Voir la version complète : Call from 'client-3' to extension '5' rejected because extension not found in context 'users'.
sunsetlive
02/07/2013, 16h03
Bonjour à tous !
Dans le cadre d'un projet, je dois monter une maquette en local avec des clients X-Lite (client-1 & client-2) et un téléphone CISCO 7942 (client-3). Le problème c'est que la communication s'effectue très bien entre les clients X-LITE mais du téléphone CISCO, je
peux uniquement recevoir des appels mais impossible d'émettre un appel depuis le téléphone.
Dans les logs Asterisk, j'obtiens ceci:
NOTICE[10163]: chan_sip.c:22718 handle_request_invite: Call from 'client-3' (192.168.1.158:49769) to extension '5' rejected because extension not found in context 'users'.
lol*CLI>
fichier SIP.conf
[client-1] ; nom du téléphone
type=friend ; type de téléphone
host=dynamic
context=users
permit=192.168.1.254 ; network setting
secret=serveur
callerid="client-1" <555> ; association user et num de tel
mailbox=client-1@192.168.1.254
[client-2]
type=friend ; type de téléphone
host=dynamic
context=users
permit=192.168.1.254 ; network setting
username=client-2 ; nom d'utilisateur associé
secret=serveur ; mot de passe
callerid="client-2" <556> ; association user et num de tel
mailbox=client-2@192.168.1.254 ;Adresse de la boite vocale et dans
;notre cas remplacer nomdomaine par
;l’adresse ip de serveur asterisk
[client-3] ; nom du téléphone
type=friend ; type de téléphone
host=dynamic ; enregistrement dynamique de
context=users
permit=192.168.1.254 ; network setting
qualify = no
nat = no
allowguest = no
username=client-3 ; nom d'utilisateur associé
secret=serveur ; mot de passe
callerid="client-3" <557> ; association user et num de tel
mailbox=client-3@192.168.1.254 ;Adresse de la boite vocale et dans
extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[users]
exten => 555,1,Dial(SIP/client-1,20)
exten => 556,1,Dial(SIP/client-2,20)
exten => 557,1,Dial(SIP/client-3,20)
MERCI d'avance ! pour ceux qui m'aide !!
tu numérotes "5" et cela n'existe pas dans ton contexte users - ce qui est effectivement le cas
Essaie de numeroter 556 ou 555...
sunsetlive
02/07/2013, 17h03
Merci Jean d'avoir répondu !
Non effectivement j'ai peut-être omis de dire que j'ai effectué plusieurs test !! Notamment la numérotation "555" et "556" pour joindre les client 1 & 2. Mais rien à faire lorsque je numérote que ce soit un nombre à 3 chiffres "435" ou 10 chiffres "2535454531" dans les logs de Asterisk il sort toujours "... to extension '5' ... alors qu'il devrait correspondre à la numérotation effectué sur l'IPphone. Par exemple si je compose '555' sur l'IPphone, il devrait me sortir en log:
Call from 'client-3' to extension '555' rejected because extension not found in context 'users'.
Hors C'est pas le cas !!! Il sort toujours une erreur de type Call from 'client-3' to extension '5' rejected because extension not found in context 'users' --> Un seul chiffre alors que j'en est composé 3 !!
Je sais pas si je suis claire dans mes propos mais voilà le souci ...
Merci encore d'avoir répondu !!
au passage: c'est excellent d'avoir mis allowguest=no, mais il vaut mieux le mettre dans la section [generral]
fais un core set verbose 9
puis sip set debug on
et copie le résultat
J
sunsetlive
02/07/2013, 17h49
J'ai fais comme tu m'a dis, j'ai mis allowguest = no dans general
puis tapez les commandes ...
J'espère que c'est les logs attendu ...
<--- Reliably Transmitting (no NAT) to 192.168.1.158:5061 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKc2be2d2a;received =192.168.1.158
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a5620017ccbc8736-cc8044ca
To: <sip:5@192.168.1.254>;tag=as31595924
Call-ID: a40cc394-a5620011-2b015b06-6de2c5da@192.168.1.158
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[Jul 2 17:44:40] NOTICE[12769]: chan_sip.c:22718 handle_request_invite: Call from 'client-3' (192.168.1.158:52726) to extension '5' rejected because extension not found in context 'users'.
Scheduling destruction of SIP dialog 'a40cc394-a5620011-2b015b06-6de2c5da@192.168.1.158' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.1.158:51110 --->
ACK sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKc2be2d2a
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a5620017ccbc8736-cc8044ca
To: <sip:5@192.168.1.254>;tag=as31595924
Call-ID: a40cc394-a5620011-2b015b06-6de2c5da@192.168.1.158
Max-Forwards: 70
Date: Tue, 02 Jul 2013 15:44:37 GMT
CSeq: 102 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'a40cc394-a5620011-2b015b06-6de2c5da@192.168.1.158' Method: ACK
il manque des infos - fais d'abord le sip set debug on, puis lance l'appel
sunsetlive
03/07/2013, 09h00
J'ai effectué la commande sip set debug on puis l'appel du client-3 mais j'obtiens toujours ce genre de logs ...
<--- SIP read from UDP:192.168.1.158:50296 --->
BYE sip:555@192.168.1.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK54886b8e
From: <sip:557@192.168.1.158:5061;transport=udp>;tag=a40cc394a562001c21929f90-1537ed53
To: "client-1" <sip:555@192.168.1.254>;tag=as6c2b921b
Call-ID: 2db2795f36d6bb4733362aad3d972fe6@192.168.1.254:506 0
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:08 GMT
CSeq: 101 BYE
User-Agent: Cisco-CP7942G/9.3.1
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.158:5061 (no NAT)
Scheduling destruction of SIP dialog '2db2795f36d6bb4733362aad3d972fe6@192.168.1.254:50 60' in 32000 ms (Method: BYE)
<--- Transmitting (no NAT) to 192.168.1.158:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK54886b8e;received =192.168.1.158
From: <sip:557@192.168.1.158:5061;transport=udp>;tag=a40cc394a562001c21929f90-1537ed53
To: "client-1" <sip:555@192.168.1.254>;tag=as6c2b921b
Call-ID: 2db2795f36d6bb4733362aad3d972fe6@192.168.1.254:506 0
CSeq: 101 BYE
Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
set_destination: Parsing <sip:client-1@192.168.1.150:5060> for address/port to send to
set_destination: set destination to 192.168.1.150:5060
Audio is at 11478
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.150:5060:
INVITE sip:client-1@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK0bfe3f18;rport
Max-Forwards: 70
From: <sip:557@192.168.1.254:5060>;tag=as16824010
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
Contact: <sip:557@192.168.1.254:5060>
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 781600293 781600295 IN IP4 192.168.1.254
s=Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
c=IN IP4 192.168.1.254
t=0 0
m=audio 11478 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:192.168.1.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK0bfe3f18;rport=50 60
Contact: <sip:client-1@192.168.1.150:5060>
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
From: <sip:557@192.168.1.254:5060>;tag=as16824010
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.0.0 stamp 67284
Content-Length: 213
v=0
o=- 13014543165961468 3 IN IP4 192.168.1.150
s=CounterPath X-Lite 5.0.0
c=IN IP4 192.168.1.150
b=AS:1638
t=0 0
m=audio 5062 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/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 192.168.1.150:5062
sunsetlive
03/07/2013, 09h01
suite des logs
set_destination: Parsing <sip:client-1@192.168.1.150:5060> for address/port to send to
set_destination: set destination to 192.168.1.150:5060
Transmitting (NAT) to 192.168.1.150:5060:
ACK sip:client-1@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK767cf41d;rport
Max-Forwards: 70
From: <sip:557@192.168.1.254:5060>;tag=as16824010
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
Contact: <sip:557@192.168.1.254:5060>
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Content-Length: 0
---
== Spawn extension (users, 557, 1) exited non-zero on 'SIP/client-1-00000000'
Scheduling destruction of SIP dialog 'YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:client-1@192.168.1.150:5060> for address/port to send to
set_destination: set destination to 192.168.1.150:5060
Reliably Transmitting (NAT) to 192.168.1.150:5060:
BYE sip:client-1@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK42bf98f8;rport
Max-Forwards: 70
From: <sip:557@192.168.1.254:5060>;tag=as16824010
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Proxy-Authorization: Digest username="client-1", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.254", nonce="", response="323d061907b0821b1decc513156e9f67"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:192.168.1.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK42bf98f8;rport=50 60
Contact: <sip:client-1@192.168.1.150:5060>
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
From: <sip:557@192.168.1.254:5060>;tag=as16824010
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 104 BYE
User-Agent: X-Lite release 5.0.0 stamp 67284
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.' Method: ACK
<--- SIP read from UDP:192.168.1.150:5060 --->
<------------->
<--- SIP read from UDP:192.168.1.158:50296 --->
INVITE sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK386bb6e8
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:20 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7942G/9.3.1
Contact: <sip:557@192.168.1.158:5061;transport=udp>
Expires: 180
Accept: application/sdp
llow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTE R,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 354
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 12876 0 IN IP4 192.168.1.158
s=SIP Call
t=0 0
m=audio 23956 RTP/AVP 0 8 18 102 116 101
c=IN IP4 192.168.1.158
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (18 headers 16 lines) ---
Sending to 192.168.1.158:50296 (NAT)
Using INVITE request as basis request - a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Found peer 'client-3' for 'client-3' from 192.168.1.158:50296
<--- Reliably Transmitting (no NAT) to 192.168.1.158:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK386bb6e8;received =192.168.1.158
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>;tag=as7134625c
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02702734"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.1.158:53122 --->
ACK sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK386bb6e8
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>;tag=as7134625c
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:20 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:192.168.1.158:50296 --->
INVITE sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKfc3dd1e1
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:20 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7942G/9.3.1
Contact: <sip:557@192.168.1.158:5061;transport=udp>
Authorization: Digest username="client-3",realm="asterisk",uri="sip:5@192.168.1.254;user=phone",response="61db6f5bf1b4ac464547bc3903ead800",nonce="02702734",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTE R,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 354
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 12876 0 IN IP4 192.168.1.158
s=SIP Call
t=0 0
m=audio 23956 RTP/AVP 0 8 18 102 116 101
c=IN IP4 192.168.1.158
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (19 headers 16 lines) ---
Sending to 192.168.1.158:5061 (no NAT)
Using INVITE request as basis request - a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Found peer 'client-3' for 'client-3' from 192.168.1.158:50296
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 102
Found RTP audio format 116
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 G729 for ID 18
Found audio description format L16 for ID 102
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x850c (ulaw|alaw|g729|ilbc|slin16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
sunsetlive
03/07/2013, 09h02
suite des logs
Peer audio RTP is at port 192.168.1.158:23956
Looking for 5 in users (domain 192.168.1.254)
<--- Reliably Transmitting (no NAT) to 192.168.1.158:5061 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKfc3dd1e1;received =192.168.1.158
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>;tag=as7134625c
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[Jul 3 08:52:31] NOTICE[14513]: chan_sip.c:22718 handle_request_invite: Call from 'client-3' (192.168.1.158:50296) to extension '5' rejected because extension not found in context 'users'.
Scheduling destruction of SIP dialog 'a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.1.158:52349 --->
ACK sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKfc3dd1e1
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>;tag=as7134625c
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:20 GMT
CSeq: 102 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'OTlkMGE1ODgyM2FjNzYyNWIzMTFkN2QzOGE3ZjIyZTU.' Method: REGISTER
Really destroying SIP dialog '2db2795f36d6bb4733362aad3d972fe6@192.168.1.254:50 60' Method: BYE
<--- SIP read from UDP:192.168.1.150:5060 --->
<------------->
Really destroying SIP dialog 'a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158' Method: ACK
<--- SIP read from UDP:192.168.1.150:5060 --->
<------------->
<--- SIP read from UDP:192.168.1.150:5060 --->
<------------->
<--- SIP read from UDP:192.168.1.150:5060 --->
<------------->
<--- SIP read from UDP:192.168.1.150:5060 --->
<------------->
lol*CLI>
on voit :
INVITE sip:client-1@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK0bfe3f18;rport
Max-Forwards: 70
From: <sip:557@192.168.1.254:5060>;tag=as16824010
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
et
INVITE sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK386bb6e8
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>
le permier est un appel du poste 557 vers "client-1", et le second (répété dans les logs) est un appel de 557 vers "5"
Asterisk se comporte donc logiquement.
Quel client utilise tu ? Vérifie la config....
sunsetlive
04/07/2013, 09h22
Ok ! merci pour tes remarques !
J'utilise 2 clients virtuels X-Lites (client-1 et client-2) et un téléphone CISCO 7942 (client-3).
En ce qui concerne ta remarque, c'est intéressant car je cherche à joindre depuis le poste "557" --> (client-3) vers le poste "555"--> (client-1), le souci c'est que j'ai effectué un seul et unique appel ! D’où il sort l'appel du "557" vers le "5" ?! je compose sur le tel (depuis le poste "557") "555" mais jamais "5" tout seul !? vers le client-1 !
Mon problème, c'est que les clients X-lites fonctionnent parfaitement entre eux, mais avec l'IPphone Cisco, je ne peux que recevoir mais pas émettre d'appel vers l’extérieur ...
Je vais revoir la config ...
je connais pas le cisco iphone -il y a peut être un paramètre à regler (genre overlap dialing). il faut peut être relancer un autre sujet avec ca dans le titre
si vraiment tu veux voir simplement le trafic qui passe vers ton asterisk, je te recommande l'utilitaire unix ngrep, (soit via apt ou un google rpm ngrep tonos), genre ngrep port 5060 and host 1.2.3.4 - c'est très lisible et tu vois vite ce qui se passe
sunsetlive
04/07/2013, 16h25
OK ! Je vais télécharger ce que tu m'a dis ! En tout cas c'est vraiment cool ! pour tes remarques et suggestions, je te remercie, je continue à poster si j'ai du nouveau !
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