Не работают исходящие вызовы DVG-6008s +Asterisk

hatun · Сообщение **hatun** » 04 дек 2012, 09:40

При исходящем звонке поставил debug.
Смущает строка :

asterisk*CLI> 
  == Using SIP RTP CoS mark 5
    -- Executing [989144211111@phones:1] Dial("SIP/3001-00000076", "SIP/2418/89144211111,,") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 11208
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.2.3.42:5060:
INVITE sip:89144211111@10.2.3.42:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK56655863
Max-Forwards: 70
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>
Contact: <sip:3001@10.2.3.42:5060>
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.18.0
Date: Tue, 04 Dec 2012 16:56:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 45549556 45549556 IN IP4 10.2.3.42
s=Asterisk PBX 1.8.18.0
c=IN IP4 10.2.3.42
t=0 0
m=audio 11208 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/2418/89144211111

<--- SIP read from UDP:10.2.3.42:5060 --->
INVITE sip:89144211111@10.2.3.42:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK56655863
Max-Forwards: 70
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>
Contact: <sip:3001@10.2.3.42:5060>
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.18.0
Date: Tue, 04 Dec 2012 16:56:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 45549556 45549556 IN IP4 10.2.3.42
s=Asterisk PBX 1.8.18.0
c=IN IP4 10.2.3.42
t=0 0
m=audio 11208 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 10.2.3.42:5060 (NAT)
Using INVITE request as basis request - 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
Found peer '3001' for '3001' from 10.2.3.42:5060

<--- Reliably Transmitting (NAT) to 10.2.3.42:5060 --->
[b]SIP/2.0 401 Unauthorized[/b]
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK56655863;received=10.2.3.42;rport=5060
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>;tag=as7c7b3ac1
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="36e311a4"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:10.2.3.42:5060 --->
[b]SIP/2.0 401 Unauthorized[/b]
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK56655863;received=10.2.3.42;rport=5060
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>;tag=as7c7b3ac1
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="36e311a4"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 10.2.3.42:5060:
ACK sip:89144211111@10.2.3.42:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK56655863
Max-Forwards: 70
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>;tag=as7c7b3ac1
Contact: <sip:3001@10.2.3.42:5060>
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.18.0
Content-Length: 0


---
Audio is at 11208
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.2.3.42:5060:
INVITE sip:89144211111@10.2.3.42:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK64a0ca4c
Max-Forwards: 70
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>
Contact: <sip:3001@10.2.3.42:5060>
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.18.0
Authorization: Digest username="2418", realm="asterisk", algorithm=MD5, uri="sip:89144211111@10.2.3.42:5060", nonce="36e311a4", response="df29e1ffc53ddf6c49e155884075e960"
Date: Tue, 04 Dec 2012 16:56:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 45549556 45549557 IN IP4 10.2.3.42
s=Asterisk PBX 1.8.18.0
c=IN IP4 10.2.3.42
t=0 0
m=audio 11208 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:10.2.3.42:5060 --->
ACK sip:89144211111@10.2.3.42:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK56655863
Max-Forwards: 70
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>;tag=as7c7b3ac1
Contact: <sip:3001@10.2.3.42:5060>
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.18.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:10.2.3.42:5060 --->
INVITE sip:89144211111@10.2.3.42:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK64a0ca4c
Max-Forwards: 70
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>
Contact: <sip:3001@10.2.3.42:5060>
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.18.0
Authorization: Digest username="2418", realm="asterisk", algorithm=MD5, uri="sip:89144211111@10.2.3.42:5060", nonce="36e311a4", response="df29e1ffc53ddf6c49e155884075e960"
Date: Tue, 04 Dec 2012 16:56:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 45549556 45549557 IN IP4 10.2.3.42
s=Asterisk PBX 1.8.18.0
c=IN IP4 10.2.3.42
t=0 0
m=audio 11208 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 12 lines) ---

<--- Transmitting (no NAT) to 10.2.3.42:5060 --->
[b]SIP/2.0 482 Loop Detected[/b]
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK64a0ca4c;received=10.2.3.42
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>;tag=as2b083b81
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:10.2.3.42:5060 --->
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK64a0ca4c;received=10.2.3.42
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>;tag=as2b083b81
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
    -- Got SIP response 482 "Loop Detected" back from 10.2.3.42:5060
Transmitting (no NAT) to 10.2.3.42:5060:
ACK sip:89144211111@10.2.3.42:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK64a0ca4c
Max-Forwards: 70
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>;tag=as2b083b81
Contact: <sip:3001@10.2.3.42:5060>
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.18.0
Content-Length: 0


---

<--- SIP read from UDP:10.2.3.42:5060 --->
ACK sip:89144211111@10.2.3.42:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK64a0ca4c
Max-Forwards: 70
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>;tag=as2b083b81
Contact: <sip:3001@10.2.3.42:5060>
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.18.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- SIP/2418-00000077 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/3001-00000076' status is 'CONGESTION'
Really destroying SIP dialog '48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060' Method: INVITE
Really destroying SIP dialog '575a2c101f63c5072cfe426e08a9987f@127.0.0.1' Method: REGISTER
asterisk*CLI>

hatun · Сообщение **hatun** » 04 дек 2012, 11:44

Насколько я понимаю, исходя из rfc 3261 пункта 16.4.

482 (Loop Detected) возникает из за множественных запросов INVATE? т.е. первый запрос обработал, а на следующие отправляет Loop Detected.

Куда копать?

ded · Сообщение **ded** » 04 дек 2012, 12:39

А какая строка смущает? Их там много.
Ранее упоминались вызовы вида
SIP/180/181
а теперь идёт
SIP/3001-00000076", "SIP/2418/89144211111
существенные изменения.

Проблема в диалплане шлюза DVG-6008s, это он направляет входящий из SIP не в АТС, а назад в Астериск, и множественные инвайты - следствие, а не причина (IMHO).

hatun · Сообщение **hatun** » 04 дек 2012, 14:32

2 ded

Код: Выделить всё

[b]SIP/2.0 482 Loop Detected[/b]
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK64a0ca4c;received=10.2.3.42
From: "3001" <sip:3001@10.2.3.42>;tag=as2b083b81
To: <sip:89144211111@10.2.3.42:5060>;tag=as2b083b81
Call-ID: 48b28a092838e4a548f67f564a06bfe7@10.2.3.42:5060
CSeq: 103 INVITE

tag одинаковый, а по логу выше разный был.

ded · Сообщение **ded** » 04 дек 2012, 15:40

Я про
-- Called SIP/180/181
-- Got SIP response 482 "Loop Detected" back from 10.2.3.42:5060
смутился несколько ранше Вас, выше - в начале страницы. Что такое tag одинаковый, а по логу выше разный был? В смысле - было
SIP/180/181 - "Loop Detected"
а сейчас SIP/2418/89144211111 ? и всё так же "Loop Detected"?

Вы забубенили (предполагаю) в диалплане DVG-6008s - направлять все (!!) вызовы, даже приходящие с SIP - в Asterisk.
Получилась петля - всё что приходит - туда же и уходит.
Возможности изучать Ваши принтскрины нету. Разберитесь в маршрутизации вызовов.

hatun · Сообщение **hatun** » 05 дек 2012, 09:09

Лог при положительном исходящем звонке

Код: Выделить всё

Connected to Asterisk 1.8.18.0 currently running on asterisk (pid = 2980)
Verbosity is at least 3
asterisk*CLI> 
asterisk*CLI> 
asterisk*CLI> 
  == Using SIP RTP CoS mark 5
    -- Executing [9700045@phones:1] Dial("SIP/3000-000000f5", "SIP/2418/700045,,tr") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/2418/700045
[Dec  6 00:33:01] NOTICE[3102]: chan_sip.c:13240 sip_reregister:    -- Re-registration for  2418@10.2.3.42
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 10.2.3.42:5060:
REGISTER sip:10.2.3.42 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK5114deaf;rport
Max-Forwards: 70
From: <sip:2418@10.2.3.42>;tag=as2e3e1a3d
To: <sip:2418@10.2.3.42>
Call-ID: 01018003251dbec66010e57b4349ec10@127.0.0.1
CSeq: 1441 REGISTER
User-Agent: Asterisk PBX 1.8.18.0
Authorization: Digest username="2418", realm="asterisk", algorithm=MD5, uri="sip:10.2.3.42", nonce="7b8b58da", response="379a274c1e107f4aa7f3ed1c1efd8b69"
Expires: 120
Contact: <sip:s@10.2.3.42:5060>
Content-Length: 0


---

<--- SIP read from UDP:10.2.3.42:5060 --->
REGISTER sip:10.2.3.42 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK5114deaf;rport
Max-Forwards: 70
From: <sip:2418@10.2.3.42>;tag=as2e3e1a3d
To: <sip:2418@10.2.3.42>
Call-ID: 01018003251dbec66010e57b4349ec10@127.0.0.1
CSeq: 1441 REGISTER
User-Agent: Asterisk PBX 1.8.18.0
Authorization: Digest username="2418", realm="asterisk", algorithm=MD5, uri="sip:10.2.3.42", nonce="7b8b58da", response="379a274c1e107f4aa7f3ed1c1efd8b69"
Expires: 120
Contact: <sip:s@10.2.3.42:5060>
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 10.2.3.42:5060 (NAT)

<--- Transmitting (no NAT) to 10.2.3.42:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK5114deaf;received=10.2.3.42;rport=5060
From: <sip:2418@10.2.3.42>;tag=as2e3e1a3d
To: <sip:2418@10.2.3.42>;tag=as2e3e1a3d
Call-ID: 01018003251dbec66010e57b4349ec10@127.0.0.1
CSeq: 1441 REGISTER
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0fec4e41"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '01018003251dbec66010e57b4349ec10@127.0.0.1' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.2.3.42:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK5114deaf;received=10.2.3.42;rport=5060
From: <sip:2418@10.2.3.42>;tag=as2e3e1a3d
To: <sip:2418@10.2.3.42>;tag=as2e3e1a3d
Call-ID: 01018003251dbec66010e57b4349ec10@127.0.0.1
CSeq: 1441 REGISTER
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0fec4e41"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name 10.2.3.42
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 10.2.3.42:5060:
REGISTER sip:10.2.3.42 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK1bfaebe9;rport
Max-Forwards: 70
From: <sip:2418@10.2.3.42>;tag=as4d935ef1
To: <sip:2418@10.2.3.42>
Call-ID: 01018003251dbec66010e57b4349ec10@127.0.0.1
CSeq: 1442 REGISTER
User-Agent: Asterisk PBX 1.8.18.0
Authorization: Digest username="2418", realm="asterisk", algorithm=MD5, uri="sip:10.2.3.42", nonce="0fec4e41", response="988c724fde58074a3e0631b93de1034f"
Expires: 120
Contact: <sip:2418@10.2.3.42:5060>
Content-Length: 0


---

<--- SIP read from UDP:10.2.3.42:5060 --->
REGISTER sip:10.2.3.42 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK1bfaebe9;rport
Max-Forwards: 70
From: <sip:2418@10.2.3.42>;tag=as4d935ef1
To: <sip:2418@10.2.3.42>
Call-ID: 01018003251dbec66010e57b4349ec10@127.0.0.1
CSeq: 1442 REGISTER
User-Agent: Asterisk PBX 1.8.18.0
Authorization: Digest username="2418", realm="asterisk", algorithm=MD5, uri="sip:10.2.3.42", nonce="0fec4e41", response="988c724fde58074a3e0631b93de1034f"
Expires: 120
Contact: <sip:2418@10.2.3.42:5060>
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 10.2.3.42:5060 (no NAT)
    -- Registered SIP '2418' at 10.2.3.42:5060
Reliably Transmitting (no NAT) to 10.2.3.42:5060:
OPTIONS sip:2418@10.2.3.42:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK164885fb
Max-Forwards: 70
From: "asterisk" <sip:2418@10.2.3.42>;tag=as738f9a45
To: <sip:2418@10.2.3.42:5060>
Contact: <sip:2418@10.2.3.42:5060>
Call-ID: 04a1378d04a713a1210b1d787140e263@10.2.3.42:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.18.0
Date: Wed, 05 Dec 2012 13:33:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 10.2.3.42:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK1bfaebe9;received=10.2.3.42;rport=5060
From: <sip:2418@10.2.3.42>;tag=as4d935ef1
To: <sip:2418@10.2.3.42>;tag=as4d935ef1
Call-ID: 01018003251dbec66010e57b4349ec10@127.0.0.1
CSeq: 1442 REGISTER
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: <sip:2418@10.2.3.42:5060>;expires=120
Date: Wed, 05 Dec 2012 13:33:01 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '01018003251dbec66010e57b4349ec10@127.0.0.1' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.2.3.42:5060 --->
OPTIONS sip:2418@10.2.3.42:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK164885fb
Max-Forwards: 70
From: "asterisk" <sip:2418@10.2.3.42>;tag=as738f9a45
To: <sip:2418@10.2.3.42:5060>
Contact: <sip:2418@10.2.3.42:5060>
Call-ID: 04a1378d04a713a1210b1d787140e263@10.2.3.42:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.18.0
Date: Wed, 05 Dec 2012 13:33:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Looking for 2418 in default (domain 10.2.3.42)

<--- Transmitting (NAT) to 10.2.3.42:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK164885fb;received=10.2.3.42;rport=5060
From: "asterisk" <sip:2418@10.2.3.42>;tag=as738f9a45
To: <sip:2418@10.2.3.42:5060>;tag=as4559d795
Call-ID: 04a1378d04a713a1210b1d787140e263@10.2.3.42:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '04a1378d04a713a1210b1d787140e263@10.2.3.42:5060' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:10.2.3.42:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK1bfaebe9;received=10.2.3.42;rport=5060
From: <sip:2418@10.2.3.42>;tag=as4d935ef1
To: <sip:2418@10.2.3.42>;tag=as4d935ef1
Call-ID: 01018003251dbec66010e57b4349ec10@127.0.0.1
CSeq: 1442 REGISTER
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: <sip:2418@10.2.3.42:5060>;expires=120
Date: Wed, 05 Dec 2012 13:33:01 GMT
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Scheduling destruction of SIP dialog '01018003251dbec66010e57b4349ec10@127.0.0.1' in 32000 ms (Method: REGISTER)
[Dec  6 00:33:01] NOTICE[3102]: chan_sip.c:21023 handle_response_register: Outbound Registration: Expiry for 10.2.3.42 is 120 sec (Scheduling reregistration in 105 s)

<--- SIP read from UDP:10.2.3.42:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.2.3.42:5060;branch=z9hG4bK164885fb;received=10.2.3.42;rport=5060
From: "asterisk" <sip:2418@10.2.3.42>;tag=as738f9a45
To: <sip:2418@10.2.3.42:5060>;tag=as4559d795
Call-ID: 04a1378d04a713a1210b1d787140e263@10.2.3.42:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '04a1378d04a713a1210b1d787140e263@10.2.3.42:5060' Method: OPTIONS
    -- SIP/2418-000000f6 answered SIP/3000-000000f5
  == Spawn extension (phones, 9700045, 1) exited non-zero on 'SIP/3000-000000f5'
asterisk*CLI> exit
[root@asterisk ~]#

ded · Сообщение **ded** » 05 дек 2012, 11:49

В этом логе нет никакого звонка. Нет ни одного пакета INVITE (возможно не заметил).
Куча регистраций, успешных и неуспешных, со статусом Unauthorized, в одних случаях
Contact: <sip:s@10.2.3.42:5060>
а в других
Contact: <sip:2418@10.2.3.42:5060>
это ерунда какая-то.

Вам бы самим разобраться?

hatun · Сообщение **hatun** » 11 дек 2012, 10:58

switch, респект!

Изменено:
не использовать строку регистрации шлюза на *.
набор номера через Dial(SIP/operator/operator${EXTEN}

switch, огромное спасибо!

SolarW · Сообщение **SolarW** » 12 дек 2012, 16:25

Всего за три страницы уговорили прочесть инструкцию?

Не работают исходящие вызовы DVG-6008s +Asterisk

Re: Не работают исходящие вызовы DVG-6008s +Asterisk

Re: Не работают исходящие вызовы DVG-6008s +Asterisk

Re: Не работают исходящие вызовы DVG-6008s +Asterisk

Re: Не работают исходящие вызовы DVG-6008s +Asterisk

Re: Не работают исходящие вызовы DVG-6008s +Asterisk

Re: Не работают исходящие вызовы DVG-6008s +Asterisk

Re: Не работают исходящие вызовы DVG-6008s +Asterisk

Re: Не работают исходящие вызовы DVG-6008s +Asterisk

Re: Не работают исходящие вызовы DVG-6008s +Asterisk