Freepbx ошибка 403

    -- Executing [s@macro-dialout-trunk:22] Dial("SIP/10004-00000183", "SIP/office/90031,300,Ttr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 18016
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to officeip:5060:
INVITE sip:90031@officeip SIP/2.0
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK5f165e0b;rport
Max-Forwards: 70
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>
Contact: <sip:10004@filialip:5060>
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(1.8.13.1)
Date: Tue, 03 Jun 2014 07:21:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 194525798 194525798 IN IP4 filialip
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 filialip
t=0 0
m=audio 18016 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=ptime:20
a=sendrecv

---
    -- Called SIP/office/90031

<--- SIP read from UDP:officeip:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK5f165e0b;received=filialip;rport=5060
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>;tag=as79445bbf
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.6.2.9)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c2ae67b"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:90031@officeip> for address/port to send to
set_destination: set destination to officeip:5060
Transmitting (NAT) to officeip:5060:
ACK sip:90031@officeip SIP/2.0
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK5f165e0b;rport
Max-Forwards: 70
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>;tag=as79445bbf
Contact: <sip:10004@filialip:5060>
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(1.8.13.1)
Content-Length: 0

---
Audio is at 18016
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to officeip:5060:
INVITE sip:90031@officeip SIP/2.0
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK7de64492;rport
Max-Forwards: 70
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>
Contact: <sip:10004@filialip:5060>
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(1.8.13.1)
Authorization: Digest username="office", realm="asterisk", algorithm=MD5, uri="sip:90031@officeip", nonce="4c2ae67b", response="df103ca6ce7459aa4f1dc34f935cee99"
Date: Tue, 03 Jun 2014 07:21:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 194525798 194525799 IN IP4 filialip
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 filialip
t=0 0
m=audio 18016 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=ptime:20
a=sendrecv

---

<--- SIP read from UDP:officeip:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK7de64492;received=filialip;rport=5060
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>;tag=as79445bbf
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 103 INVITE
Server: FPBX-2.9.0(1.6.2.9)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:90031@officeip> for address/port to send to
set_destination: set destination to officeip:5060
Transmitting (NAT) to officeip:5060:
ACK sip:90031@officeip SIP/2.0
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK7de64492;rport
Max-Forwards: 70
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>;tag=as79445bbf
Contact: <sip:10004@filialip:5060>
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(1.8.13.1)
Content-Length: 0

---
[2014-06-03 14:21:29] WARNING[26892]: chan_sip.c:20366 handle_response_invite: Received response: "Forbidden" from '"Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2'
Scheduling destruction of SIP dialog '6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060' in 32000 ms (Method: INVITE)

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sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | configuration parameters after validation:
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · viaHost: «192.0.2.138»
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · uri: sip:1060@192.168.0.110
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · wsServers: [{«ws_uri»:»ws://192.168.0.110:8088/ws»,»sip_uri»:»sip:192.168.0.110:8088;transport=ws;lr»,»weight»:0,»status»:0,»scheme»:»WS»}]
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · password: NOT SHOWN
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · registerExpires: 600
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · register: true
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · registrarServer: sip:192.168.0.110
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · wsServerMaxReconnection: 3
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · wsServerReconnectionTimeout: 4
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · connectionRecoveryMinInterval: 2
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · connectionRecoveryMaxInterval: 30
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · usePreloadedRoute: false
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · userAgentString: «SIP.js/0.6.3»
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · noAnswerTimeout: 60000
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · stunServers: [«stun:stun.l.google.com:19302»]
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · turnServers: []
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · traceSip: true
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · hackViaTcp: false
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · hackIpInContact: true
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · autostart: true
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · rel100: «none»
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · mediaHandlerFactory: function (e,n){return new f(e,n)}
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · authorizationUser: «1060»
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · instanceId: «f9fb2ac6-5a01-47ad-ad7d-7ca02d0c0ff1»
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · sipjsId: «k82uf»
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · hostportParams: «192.168.0.110»
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | · media: undefined
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event connecting
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event connected
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event disconnected
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event newTransaction
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event transactionDestroyed
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event registered
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event unregistered
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event registrationFailed
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event invite
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event newSession
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event message
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event ack
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event cancel
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event bye
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event options
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event info
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | adding event notify
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.clientcontext | adding event progress
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.clientcontext | adding event accepted
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.clientcontext | adding event rejected
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.clientcontext | adding event failed
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.clientcontext | adding event cancel
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.registercontext | adding event registered
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.registercontext | adding event unregistered
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.registercontext | new listener added to event failed
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.registercontext | new listener added to event registered
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.registercontext | new listener added to event unregistered
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | user requested startup…
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transport | connecting to WebSocket ws://192.168.0.110:8088/ws
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | emitting event connecting
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transport | WebSocket ws://192.168.0.110:8088/ws connected
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | connection state set to 0
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | emitting event newTransaction
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transaction.nict | adding event stateChanged
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transaction.nict | emitting event stateChanged
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transport | sending WebSocket message:

REGISTER sip:192.168.0.110 SIP/2.0
Via: SIP/2.0/WS 192.0.2.138;branch=z9hG4bK6100599
Max-Forwards: 70
To: sip:1060@192.168.0.110
From: sip:1060@192.168.0.110;tag=7mm6osi0qe
Call-ID: cgfg4cb730jjif8p9mpdcl
CSeq: 81 REGISTER
Contact: sip:lea4u4e8@192.0.2.138;transport=ws;reg-id=1;+sip.instance=»urn:uuid:f9fb2ac6-5a01-47ad-ad7d-7ca02d0c0ff1″;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: SIP.js/0.6.3
Content-Length: 0

sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | emitting event connected
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS 192.0.2.138;branch=z9hG4bK6100599;received=192.168.0.102
From: sip:1060@192.168.0.110;tag=7mm6osi0qe
To: sip:1060@192.168.0.110;tag=as5928b881
Call-ID: cgfg4cb730jjif8p9mpdcl
CSeq: 81 REGISTER
Server: Asterisk PBX 11.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=»asterisk», nonce=»674b43f6″
Content-Length: 0

sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transaction.nict | emitting event stateChanged
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | emitting event newTransaction
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transaction.nict | adding event stateChanged
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transaction.nict | emitting event stateChanged
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transport | sending WebSocket message:

REGISTER sip:192.168.0.110 SIP/2.0
Via: SIP/2.0/WS 192.0.2.138;branch=z9hG4bK3318989
Max-Forwards: 70
To: sip:1060@192.168.0.110
From: sip:1060@192.168.0.110;tag=7mm6osi0qe
Call-ID: cgfg4cb730jjif8p9mpdcl
CSeq: 82 REGISTER
Authorization: Digest algorithm=MD5, username=»1060″, realm=»asterisk», nonce=»674b43f6″, uri=»sip:192.168.0.110″, response=»f716951f56eae021675ed20752a9e5ec»
Contact: sip:lea4u4e8@192.0.2.138;transport=ws;reg-id=1;+sip.instance=»urn:uuid:f9fb2ac6-5a01-47ad-ad7d-7ca02d0c0ff1″;expires=600
Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: SIP.js/0.6.3
Content-Length: 0

sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transport | received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WS 192.0.2.138;branch=z9hG4bK3318989;received=192.168.0.102
From: sip:1060@192.168.0.110;tag=7mm6osi0qe
To: sip:1060@192.168.0.110;tag=as5928b881
Call-ID: cgfg4cb730jjif8p9mpdcl
CSeq: 82 REGISTER
Server: Asterisk PBX 11.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: sip:lea4u4e8@192.0.2.138;transport=ws;expires=600
Date: Thu, 27 Nov 2014 14:23:58 GMT
Content-Length: 0

sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transaction.nict | emitting event stateChanged
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.registercontext | emitting event accepted
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.registercontext | emitting event registered
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | emitting event registered
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transaction.nict | emitting event stateChanged
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | emitting event transactionDestroyed
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.transaction.nict | emitting event stateChanged
sip-0.6.3.min.js:36 Thu Nov 27 2014 22:23:58 GMT+0800 (Malay Peninsula Standard Time) | sip.ua | emitting event transactionDestroyed

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Откуда: 123123. Москва, Какоенибудьтам шоссе, д. 111.

Сообщений: 8

SIP/2.0 403 Forbidden

Не работает исходящий SIP звонок. Не знаю куда копать.

trixbox1*CLI> sip show peers

Name/username Host Dyn Nat ACL Port Status

internetcalls/skynet66 77.72.169.129 5060 Unmonitored

gts-sip/5620640 89.232.125.48 N 5060 Unmonitored

105 (Unspecified) D N A 5060 UNKNOWN

104 (Unspecified) D N A 5060 UNKNOWN

103/103 192.168.101.154 D N A 5061 OK (16 ms)

102/102 192.168.101.154 D N A 5060 OK (16 ms)

101/101 192.168.101.153 D N A 5072 OK (15 ms)

100/100 192.168.101.153 D N A 5071 OK (15 ms)

8 sip peers [Monitored: 4 online, 2 offline Unmonitored: 2 online, 0 offline]

trixbox1*CLI> sip show registry

Host Username Refresh State Reg.Time

89.xxx.125.48:5060 5620640 101 Registered Fri, 19 Mar 2010 12:10:51

1 SIP registrations.

Really destroying SIP dialog ’11ce763328da47402307a074021634c5@127.0.0.1′ Method: REGISTER

PEER Details:

host=89.232.125.48

insecure=port,invite

nat=yes

fromuser=5620640

username=5620640

secret=xxxxxxxx

type=peer

disallow=all

allow=ulaw&g711a&g711

а это кусочек лога с SIP/2.0 403 от SIP прокси:

— Executing [s@macro-dialout-trunk:18] GotoIf(«SIP/103-08d2bf20», «0?customtrunk») in new stack

— Executing [s@macro-dialout-trunk:19] Dial(«SIP/103-08d2bf20», «SIP/gts-sip/2909294,300,») in new stack

== Using SIP RTP TOS bits 184

== Using SIP RTP CoS mark 5

== Using SIP VRTP TOS bits 136

== Using SIP VRTP CoS mark 6

Audio is at 192.168.101.180 port 16400

Adding codec 0x4 (ulaw) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (NAT) to 89.xxx.125.48:5060:

INVITE sip:2909294@89.xxx.125.48 SIP/2.0

Via: SIP/2.0/UDP 192.168.101.180:5060;branch=z9hG4bK0c37c7a5;rport

Max-Forwards: 70

From: «103» <sip:5620640@192.168.101.180>;tag=as4d695e92

To: <sip:2909294@89.xxx.125.48>

Contact: <sip:5620640@192.168.101.180>

Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180

CSeq: 102 INVITE

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

ate: Fri, 19 Mar 2010 09:02:49 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 278

v=0

o=root 1153224551 1153224551 IN IP4 192.168.101.180

s=Asterisk PBX 1.6.0.9-samy-r27

c=IN IP4 192.168.101.180

t=0 0

m=audio 16400 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off — — — —

a=ptime:20

a=sendrecv

—

— Called gts-sip/2909294

trixbox1*CLI>

<— SIP read from UDP://89.xxx.125.48:5060 —>

SIP/2.0 100 Trying

From: «103»<sip:5620640@192.168.101.180>;tag=as4d695e92

To: <sip:2909294@89.xxx.125.48>

Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180

CSeq: 102 INVITE

Via: SIP/2.0/UDP 192.168.101.180:5060;received=78.138.144.127;rport=5060;branch=z9hG4bK0c37c7a5

Content-Length: 0

<————->

— (7 headers 0 lines) —

trixbox1*CLI>

<— SIP read from UDP://89.xxx.125.48:5060 —>

SIP/2.0 403 Forbidden

From: «103»<sip:5620640@192.168.101.180>;tag=as4d695e92

To: <sip:2909294@89.xxx.125.48>;tag=2083351771

Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180

CSeq: 102 INVITE

Via: SIP/2.0/UDP 192.168.101.180:5060;received=78.138.144.127;rport=5060;branch=z9hG4bK0c37c7a5

contact: <sip:2909294@tattele.com:5060;maddr=89.xxx.125.48>

supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,com.nortelnetworks.im.encryption

Content-Length: 0

<————->

— (9 headers 0 lines) —

Transmitting (NAT) to 89.xx.125.48:5060:

ACK sip:2909294@89.xxx.125.48 SIP/2.0

Via: SIP/2.0/UDP 192.168.101.180:5060;branch=z9hG4bK0c37c7a5;rport

Max-Forwards: 70

From: «103» <sip:5620640@192.168.101.180>;tag=as4d695e92

To: <sip:2909294@89.xxx.125.48>;tag=2083351771

Contact: <sip:5620640@192.168.101.180>

Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180

CSeq: 102 ACK

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

Content-Length: 0

—

— SIP/gts-sip-b79a2530 is circuit-busy

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

— Executing [s@macro-dialout-trunk:20] Goto(«SIP/103-08d2bf20», «s-CONGESTION,1») in new stack

— Goto (macro-dialout-trunk,s-CONGESTION,1)

— Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf(«SIP/103-08d2bf20», «1?noreport») in new stack

— Goto (macro-dialout-trunk,s-CONGESTION,3)

— Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp(«SIP/103-08d2bf20», «TRUNK Dial failed due to CONGESTION — failing through to other trunks») in new stack

— Executing [78432909294@from-internal:5] Macro(«SIP/103-08d2bf20», «outisbusy,») in new stack

— Executing [s@macro-outisbusy:1] Playback(«SIP/103-08d2bf20», «all-circuits-busy-now,noanswer») in new stack

— <SIP/103-08d2bf20> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)

Really destroying SIP dialog ‘75408280436ca6a31df57c0f30f3e8d6@192.168.101.180’ Method: INVITE

— Executing [s@macro-outisbusy:2] Playback(«SIP/103-08d2bf20», «pls-try-call-later,noanswer») in new stack

— <SIP/103-08d2bf20> Playing ‘pls-try-call-later.ulaw’ (language ‘en’)

— Executing [s@macro-outisbusy:3] Macro(«SIP/103-08d2bf20», «hangupcall») in new stack

— Executing [s@macro-hangupcall:1] ResetCDR(«SIP/103-08d2bf20», «vw») in new stack

— Executing [s@macro-hangupcall:2] NoCDR(«SIP/103-08d2bf20», «») in new stack

— Executing [s@macro-hangupcall:3] GotoIf(«SIP/103-08d2bf20», «1?skiprg») in new stack

— Goto (macro-hangupcall,s,6)

— Executing [s@macro-hangupcall:6] GotoIf(«SIP/103-08d2bf20», «1?skipblkvm») in new stack

— Goto (macro-hangupcall,s,9)

— Executing [s@macro-hangupcall:9] GotoIf(«SIP/103-08d2bf20», «1?theend») in new stack

— Goto (macro-hangupcall,s,11)

— Executing [s@macro-hangupcall:11] Hangup(«SIP/103-08d2bf20», «») in new stack

== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/103-08d2bf20’ in macro ‘hangupcall’

== Spawn extension (macro-outisbusy, s, 3) exited non-zero on ‘SIP/103-08d2bf20’ in macro ‘outisbusy’

== Spawn extension (from-internal, 78432909294, 5) exited non-zero on ‘SIP/103-08d2bf20’

— Executing [h@from-internal:1] Macro(«SIP/103-08d2bf20», «hangupcall») in new stack

— Executing [s@macro-hangupcall:1] ResetCDR(«SIP/103-08d2bf20», «vw») in new stack

— Executing [s@macro-hangupcall:2] NoCDR(«SIP/103-08d2bf20», «») in new stack

— Executing [s@macro-hangupcall:3] GotoIf(«SIP/103-08d2bf20», «1?skiprg») in new stack

— Goto (macro-hangupcall,s,6)

— Executing [s@macro-hangupcall:6] GotoIf(«SIP/103-08d2bf20», «1?skipblkvm») in new stack

— Goto (macro-hangupcall,s,9)

— Executing [s@macro-hangupcall:9] GotoIf(«SIP/103-08d2bf20», «1?theend») in new stack

— Goto (macro-hangupcall,s,11)

— Executing [s@macro-hangupcall:11] Hangup(«SIP/103-08d2bf20», «») in new stack

== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/103-08d2bf20’ in macro ‘hangupcall’

== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/103-08d2bf20’

== End MixMonitor Recording SIP/103-08d2bf20

Источник

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