Asterisk asterisk appel: 403 Forbidden
J'ai 2 serveurs avec des Astérisques: 192.168.241.98 et 192.168.243.112.
Il y a un enregistrement valable sur la première:
register => wagateway:[email protected]:5060
De la CLI de sortie:
CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
192.168.243.112:5060 N wagateway 105 Registered Wed, 26 Jun 2013 16:42:42
Et des pairs sur 243.112 sont juste très bien:
CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status Description
wacaller/wacaller 192.168.242.235 D a 5062 OK (13 ms)
wagateway/s 192.168.241.98 D a 5060 OK (1 ms)
extensions.conf sur 243.112:
[watest]
exten => 123123123,1,NoOp()
exten => 123123123,n,Dial(SIP/wagateway)
exten => 123123123,n,Hangup()
sip.conf sur 243.112:
[wacaller]
type=friend
secret=qwerty
host=dynamic
context=watest
qualify=yes
allow=ulaw
allow=alaw
[wagateway]
type=friend
secret=qwerty
fromuser=wagateway
host=dynamic
context=watest
qualify=yes
allow=ulaw
allow=alaw
Maintenant j'essaie d'appeler 123123123 de wacaller de Grandstream téléphone.
243.112 CLI dit:
[Jun 27 09:27:54] WARNING[20447][C-0000000b]: chan_sip.c:23213 handle_response_invite: Received response: "Forbidden" from '"WACaller" <sip:[email protected]>;tag=as30b27eae'
Sip debug sur 243.112:
<--- SIP read from UDP:192.168.242.235:5062 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;rport
From: "WACaller" <sip:[email protected]>;tag=1014197566
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 240 INVITE
Contact: "WACaller" <sip:[email protected]:5062>
Max-Forwards: 70
User-Agent: Grandstream GXP1400 1.0.4.13
Privacy: none
P-Preferred-Identity: "WACaller" <sip:[email protected]>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 412
v=0
o=wacaller 8000 8000 IN IP4 192.168.242.235
s=SIP Call
c=IN IP4 192.168.242.235
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (16 headers 19 lines) ---
Sending to 192.168.242.235:5062 (no NAT)
Sending to 192.168.242.235:5062 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer 'wacaller' for 'wacaller' from 192.168.242.235:5062
<--- Reliably Transmitting (no NAT) to 192.168.242.235:5062 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;received=192.168.242.235;rport=5062
From: "WACaller" <sip:[email protected]>;tag=1014197566
To: <sip:[email protected]>;tag=as5a3de236
Call-ID: [email protected]
CSeq: 240 INVITE
Server: Asterisk PBX SVN-trunk-r385782
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f84bef0"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.242.235:5062 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;rport
From: "WACaller" <sip:[email protected]>;tag=1014197566
To: <sip:[email protected]>;tag=as5a3de236
Call-ID: [email protected]
CSeq: 240 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:192.168.242.235:5062 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK1881861609;rport
From: "WACaller" <sip:[email protected]>;tag=1014197566
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 241 INVITE
Contact: "WACaller" <sip:[email protected]:5062>
Authorization: Digest username="wacaller", realm="asterisk", nonce="4f84bef0", uri="sip:[email protected]", response="53cdb5b8c1822c80870faab15a6dc6d2", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXP1400 1.0.4.13
Privacy: none
P-Preferred-Identity: "WACaller" <sip:[email protected]>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 412
v=0
o=wacaller 8000 8000 IN IP4 192.168.242.235
s=SIP Call
c=IN IP4 192.168.242.235
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 19 lines) ---
Sending to 192.168.242.235:5062 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer 'wacaller' for 'wacaller' from 192.168.242.235:5062
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 97
Found RTP audio format 2
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 G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 97
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.242.235:5004
Looking for 123123123 in watest (domain 192.168.243.112)
list_route: route/path hop: <sip:[email protected]:5062>
<--- Transmitting (no NAT) to 192.168.242.235:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK1881861609;received=192.168.242.235;rport=5062
From: "WACaller" <sip:[email protected]>;tag=1014197566
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 241 INVITE
Server: Asterisk PBX SVN-trunk-r385782
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
Audio is at 17372
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.241.98:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;rport
Max-Forwards: 70
From: "WACaller" <sip:[email protected]>;tag=as3f5f372a
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Date: Wed, 26 Jun 2013 08:31:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326
v=0
o=root 2059284449 2059284449 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 17372 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 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:192.168.241.98:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;received=192.168.243.112;rport=5060
From: "WACaller" <sip:[email protected]>;tag=as3f5f372a
To: <sip:[email protected]:5060>;tag=as22eeeac0
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="603b4bbf"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 192.168.241.98:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;rport
Max-Forwards: 70
From: "WACaller" <sip:[email protected]>;tag=as3f5f372a
To: <sip:[email protected]:5060>;tag=as22eeeac0
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0
---
Audio is at 17372
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.241.98:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;rport
Max-Forwards: 70
From: "WACaller" <sip:[email protected]>;tag=as3f5f372a
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Authorization: Digest username="s", realm="asterisk", algorithm=MD5, uri="sip:[email protected]:5060", nonce="603b4bbf", response="059cae207fb81fb76ea9061f71258895"
Date: Wed, 26 Jun 2013 08:31:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326
v=0
o=root 2059284449 2059284450 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 17372 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 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:192.168.241.98:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;received=192.168.243.112;rport=5060
From: "WACaller" <sip:[email protected]>;tag=as3f5f372a
To: <sip:[email protected]:5060>;tag=as22eeeac0
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.241.98:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;rport
Max-Forwards: 70
From: "WACaller" <sip:[email protected]>;tag=as3f5f372a
To: <sip:[email protected]:5060>;tag=as22eeeac0
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0
---
[Jun 26 16:31:48] WARNING[20447][C-0000000a]: chan_sip.c:23213 handle_response_invite: Received response: "Forbidden" from '"WACaller" <sip:[email protected]>;tag=as3f5f372a'
Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
Sip debug sur le serveur de destination:
<--- SIP read from UDP:192.168.243.112:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;rport
Max-Forwards: 70
From: "WACaller" <sip:[email protected]>;tag=as30b27eae
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Date: Thu, 27 Jun 2013 01:27:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326
v=0
o=root 1301894386 1301894386 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 15838 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 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 14 lines) ---
Sending to 192.168.243.112:5060 (NAT)
Using INVITE request as basis request - [email protected]:5060
Found peer 'wagateway' for 'wagateway' from 192.168.243.112:5060
<--- Reliably Transmitting (no NAT) to 192.168.243.112:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;received=192.168.243.112;rport=5060
From: "WACaller" <sip:[email protected]>;tag=as30b27eae
To: <sip:[email protected]:5060>;tag=as671c0824
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b63a660"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.243.112:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;rport
Max-Forwards: 70
From: "WACaller" <sip:[email protected]>;tag=as30b27eae
To: <sip:[email protected]:5060>;tag=as671c0824
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:192.168.243.112:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;rport
Max-Forwards: 70
From: "WACaller" <sip:[email protected]>;tag=as30b27eae
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Authorization: Digest username="s", realm="asterisk", algorithm=MD5, uri="sip:[email protected]:5060", nonce="0b63a660", response="537f37fe2fb8d0fd40733cb190ea70c8"
Date: Thu, 27 Jun 2013 01:27:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326
v=0
o=root 1301894386 1301894387 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 15838 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 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 14 lines) ---
Sending to 192.168.243.112:5060 (no NAT)
Using INVITE request as basis request - [email protected]:5060
Found peer 'wagateway' for 'wagateway' from 192.168.243.112:5060
<--- Reliably Transmitting (no NAT) to 192.168.243.112:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;received=192.168.243.112;rport=5060
From: "WACaller" <sip:[email protected]>;tag=as30b27eae
To: <sip:[email protected]:5060>;tag=as671c0824
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.243.112:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;rport
Max-Forwards: 70
From: "WACaller" <sip:[email protected]>;tag=as30b27eae
To: <sip:[email protected]:5060>;tag=as671c0824
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
dev-ast*CLI> sip set debug off
SIP Debugging Disabled
Toute aide?
OriginalL'auteur Ilya Khaustov | 2013-06-27
Vous devez vous connecter pour publier un commentaire.
Un autre problème que vous avez est une boucle, vous pouvez envoyer l'appel vers votre passerelle, et lors de l'appel de venir à votre passerelle de vous envoyer de nouveau à la passerelle, c'est le pourquoi êtes-vous un interdit, lorsque vous composez le SIP/wagateway (sur wagateway) vous n'avez pas les extensions, votre appeler du client ---> passerelle ---> passerelle , essayez de vous changer l'extension de watest à quelque chose comme ci-dessous
OriginalL'auteur Mike Tesliuk
Comparer à l'un de mes Astérisque-à-Asterisk SIP trunks...
Il ressemble à ce que j'utilise est le
defaultuser=paramètre dans monsip.confpar opposition àfromuser=À partir de l'original
sip.confqui vient avec unemake samples--defaultuserest décrit comme "l'Authentification de l'utilisateur pour le proxy sortant". Alors que ce n'est pas un proxy dans ce cas, je crois que c'est le paramètre qui sera utilisé lors de la prise de cette demande de SIP.Cela étant dit, vous pouvez également envisager l'utilisation de la
iaxprotocole lorsque vous avez la commodité de la mise en place d'un tronc entre deux serveurs asterisk. Les normes de "l'Inter-Asterisk eXchange", et je trouve que c'est plus simple à utiliser. Et surtout plus simple ne semble pas souffrir des mêmes maux que le SIP n'lors de la traversée de NAT.Voici un exemple d'un trunk SIP-je entre deux astérisque boîtes.
La Case A, "New York":
Et sur la Boîte B, "Tokyo":
Notez comment le
defaultusersur la Boîte d'Une configuration de parler à tokyo (aka la Case B) correspond au nom du périphérique[newyork]sur la Case B dusip.conf[tokyo]. À Tokyo, je fais appel à des pairs[newyork]. Ce réseau est inscrit à partir de New York comme une passerelle. Appel va à New York serveur et là, il devrait aller en entrant contexte, spécialement conçu pour de tels cas. Il n'a pas. L'appel s'arrête quelque part entre les deux. Aprèswagatewayenvoie INVITER àsextension à New York, il se passe quelque chose et je ne sais pas quoi.De ma faute. Il est
[tokyo], et c'est important. Après l'ajout deport=invite,insecuredans sa config tout a fonctionné comme prévu.avez-vous ajouter
insecure=invite,portouport=invite,insecure?Oh,
insecure=invite,portbien sûr, désolé.OriginalL'auteur dougBTV
Avez-vous essayé avec:
Vous envoyez appel sur s extension dans
[watest]contexte (qui est par défaut si vous ne spécifiez pas l'extension), mais s n'existe pas, seulement 123123123.edit1:
Ok que d'ajouter de modifier
[wacaller]ajouter:laissez-moi savoir si cela a fonctionné, merci.
edit2:
essayez de supprimer/commentaire
Vérifier la Grandstream forum, il est le plus susceptible de téléphone de problème.
edit3:
Problème de 99% réside dans le fait que vous vous inscrivez à un serveur (192.168.243.112) et des invitations sont envoyées à wagateway/s(192.168.241.98) autre serveur ou IP
La chaîne de registre n'est pas le même que celui de l'inviter, et là vous obtenez interdit message.
cela devrait vous aider:
;insecure=inviter,port
sur la passerelle de l'appelant tronc, si vous voulez garder cette configuration du réseau.
Ce qui concerne
/${EXTEN}ne fonctionne pas de toute façon.ok je ajouter de la modifier après avoir vu mieux
Ne fonctionne pas, désolé.
ajouter
nat=yestropnat=yesn'a pas aidé.OriginalL'auteur mirkobrankovic