[RESOLU] Problème dial plan asterisk ligne free

**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 !

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=247 76-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=247 76-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=247 76-CU-104cd6ed-352a20ba5
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as6eb4b7 c8
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 !!

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

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/conte...-Les-contextes

Une dernière chose,

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

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

**LeRenard** · 22/09/2011, 14h25

Envoyé par Reaper

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/conte...-Les-contextes

Ok, je vais voir ca...

Envoyé par Reaper

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.

Envoyé par Reaper

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 !!
Tu m'as vraiment beaucoup aider et éclaircie sur certains points

**benjborhen** · 22/09/2011, 16h28

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 .

**LeRenard** · 22/09/2011, 16h40

Envoyé par benjborhen

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

Reaper m'a dis de regrouper le tout

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 ?!

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 !

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

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",u ri="sip:202@DOMAINE",response="e0067352b88987ff830 1fee4507c6be6",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",u ri="sip:00326XXXXXX@DOMAINE",response="1825e8189dc eeb1edf2ca0b072fa817b",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.

Discussion: [RESOLU] Problème dial plan asterisk ligne free

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