Problème dial plan asterisk communication free

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Auteur	Sujet : Problème dial plan asterisk communication free

lerenard51

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:

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

Citation :

SRVVOIP1*CLI>sip set debug ip freephonie.net
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
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]#

Fichier sip.conf

Citation :

[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
defaultexpirey=1800

[0950XXXXXX]
; 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

[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

Citation :

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

; 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@sip.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()

Message édité par lerenard51 le 15-09-2011 à 11:19:14

Publicité

lerenard51

Ajout:

J'ai ajouter defaultexpirey=1800 dans la section general de sip.conf.
Puis, reload la conf et voici le résultat de "sip show registry" dans la CLI...

Citation :

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>

black_lord

Modérateur
Truth speaks from peacefulness

[:ciler]

---------------
uptime is for lousy system administrators what Viagra is for impotent people - mes unixeries - github me

lerenard51

Personne ?! :pt1cable:

black_lord

Modérateur
Truth speaks from peacefulness

vu la tête de ton dialplan ça ne risque pas d'arriver Accessoirement masquer les IPs c'est une très mauvaise idée : ça complique le debug dans ton cas. (et en plus c'est ridicule)

---------------
uptime is for lousy system administrators what Viagra is for impotent people - mes unixeries - github me

lerenard51

C'est à dire, pourquoi tu dis ca concernant le dialplan ?
Compliqué ? Mal fait ? :??:

Euh pour les IPs, oui je sais...mais même si c'est ridicule, je préfère

black_lord

Modérateur
Truth speaks from peacefulness

mal fait, mal condensé / trop compliqué oui. Et pour les IPs c'est ton choix, mais clairement ça n'aide pas.

Message édité par black_lord le 15-09-2011 à 15:17:36

---------------
uptime is for lousy system administrators what Viagra is for impotent people - mes unixeries - github me

lerenard51

Que peux-tu me conseiller/modifier pour l'améliorer ?
Et en passant, une idée sur ma problématique ? Merci d'avance.

black_lord

Modérateur
Truth speaks from peacefulness

refactorer toutes tes actions des menus / auto*
regarder du coté de AEL, c'est puissant
pour ton souci j'ai regardé vite fait

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

c'est la clé

---------------
uptime is for lousy system administrators what Viagra is for impotent people - mes unixeries - github me

lerenard51

Ok, merci de ton aide ! Je vais appliquer tes conseils

D'après ce que tu me dis, le problème viendrais du fichier extensions.conf
et plus particulièrement sur cette partie:

Citation :

; 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

Donc, peux-tu m'en dire plus concernant:

Citation :

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

Merci d'avance !

Publicité

lerenard51

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:

Citation :

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.

Message édité par lerenard51 le 20-09-2011 à 16:05:58

lerenard51

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:

Citation :

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>

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