* + Addpac 1108f не проходят звонки
need help!
|
Сообщений: 6
|
* + Addpac 1108f не проходят звонки
Доброго времени суток. Проблема в следующем, есть Asterisk по сети связан с Addpac 1108f, до установки FreePBX все работало, после установки обычные телефоны между собой звонят нормально, при звонке на шлюз выдает Service Unavailable. Все пиры зарегистрированы на сервере.
312/312 10.7.8.29 D N A 36423 OK (115 ms)
311 (Unspecified) D N A 0 UNKNOWN
310/310 10.2.3.23 D A 63460 OK (103 ms)
309 (Unspecified) D N A 0 UNKNOWN
308/308 10.2.3.232 D N A 5060 OK (46 ms)
307/307 10.2.3.94 D A 5060 OK (35 ms)
306/306 10.2.3.94 D A 5060 OK (38 ms)
305/305 10.2.3.94 D A 5060 OK (38 ms)
304/304 10.2.3.94 D A 5060 OK (38 ms)
303/303 10.2.3.94 D A 5060 OK (38 ms)
302/302 10.2.3.94 D A 5060 OK (38 ms)
301/301 10.2.3.94 D A 5060 OK (43 ms)
300/300 10.2.3.94 D A 5060 OK (38 ms)
вот что выдает debug
[Jan 21 14:55:18] Scheduling destruction of SIP dialog '01f887f634618c9e58328ddf38aa7b0a@10.2.3.2' in 6400 ms (Method: NOTIFY)
[Jan 21 14:55:18] Reliably Transmitting (no NAT) to 10.2.3.94:5060:
NOTIFY sip:300@10.2.3.94 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK05fb766f;rport
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as043781ae
To: <sip:300@10.2.3.94>
Contact: <sip:Unknown@10.2.3.2>
Call-ID: 01f887f634618c9e58328ddf38aa7b0a@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 86
Messages-Waiting: no
Message-Account: sip:*97@10.2.3.2
Voice-Message: 0/0 (0/0)
---
[Jan 21 14:55:19]
<--- SIP read from 10.2.3.94:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.2.3.25060;branch=z9hG4bK05fb766f;rport
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as043781ae
To: <sip:300@10.2.3.94>
Call-ID: 01f887f634618c9e58328ddf38aa7b0a@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0
<------------->
[Jan 21 14:55:19] --- (8 headers 0 lines) ---
[Jan 21 14:55:20] Scheduling destruction of SIP dialog '51770c0906f545f3364fc0d51aad3e24@10.2.3.2' in 6400 ms (Method: NOTIFY)
[Jan 21 14:55:20] Reliably Transmitting (no NAT) to 10.2.3.94:5060:
NOTIFY sip:305@10.2.3.94 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK5aa66152
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as56e3a207
To: <sip:305@10.2.3.94>
Contact: <sip:Unknown@10.2.3.2>
Call-ID: 51770c0906f545f3364fc0d51aad3e24@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 86
Messages-Waiting: no
Message-Account: sip:*97@10.2.3.2
Voice-Message: 0/0 (0/0)
---
[Jan 21 14:55:20] Scheduling destruction of SIP dialog '7e6110ce2d78e6c7171fdf194adfe9c3@10.2.3.2' in 6400 ms (Method: NOTIFY)
[Jan 21 14:55:20] Reliably Transmitting (no NAT) to 10.2.3.94:5060:
NOTIFY sip:304@10.2.3.94 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK487bea11
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as00118516
To: <sip:304@10.2.3.94>
Contact: <sip:Unknown@10.2.3.2>
Call-ID: 7e6110ce2d78e6c7171fdf194adfe9c3@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 86
Messages-Waiting: no
Message-Account: sip:*97@10.2.3.2
Voice-Message: 0/0 (0/0)
---
[Jan 21 14:55:20] Scheduling destruction of SIP dialog '03cf103d770300f9171e130b6d529945@10.2.3.2' in 6400 ms (Method: NOTIFY)
[Jan 21 14:55:20] Reliably Transmitting (no NAT) to 10.2.3.94:5060:
NOTIFY sip:303@10.2.3.94 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK53010817
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as75e76212
To: <sip:303@10.2.3.94>
Contact: <sip:Unknown@10.2.3.2>
Call-ID: 03cf103d770300f9171e130b6d529945@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 86
Messages-Waiting: no
Message-Account: sip:*97@10.2.3.2
Voice-Message: 0/0 (0/0)
---
[Jan 21 14:55:20] Scheduling destruction of SIP dialog '08d4d76765c8807664cca3233164b8d2@10.2.3.2' in 6400 ms (Method: NOTIFY)
[Jan 21 14:55:20] Reliably Transmitting (no NAT) to 10.2.3.94:5060:
NOTIFY sip:302@10.2.3.94 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK34e76737
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as7081d17d
To: <sip:302@10.2.3.94>
Contact: <sip:Unknown@10.2.3.2>
Call-ID: 08d4d76765c8807664cca3233164b8d2@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 86
Messages-Waiting: no
Message-Account: sip:*97@10.2.3.2
Voice-Message: 0/0 (0/0)
---
[Jan 21 14:55:20] Scheduling destruction of SIP dialog '481b47762505268337de3d951dea0a64@10.2.3.2' in 6400 ms (Method: NOTIFY)
[Jan 21 14:55:20] Reliably Transmitting (no NAT) to 10.2.3.94:5060:
NOTIFY sip:301@10.2.3.94 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK5ab4bf1f
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as0d69500b
To: <sip:301@10.2.3.94>
Contact: <sip:Unknown@10.2.3.2>
Call-ID: 481b47762505268337de3d951dea0a64@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 86
Messages-Waiting: no
Message-Account: sip:*97@10.2.3.2
Voice-Message: 0/0 (0/0)
---
[Jan 21 14:55:20]
<--- SIP read from 10.2.3.94:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK5aa66152
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as56e3a207
To: <sip:305@10.2.3.94>
Call-ID: 51770c0906f545f3364fc0d51aad3e24@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0
<------------->
[Jan 21 14:55:20] --- (8 headers 0 lines) ---
[Jan 21 14:55:20]
<--- SIP read from 10.2.3.94:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK487bea11
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as00118516
To: <sip:304@10.2.3.94>
Call-ID: 7e6110ce2d78e6c7171fdf194adfe9c3@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0
<------------->
[Jan 21 14:55:20] --- (8 headers 0 lines) ---
[Jan 21 14:55:20]
<--- SIP read from 10.2.3.23:63460 --->
INVITE sip:300@10.2.3.2 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-cc39ad161b542874-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:310@10.2.3.23:63460>
To: "300"<sip:300@10.2.3.2>
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 303
v=0
o=- 0 2 IN IP4 10.2.3.23
s=CounterPath X-Lite 3.0
c=IN IP4 10.2.3.23
t=0 0
m=audio 63362 RTP/AVP 107 0 8 101
a=alt:1 2 : P9X/l103 iTf16sGf 10.2.3.23 63362
a=alt:2 1 : 4r5W1gQG CF2xNxZX 10.2.25.15 63362
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
[Jan 21 14:55:20] --- (12 headers 12 lines) ---
[Jan 21 14:55:20] Sending to 10.2.3.23 : 63460 (NAT)
[Jan 21 14:55:20] Using INVITE request as basis request - NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
[Jan 21 14:55:20]
<--- Reliably Transmitting (no NAT) to 10.2.3.23:63460 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-cc39ad161b542874-1---d8754z-;received=10.2.3.23;rport=63460
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
To: "300"<sip:300@10.2.3.2>;tag=as2c05fe3a
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="177fd972"
Content-Length: 0
<------------>
[Jan 21 14:55:20] Scheduling destruction of SIP dialog 'NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.' in 32000 ms (Method: INVITE)
[Jan 21 14:55:20] Found user '310'
[Jan 21 14:55:20]
<--- SIP read from 10.2.3.94:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK53010817
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as75e76212
To: <sip:303@10.2.3.94>
Call-ID: 03cf103d770300f9171e130b6d529945@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0
<------------->
[Jan 21 14:55:20] --- (8 headers 0 lines) ---
[Jan 21 14:55:20]
<--- SIP read from 10.2.3.23:63460 --->
ACK sip:300@10.2.3.2 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-cc39ad161b542874-1---d8754z-;rport
To: "300"<sip:300@10.2.3.2>;tag=as2c05fe3a
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 1 ACK
Content-Length: 0
<------------->
[Jan 21 14:55:20] --- (7 headers 0 lines) ---
[Jan 21 14:55:20]
<--- SIP read from 10.2.3.23:63460 --->
INVITE sip:300@10.2.3.2 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:310@10.2.3.23:63460>
To: "300"<sip:300@10.2.3.2>
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="310",realm="asterisk",nonce="177fd972",uri="sip:300@10.2.3.2",response="55f251ad6fba4b09d65ecf19bdc05071",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 303
v=0
o=- 0 2 IN IP4 10.2.3.23
s=CounterPath X-Lite 3.0
c=IN IP4 10.2.3.23
t=0 0
m=audio 63362 RTP/AVP 107 0 8 101
a=alt:1 2 : P9X/l103 iTf16sGf 10.2.3.23 63362
a=alt:2 1 : 4r5W1gQG CF2xNxZX 10.2.25.15 63362
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
[Jan 21 14:55:20] --- (13 headers 12 lines) ---
[Jan 21 14:55:20] Sending to 10.2.3.23 : 63460 (NAT)
[Jan 21 14:55:20] Using INVITE request as basis request - NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
[Jan 21 14:55:20] Found user '310'
[Jan 21 14:55:20] Found RTP audio format 107
[Jan 21 14:55:20] Found RTP audio format 0
[Jan 21 14:55:20] Found RTP audio format 8
[Jan 21 14:55:20] Found RTP audio format 101
[Jan 21 14:55:20] Peer audio RTP is at port 10.2.3.23:63362
[Jan 21 14:55:20] Found unknown media description format BV32 for ID 107
[Jan 21 14:55:20] Found audio description format telephone-event for ID 101
[Jan 21 14:55:20] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Jan 21 14:55:20] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Jan 21 14:55:20] Peer audio RTP is at port 10.2.3.23:63362
[Jan 21 14:55:20] Looking for 300 in from-internal (domain 10.2.3.2)
[Jan 21 14:55:20] list_route: hop: <sip:310@10.2.3.23:63460>
[Jan 21 14:55:20]
<--- Transmitting (no NAT) to 10.2.3.23:63460 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;received=10.2.3.23;rport=63460
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
To: "300"<sip:300@10.2.3.2>
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:300@10.2.3.2>
Content-Length: 0
<------------>
[Jan 21 14:55:20]
<--- SIP read from 10.2.3.94:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK34e76737
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as7081d17d
To: <sip:302@10.2.3.94>
Call-ID: 08d4d76765c8807664cca3233164b8d2@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0
<------------->
[Jan 21 14:55:20] --- (8 headers 0 lines) ---
[Jan 21 14:55:20]
<--- SIP read from 10.2.3.94:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK5ab4bf1f
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as0d69500b
To: <sip:301@10.2.3.94>
Call-ID: 481b47762505268337de3d951dea0a64@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0
<------------->
[Jan 21 14:55:20] --- (8 headers 0 lines) ---
[Jan 21 14:55:21] Scheduling destruction of SIP dialog '3326a66c697667420423a5664c9f8f77@10.2.3.2' in 6400 ms (Method: NOTIFY)
[Jan 21 14:55:22] Reliably Transmitting (no NAT) to 10.2.3.94:5060:
NOTIFY sip:307@10.2.3.94 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK1040fa4a
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as4eea1288
To: <sip:307@10.2.3.94>
Contact: <sip:Unknown@10.2.3.2>
Call-ID: 3326a66c697667420423a5664c9f8f77@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 86
Messages-Waiting: no
Message-Account: sip:*97@10.2.3.2
Voice-Message: 0/0 (0/0)
---
[Jan 21 14:55:22]
<--- SIP read from 10.2.3.23:63460 --->
CANCEL sip:300@10.2.3.2 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;rport
To: "300"<sip:300@10.2.3.2>
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 CANCEL
Proxy-Authorization: Digest username="310",realm="asterisk",nonce="177fd972",uri="sip:300@10.2.3.2",response="a3924b69ee40acecf10cbd17802c9b5e",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0
<------------->
[Jan 21 14:55:22] --- (9 headers 0 lines) ---
[Jan 21 14:55:22] Sending to 10.2.3.23 : 63460 (NAT)
[Jan 21 14:55:22]
<--- Reliably Transmitting (NAT) to 10.2.3.23:63460 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;received=10.2.3.23;rport=63460
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
To: "300"<sip:300@10.2.3.2>;tag=as395d5842
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[Jan 21 14:55:22]
<--- Transmitting (NAT) to 10.2.3.23:63460 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;received=10.2.3.23;rport=63460
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
To: "300"<sip:300@10.2.3.2>;tag=as395d5842
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:300@10.2.3.2>
Content-Length: 0
<------------>
[Jan 21 14:55:22] Scheduling destruction of SIP dialog '799c72a210bc51bc5b9c3b7f2e9d4396@10.2.3.2' in 6400 ms (Method: NOTIFY)
[Jan 21 14:55:22] Reliably Transmitting (no NAT) to 10.2.3.94:5060:
NOTIFY sip:306@10.2.3.94 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK4cb1148d
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as5e3995f2
To: <sip:306@10.2.3.94>
Contact: <sip:Unknown@10.2.3.2>
Call-ID: 799c72a210bc51bc5b9c3b7f2e9d4396@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 86
Messages-Waiting: no
Message-Account: sip:*97@10.2.3.2
Voice-Message: 0/0 (0/0)
---
[Jan 21 14:55:22]
<--- SIP read from 10.2.3.23:63460 --->
CANCEL sip:300@10.2.3.2 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;rport
To: "300"<sip:300@10.2.3.2>
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 CANCEL
Proxy-Authorization: Digest username="310",realm="asterisk",nonce="177fd972",uri="sip:300@10.2.3.2",response="a3924b69ee40acecf10cbd17802c9b5e",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0
<------------->
[Jan 21 14:55:22] --- (9 headers 0 lines) ---
[Jan 21 14:55:22] Sending to 10.2.3.23 : 63460 (NAT)
[Jan 21 14:55:22]
<--- Reliably Transmitting (NAT) to 10.2.3.23:63460 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;received=10.2.3.23;rport=63460
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
To: "300"<sip:300@10.2.3.2>;tag=as395d5842
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[Jan 21 14:55:22]
<--- Transmitting (NAT) to 10.2.3.23:63460 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;received=10.2.3.23;rport=63460
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
To: "300"<sip:300@10.2.3.2>;tag=as395d5842
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:300@10.2.3.2>
Content-Length: 0
<------------>
[Jan 21 14:55:22]
<--- SIP read from 10.2.3.23:63460 --->
ACK sip:300@10.2.3.2 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;rport
To: "300"<sip:300@10.2.3.2>;tag=as395d5842
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 ACK
Content-Length: 0
<------------->
[Jan 21 14:55:22] --- (7 headers 0 lines) ---
[Jan 21 14:55:22]
<--- SIP read from 10.2.3.23:63460 --->
ACK sip:300@10.2.3.2 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;rport
To: "300"<sip:300@10.2.3.2>;tag=as395d5842
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 ACK
Content-Length: 0
<------------->
[Jan 21 14:55:22] --- (7 headers 0 lines) ---
[Jan 21 14:55:22]
<--- SIP read from 10.2.3.94:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK1040fa4a
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as4eea1288
To: <sip:307@10.2.3.94>
Call-ID: 3326a66c697667420423a5664c9f8f77@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0
<------------->
[Jan 21 14:55:22] --- (8 headers 0 lines) ---
[Jan 21 14:55:22]
<--- SIP read from 10.2.3.94:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK4cb1148d
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as5e3995f2
To: <sip:306@10.2.3.94>
Call-ID: 799c72a210bc51bc5b9c3b7f2e9d4396@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0
<------------->
[Jan 21 14:55:22] --- (8 headers 0 lines) ---
[Jan 21 14:55:23] Retransmitting #1 (NAT) to 10.2.3.23:63460:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;received=10.2.3.23;rport=63460
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
To: "300"<sip:300@10.2.3.2>;tag=as395d5842
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Jan 21 14:55:23] bug
<--- SIP read from 10.2.3.23:63460 --->
ACK sip:300@10.2.3.2 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;rport
To: "300"<sip:300@10.2.3.2>;tag=as395d5842
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 ACK
Content-Length: 0
<------------->
[Jan 21 14:55:23] --- (7 headers 0 lines) ---
[Jan 21 14:55:24] Retransmitting #2 (NAT) to 10.2.3.23:63460:ZDliOGM3YmMzNTJhM2FjMjFiYWMwNDQzNmMxYmE.' Method: ACK
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;received=10.2.3.23;rport=63460
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
To: "300"<sip:300@10.2.3.2>;tag=as395d5842
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Jan 21 14:55:24] 4:54:16] Really destroying SIP dialog 'M2Y4ZDliOGM3YmMzNTJhM2FjMjFiYWMwNDQzNmMxYmE.' Method: ACK
<--- SIP read from 10.2.3.23:63460 --->
ACK sip:300@10.2.3.2 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.23:63460;branch=z9hG4bK-d8754z-b71c0c424e1d7c56-1---d8754z-;rport
To: "300"<sip:300@10.2.3.2>;tag=as395d5842
From: "nucleo"<sip:310@10.2.3.2>;tag=e64d795e
Call-ID: NGZhYzM4NmQwZTk4MTc2MDdhM2FmYTk1ZjZkYWQwYmM.
CSeq: 2 ACK
Content-Length: 0
пытаюсь хоть, что то сделать уже 4-й день... все здравые мысли покинули приделы мозга...
|
|
Сообщений: 87
|
Re: * + Addpac 1108f не проходят звонки
а что говорит дебаг на аддпаке?
|
|
Откуда: SPb
Сообщений: 1307
|
Re: * + Addpac 1108f не проходят звонки
311 (Unspecified) D N A 0 UNKNOWN
309 (Unspecified) D N A 0 UNKNOWN
Прям таки все зарегестрированы?
|
|
Откуда: SPb
Сообщений: 1307
|
Re: * + Addpac 1108f не проходят звонки
поставьте пирам
nat=no
|
|
Сообщений: 6521
|
Re: * + Addpac 1108f не проходят звонки
Уважаемый nucleo,
не дофига ли плюхаете сходу цифробуквенного мусору? Ещё только лога загрузки не хватает.
Event: message-summary - это сообщения о наличии сообщений в ящиках голосовой почты. На кой их сюда?
Кстати, в сутках ничего такого, чтобы напоминало о присутствии доброго/злого времени нет.
|
|
Сообщений: 6
|
Re: * + Addpac 1108f не проходят звонки
уважаемый ded, не спорю, что много, но если не дать ничего, начнутся рассказы про телепатов, сам точно также поступаю зачастую, когда юзеры говорят "у меня тут сломалось"...
zzuz - эти пиры отключены от сети физически, следовательно они и не зарегистрированы...
sles - вот дебаг
Sending SIP PDU to ( 10.2.3.2:5060 ) from 5060
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK6f46f9ac;rport
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as1e5d06bb
To: <sip:300@10.2.3.94>
Call-ID: 5c3ad71e49be02145231082815f395ac@10.2.3.2
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0
Received SIP PDU from ( 10.2.3.2:5060 )
OPTIONS sip:300@10.2.3.94 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK0e0d7cf6;rport
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as4b76ac2f
To: <sip:300@10.2.3.94>
Contact: <sip:Unknown@10.2.3.2>
Call-ID: 3e12eb3f7f1e4f7322c50dc114dbbf43@94.158.46.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 21 Jan 2010 13:25:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Sending SIP PDU to ( 10.2.3.2:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.3.2:5060;branch=z9hG4bK0e0d7cf6;rport
From: "Unknown" <sip:Unknown@10.2.3.2>;tag=as4b76ac2f
To: <sip:300@10.2.3.94>
Call-ID: 3e12eb3f7f1e4f7322c50dc114dbbf43@10.2.3.2
CSeq: 102 OPTIONS
User-Agent: AddPac SIP Gateway
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Length: 0
Sending SIP PDU to ( 10.2.3.2:5060 ) from 5060
INVITE sip:300@sip.lds.net.ua SIP/2.0
Via: SIP/2.0/UDP 10.2.3.94:5060;branch=z9hG4bK2c4ba20da43
From: <sip:125@10.2.3.2>;tag=2c4ba20da4
To: <sip:300@10.2.3.2>
Call-ID: 2c77584b-8975-a28a-800d-0002a4039218@10.2.3.94
CSeq: 3 INVITE
Supported: timer, replaces, early-session
Min-SE: 1800
Date: Thu, 21 Jan 2010 15:47:56 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:125@10.2.3.94>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 298
Max-Forwards: 70
v=0
o=125 1264088876 1264088876 IN IP4 10.2.3.94
s=AddPac Gateway SDP
c=IN IP4 10.2.3.94
t=1264088876 0
m=audio 23004 RTP/AVP 18 4 8 0 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Received SIP PDU from ( 10.2.3.2:5060 )
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.2.3.94:5060;branch=z9hG4bK2c4ba20da43;received=10.2.3.94
From: <sip:125@10.2.3.2>;tag=2c4ba20da4
To: <sip:300@10.2.3.2>;tag=as0fbe05c7
Call-ID: 2c77584b-8975-a28a-800d-0002a4039218@10.2.3.94
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4d8fd0c5"
Content-Length: 0
Sending SIP PDU to ( 10.2.3.2:5060 ) from 5060
ACK sip:300@10.2.3.2 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.94:5060;branch=z9hG4bK2c4ba20da43
From: <sip:125@10.2.3.2>;tag=2c4ba20da4
To: <sip:300@10.2.3.2>;tag=as0fbe05c7
Call-ID: 2c77584b-8975-a28a-800d-0002a4039218@10.2.3.94
CSeq: 3 ACK
Content-Length: 0
Max-Forwards: 70
Sending SIP PDU to ( sip.lds.net.ua:5060 ) from 5060
INVITE sip:300@sip.lds.net.ua SIP/2.0
Via: SIP/2.0/UDP 10.2.3.94:5060;branch=z9hG4bK2c4ba20da44
From: <sip:125@10.2.3.2>;tag=2c4ba20da4
To: <sip:300@10.2.3.2>
Call-ID: 2c77584b-8975-a28a-800d-0002a4039218@10.2.3.94
CSeq: 4 INVITE
Supported: timer, replaces, early-session
Min-SE: 1800
Date: Thu, 21 Jan 2010 15:47:56 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:125@10.2.3.94>
Accept: application/sdp
Proxy-Authorization: Digest username="300", realm="asterisk", nonce="4d8fd0c5", uri="sip:300@10.2.3.2", response="53bcde5fd563ddc9046c3b8d63a468c7", algorithm=MD5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 298
Max-Forwards: 70
v=0
o=125 1264088876 1264088876 IN IP4 10.2.3.94
s=AddPac Gateway SDP
c=IN IP4 10.2.3.94
t=1264088876 0
m=audio 23004 RTP/AVP 18 4 8 0 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Received SIP PDU from ( 10.2.3.2:5060 )
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.2.3.94:5060;branch=z9hG4bK2c4ba20da44;received=10.2.3.94
From: <sip:125@10.2.3.2>;tag=2c4ba20da4
To: <sip:300@10.2.3.2>;tag=as0fbe05c7
Call-ID: 2c77584b-8975-a28a-800d-0002a4039218@10.2.3.94
CSeq: 4 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Sending SIP PDU to ( 10.2.3.2:5060 ) from 5060
ACK sip:300@10.2.3.2 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.94:5060;branch=z9hG4bK2c4ba20da44
From: <sip:125@10.2.3.2>;tag=2c4ba20da4
To: <sip:300@10.2.3.2>;tag=as0fbe05c7
Call-ID: 2c77584b-8975-a28a-800d-0002a4039218@10.2.3.94
CSeq: 4 ACK
Proxy-Authorization: Digest username="300", realm="asterisk", nonce="4d8fd0c5", uri="sip:300@10.2.3.2", response="8703c7747d59629f3eb879a005184177", algorithm=MD5
Content-Length: 0
Max-Forwards: 70
забыл добавить в адпаке все 8 портов FXO
|
|
Сообщений: 6521
|
Re: * + Addpac 1108f не проходят звонки
Все ответы внутри ваших вопросов.
SIP/2.0 407 Proxy Authentication Required - для тех кто в броне.
SIP/2.0 403 Forbidden на CSeq: 4 INVITE потому что не пустит звонить без авторизации.
Умейте регистрировать Аддпак как 300@sip.lds.net.ua на своём ресурсе :)
Между "если не дать ничего" и горой мусора - огромное пространство для профессионалов.
|
|
Сообщений: 6
|
Re: * + Addpac 1108f не проходят звонки
меня это тоже немного смутило... но настройки не менялись, единственное, что изменилось, это установка freePBX, после чего конфиги Asteriska были удалены, пиры добавлял через веб-интерфейс точно такие же, как и были до этого... тогда возникае вопрос, почему аддпак не хочет регаться?
|
|
Сообщений: 6521
|
Re: * + Addpac 1108f не проходят звонки
Он хочет возможно, но не может?
пиры добавлял через веб-интерфейс точно такие же
уже сомневаюсь!
Сравните у себя две картинки: страые конфиги (ещё не позно их достать из мусорника) и то что получилось. |
|
Сообщений: 6
|
Re: * + Addpac 1108f не проходят звонки
возможно, тогда может подскажите с какой стороны капать
вот конфиг с аддпка
Pots peer 0
dest-pattern = 125
port = 0/0
prefix =
forward digits = default
register E.164 = yes
user-name = 300
user-password = ***
preference = 0
huntstop = no
translate-outgoing called-number = -1
translate-outgoing calling-number = -1
call diversion = -1
allowed inbound pots peer = -1
outbound call barred group = -1
enable Call Waiting = no
administrative status = up
VoIP peer 300
dest-pattern = T
session-target = sip-server
session-protocol = SIP
answer-address =
codec = default
voice codec class = 1
dtmfRelay = rtp-2833
vad = yes
sid = yes
redundant RTP = no
description =
preference = 0
huntstop = no
translate-outgoing called-number = -1
translate-outgoing calling-number = -1
translate-outgoing digits in call = -1
call diversion = -1
outbound call barred group = -1
fax mode = system
fax rate (bps) = system
fax T38 redundancy = system
max call forward hop = 4
administrative status = up
вот конфиг *
[300]
deny=0.0.0.0/0.0.0.0
disallow=all
secret=***
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=never
port=5060
qualify=yes
callgroup=
pickupgroup=
allow=alaw
allow=ulaw
dial=SIP/300,30,Ttm
accountcode=
mailbox=300@device
permit=0.0.0.0/0.0.0.0
callerid=device <300>
call-limit=50
честно знаю, что ошибка очевидная, но я уже не могу ее найти... голова взрывается...
|
|
