXXXXX,2,Hangup

; Numero GRATUIT
exten => _0080[05]XXXXXX,1,Dial(SIP/0950XXXXXX/${EXTEN:1},240,T)
exten => _0080[05]XXXXXX,2,Hangup

; Numero INTERNET
exten => _009XXXXXXXX,1,Dial(SIP/0950XXXXXX/${EXTEN:1},240,T)
exten => _009XXXXXXXX,2,Hangup
; -----------------------------------------

; LOOPBACK DE TEST

exten => 99,1,Answer()
exten => 99,2,SetLanguage(fr)
exten => 99,3,Echo()
exten => 99,4,Playback(vm-goodbye)
exten => 99,5,Hangup()

[local_voicemail]

Exten=>888,1,Answer()
Exten=>888,2,VoiceMailMain()
Exten=>888,3,Hangup()

[horloge_parlante]

Exten=>3344,1,Answer()
Exten=>3344,2,SayUnixTime(,Europe/Paris,AdBY kM)
Exten=>3344,3,Hangup()

LeRenard

27/09/2011, 12h06

Je fais appel à vous une dernière fois !
En effet, comme vous pouvez le constater sur le schéma précédent, j'ai un problème au niveau des appels du peer 200.

Voici les résultats que la CLI me dit:

APPEL 200 vers 202

SRVVOIP1*CLI> sip set debug peer 200
SIP Debugging Enabled for IP: 10.200.X.X:8024
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8024 --->
INVITE sip:202@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-186b4174337703ac-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:200@10.200.X.X:8024>
To: <sip:202@DOMAINE>
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 488

v=0
o=- 12961589563258089 1 IN IP4 10.200.X.X
s=CounterPath X-Lite 4.1
c=IN IP4 10.200.X.X
t=0 0
a=ice-ufrag:a323b6
a=ice-pwd:3d8bb9985126dad4d71a6bace385a126
m=audio 8058 RTP/AVP 100 97 105 98 3 101
a=rtpmap:100 SPEEX/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 10.200.X.X 8058 typ host
a=candidate:1 2 UDP 659134 10.200.X.X 8059 typ host

<------------->
--- (13 headers 17 lines) ---
== Using SIP RTP CoS mark 5
Sending to 10.200.X.X : 8024 (NAT)
Using INVITE request as basis request - MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
Found user '200' for '200'

<--- Reliably Transmitting (no NAT) to 10.200.X.X:8024 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-186b4174337703ac-1---d8754z-;received=10.200.10.44;rport=8024
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
To: <sip:202@DOMAINE>;tag=as25f0b610
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
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="61a5b176"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP://10.200.X.X:8024 --->
ACK sip:202@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-186b4174337703ac-1---d8754z-;rport
Max-Forwards: 70
To: <sip:202@DOMAINE>;tag=as25f0b610
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8024 --->
INVITE sip:202@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-425621f5fe1674ee-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:200@10.200.X.X:8024>
To: <sip:202@DOMAINE>
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="200",realm="asterisk",nonce="61a5b176",uri="sip:202@DOMAINE",response="e0067352b88987ff8301fee4507c6be6",algorithm=MD5
Content-Length: 488

v=0
o=- 12961589563258089 1 IN IP4 10.200.X.X
s=CounterPath X-Lite 4.1
c=IN IP4 10.200.X.X
t=0 0
a=ice-ufrag:a323b6
a=ice-pwd:3d8bb9985126dad4d71a6bace385a126
m=audio 8058 RTP/AVP 100 97 105 98 3 101
a=rtpmap:100 SPEEX/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 10.200.X.X 8058 typ host
a=candidate:1 2 UDP 659134 10.200.X.X 8059 typ host

<------------->
--- (14 headers 17 lines) ---
Sending to 10.200.X.X : 8024 (no NAT)
Using INVITE request as basis request - MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
Found user '200' for '200'
Found RTP audio format 100
Found RTP audio format 97
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 3
Found RTP audio format 101
Found audio description format SPEEX for ID 100
Found audio description format SPEEX for ID 97
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x602 (gsm|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

<--- Reliably Transmitting (no NAT) to 10.200.X.X:8024 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-425621f5fe1674ee-1---d8754z-;received=10.200.X.X;rport=8024
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
To: <sip:202@DOMAINE>;tag=as25f0b610
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
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

<------------>
Scheduling destruction of SIP dialog 'MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8024 --->
ACK sip:202@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8024;branch=z9hG4bK-d8754z-425621f5fe1674ee-1---d8754z-;rport
Max-Forwards: 70
To: <sip:202@DOMAINE>;tag=as25f0b610
From: "bob"<sip:200@DOMAINE>;tag=573b1aa0
Call-ID: MDlkMmM4MDFkMzNiMzdlNzBjNjQ0NTFhYzgwNDM5ZmQ.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'NWJlYWNhOTkwYmEzZjc4MWIwZTJlNDM3NTk1MjIzZTA.' Method: ACK
SRVVOIP1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[SRVVOIP1.localdomain asterisk]#

LeRenard

27/09/2011, 12h06

APPEL 200 vers 0326XXXXXX

SRVVOIP1*CLI> sip set debug peer 200
SIP Debugging Enabled for IP: 10.200.X.X:8020
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8020 --->
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8020 --->
INVITE sip:00326XXXXXX@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-ca1bf268ec73c165-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:200@10.200.X.X:8020>
To: <sip:00326XXXXXX@DOMAINE>
From: "bob"<sip:200@DOMAINE>;tag=0394c0b8
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 370

v=0
o=- 12961590236339587 1 IN IP4 10.200.X.X
s=CounterPath X-Lite 4.1
c=IN IP4 10.200.X.X
t=0 0
a=ice-ufrag:50c323
a=ice-pwd:f1e6b24f672647dfb9ae17ad5d151de7
m=audio 8024 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 10.200.X.X 8024 typ host
a=candidate:1 2 UDP 659134 10.200.X.X 8025 typ host

<------------->
--- (13 headers 13 lines) ---
== Using SIP RTP CoS mark 5
Sending to 10.200.X.X : 8020 (NAT)
Using INVITE request as basis request - NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
Found user '200' for '200'
SRVVOIP1*CLI>
<--- Reliably Transmitting (no NAT) to 10.200.X.X:8020 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-ca1bf268ec73c165-1---d8754z-;received=10.200.10.44;rport=8020
From: "bob"<sip:200@nestal.localnet>;tag=0394c0b8
To: <sip:00326XXXXXX@DOMAINE>;tag=as02783670
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
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="1d152016"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8020 --->
ACK sip:00326XXXXXX@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-ca1bf268ec73c165-1---d8754z-;rport
Max-Forwards: 70
To: <sip:00326XXXXXX@DOMAINE>;tag=as02783670
From: "bob"<sip:200@DOMAINE>;tag=0394c0b8
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8020 --->
INVITE sip:00326XXXXXX@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-1e090f4007c8f285-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:200@10.200.X.X:8020>
To: <sip:00326XXXXXX@DOMAINE>
From: "bob"<sip:200@DOMAINE>;tag=0394c0b8
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="200",realm="asterisk",nonce="1d152016",uri="sip:00326XXXXXX@DOMAINE",response="1825e8189dceeb1edf2ca0b072fa817b",algorithm=MD5
Content-Length: 370

v=0
o=- 12961590236339587 1 IN IP4 10.200.X.X
s=CounterPath X-Lite 4.1
c=IN IP4 10.200.X.X
t=0 0
a=ice-ufrag:50c323
a=ice-pwd:f1e6b24f672647dfb9ae17ad5d151de7
m=audio 8024 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 10.200.X.X 8024 typ host
a=candidate:1 2 UDP 659134 10.200.X.X 8025 typ host

<------------->
--- (14 headers 13 lines) ---
Sending to 10.200.X.X : 8020 (no NAT)
Using INVITE request as basis request - NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
Found user '200' for '200'
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
SRVVOIP1*CLI>
<--- Reliably Transmitting (no NAT) to 10.200.X.X:8020 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-1e090f4007c8f285-1---d8754z-;received=10.200.X.X;rport=8020
From: "bob"<sip:200@DOMAINE>;tag=0394c0b8
To: <sip:00326XXXXXX@DOMAINE>;tag=as02783670
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
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

<------------>
Scheduling destruction of SIP dialog 'NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://10.200.X.X:8020 --->
ACK sip:00326XXXXXX@DOMAINE SIP/2.0
Via: SIP/2.0/UDP 10.200.X.X:8020;branch=z9hG4bK-d8754z-1e090f4007c8f285-1---d8754z-;rport
Max-Forwards: 70
To: <sip:00326XXXXXX@DOMAINE>;tag=as02783670
From: "bob"<sip:200@DOMAINE>;tag=0394c0b8
Call-ID: NmQxMWI2MTM4OTk3NzVmODUwZGQ5YjE1MjA3NTc3ZTI.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[SRVVOIP1.localdomain asterisk]#

Reaper

27/09/2011, 12h18

Il vous faut commencer lire les traces vous même et utiliser google:

Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
SRVVOIP1*CLI>
<--- Reliably Transmitting (no NAT) to 10.200.X.X:8020 --->
SIP/2.0 488 Not acceptable here

Problème de codec.

LeRenard

27/09/2011, 12h22

LeRenard

27/09/2011, 12h56

En fait, c'est un problème avec X-Lite car je viens de tester sur ZoIPer et 3CX Phone, ca marche nickel :D

Problème résolu ! :)

Merci à tous et surtout à toi Reaper :ouimaitre: