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Нет входящих звонков

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Нет входящих звонков

Добавлено: 30 янв 2017, 19:11
Legat
Добрый день.
Помогите советом.
Есть транки на SIP провайдера (Велтон), исходящие звонки проходят нормально, а входящих звонков нет.
При входящем звонке вижу:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
--- Transmitting (NAT) to 85.XX.1X2.218:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.XX.1X2.218:5060;branch=z9hG4bK01c3d1f1;received=85.XX.1X2.218;rport=5060
From: "asterisk" <sip:asterisk@85.XX.1X2.218>;tag=as29c395ad
To: <sip:57XXXX722@81.XXX.XXX.30:5660>;tag=as17a9be76
Call-ID: 35e8731510eadecc71684d395044c451@85.XX.1X2.218:5060
CSeq: 102 OPTIONS
Server: FPBX-AsteriskNOW-13.0.190.7(11.21.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:81.XXX.XXX.30:5660>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '35e8731510eadecc71684d395044c451@85.XX.1X2.218:5060' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 85.XX.1X2.218:5060:
OPTIONS sip:85.XX.1X2.218 SIP/2.0
Via: SIP/2.0/UDP 81.XXX.XXX.30:5660;branch=z9hG4bK49f96fce;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@81.XXX.XXX.30:5660>;tag=as474b7860
To: <sip:85.XX.1X2.218>
Contact: <sip:Unknown@81.XXX.XXX.30:5660>
Call-ID: 2c0a7c79557729f1307010112ee7bb5f@81.XXX.XXX.30:5660
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-13.0.190.7(11.21.2)
Date: Mon, 30 Jan 2017 16:05:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 85.XX.1X2.218:5060:
OPTIONS sip:85.XX.1X2.218 SIP/2.0
Via: SIP/2.0/UDP 81.XXX.XXX.30:5660;branch=z9hG4bK25c81d35;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@81.XXX.XXX.30:5660>;tag=as4950a24f
To: <sip:85.XX.1X2.218>
Contact: <sip:Unknown@81.XXX.XXX.30:5660>
Call-ID: 0532673871f1be9f664ea3755e73be72@81.XXX.XXX.30:5660
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-13.0.190.7(11.21.2)
Date: Mon, 30 Jan 2017 16:05:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 85.XX.1X2.218:5060:
OPTIONS sip:85.XX.1X2.218 SIP/2.0
Via: SIP/2.0/UDP 81.XXX.XXX.30:5660;branch=z9hG4bK7aeb534e;rport
Max-Forwards: 70
From: "Unknown" <sip:57XXXX722@81.XXX.XXX.30:5660>;tag=as41b5d810
To: <sip:85.XX.1X2.218>
Contact: <sip:57XXXX722@81.XXX.XXX.30:5660>
Call-ID: 7efd6e7005e6a5002787d1a11612aa98@81.XXX.XXX.30:5660
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-13.0.190.7(11.21.2)
Date: Mon, 30 Jan 2017 16:05:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:85.XX.1X2.218:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 81.XXX.XXX.30:5660;branch=z9hG4bK49f96fce;received=81.XXX.XXX.30;rport=8972
From: "Unknown" <sip:Unknown@81.XXX.XXX.30:5660>;tag=as474b7860
To: <sip:85.XX.1X2.218>;tag=as33afc5b4
Call-ID: 2c0a7c79557729f1307010112ee7bb5f@81.XXX.XXX.30:5660
CSeq: 102 OPTIONS
Server: Alcatel 5020
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 '2c0a7c79557729f1307010112ee7bb5f@81.XXX.XXX.30:5660' Method: OPTIONS

<--- SIP read from UDP:85.XX.1X2.218:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 81.XXX.XXX.30:5660;branch=z9hG4bK25c81d35;received=81.XXX.XXX.30;rport=8972
From: "Unknown" <sip:Unknown@81.XXX.XXX.30:5660>;tag=as4950a24f
To: <sip:85.XX.1X2.218>;tag=as559ae72c
Call-ID: 0532673871f1be9f664ea3755e73be72@81.XXX.XXX.30:5660
CSeq: 102 OPTIONS
Server: Alcatel 5020
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 '0532673871f1be9f664ea3755e73be72@81.XXX.XXX.30:5660' Method: OPTIONS

<--- SIP read from UDP:85.XX.1X2.218:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 81.XXX.XXX.30:5660;branch=z9hG4bK7aeb534e;received=81.XXX.XXX.30;rport=8972
From: "Unknown" <sip:57XXXX722@81.XXX.XXX.30:5660>;tag=as41b5d810
To: <sip:85.XX.1X2.218>;tag=as6b5d81da
Call-ID: 7efd6e7005e6a5002787d1a11612aa98@81.XXX.XXX.30:5660
CSeq: 102 OPTIONS
Server: Alcatel 5020
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 '7efd6e7005e6a5002787d1a11612aa98@81.XXX.XXX.30:5660' Method: OPTIONS
Really destroying SIP dialog '57ecbd1f2a3126e6319babc07f5b291d@85.XX.1X2.218:5060' Method: OPTIONS
Really destroying SIP dialog '587956d43e7542065c2935186ffdf576@85.XX.1X2.218:5060' Method: OPTIONS
Настройки Транков:
Out:
username=577XXXX646
type=peer
secret=XXXXXXXXXX
qualify=yes
nat=yes&yes
host=85.XX.1X2.218
disallow=all
context=from-internal
allow=ulaw&alaw

Ink:
type=user
username=57XXXXX46
secret=RmaJ6ATx8o
qualify=yes
nat=yes
host=85.XX.XXX.218
dtmfmode=rfc2833
disallow=all
context=from-internal
insecure=invite,port
canreinvite=no
allow=ulaw&alaw

Reg Str:
57XXXXXX46:XXXXXXX@85.XXX.1X2.218/57XXXX646

Подскажите в какую сторону копать?
Заранее спасибо.

Re: Нет входящих звонков

Добавлено: 31 янв 2017, 07:23
awsswa
В логах должен быть пакет INVITE
sip set debug ip 81.XXX.XXX.30

и что там у вас за пляски с портами 5060 и 5660 ?

Re: Нет входящих звонков

Добавлено: 31 янв 2017, 09:03
virus_net
Дебаг нужно на 85.XX.XXX.218 включать, т.к. 81.XXX.XXX.30 это его IP. ИМХО лучше использовать tcpdump.

А вот это:
Legat писал(а):nat=yes&yes
host=85.XX.1X2.218
а) Это кто вас такому научил ? б) Откуда тут NAT вообще ?

Топик для "Вопросы новичков" где бы сразу узрели:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
Изображение
Я не силен в FreePBX, но вот это:
Legat писал(а):context=from-internal
Кажется дырой в безопасности, т.к. насколько я помню контекст from-internal это внутренняя "кухня".

Приговор: Сначала идти читать книжку и доку, а тока потом пытаться что либо настраивать.

Re: Нет входящих звонков

Добавлено: 31 янв 2017, 11:31
Legat
Да, FreePBX стоит за натом, в принципе исходящие работают как с nat=no, так и с nat=yes, в других транках прописан nat=yes...

Дамп:

tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes
09:44:22.192492 IP (tos 0x0, ttl 59, id 62826, offset 0, flags [none], proto UDP (17), length 999)
85.XXX.XXX.218.sip > 10.XXX.XXX.234.5660: SIP, length: 971
INVITE sip:57XXXX22@81.17.138.30:8972 SIP/2.0
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK00162b60;rport
Max-Forwards: 70
From: "050XXXXX06" <sip:050XXXXX06@85.XXX.XXX.218>;tag=as541b1432
To: <sip:57XXXX22@81.17.138.30:8972>
Contact: <sip:050XXXXX06@85.XXX.XXX.218:5060>
Call-ID: 049b7fad06daf12d686add36415bfe32@85.XXX.XXX.218:5060
CSeq: 102 INVITE
User-Agent: Alcatel 5020
Date: Tue, 31 Jan 2017 07:44:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 380

v=0
o=root 171416626 171416626 IN IP4 85.XXX.XXX.218
s=Alcatel 5020
c=IN IP4 85.XXX.XXX.218
t=0 0
m=audio 16406 RTP/AVP 8 18 111 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

09:44:22.193246 IP (tos 0x60, ttl 64, id 42482, offset 0, flags [none], proto UDP (17), length 611)
10.XXX.XXX.234.5660 > 85.XXX.XXX.218.sip: SIP, length: 583
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK00162b60;received=85.XXX.XXX.218;rport=5060
From: "050XXXXX06" <sip:050XXXXX06@85.XXX.XXX.218>;tag=as541b1432
To: <sip:57XXXX22@81.17.138.30:8972>;tag=as46f707cb
Call-ID: 049b7fad06daf12d686add36415bfe32@85.XXX.XXX.218:5060
CSeq: 102 INVITE
Server: FPBX-AsteriskNOW-13.0.190.7(11.21.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4d16bf93"
Content-Length: 0


09:44:22.198653 IP (tos 0x0, ttl 59, id 62827, offset 0, flags [none], proto UDP (17), length 447)
85.XXX.XXX.218.sip > 10.XXX.XXX.234.5660: SIP, length: 419
ACK sip:57XXXX22@81.17.138.30:8972 SIP/2.0
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK00162b60;rport
Max-Forwards: 70
From: "050XXXXX06" <sip:050XXXXX06@85.XXX.XXX.218>;tag=as541b1432
To: <sip:57XXXX22@81.17.138.30:8972>;tag=as46f707cb
Contact: <sip:050XXXXX06@85.XXX.XXX.218:5060>
Call-ID: 049b7fad06daf12d686add36415bfe32@85.XXX.XXX.218:5060
CSeq: 102 ACK
User-Agent: Alcatel 5020
Content-Length: 0


09:44:22.198766 IP (tos 0x0, ttl 59, id 62828, offset 0, flags [none], proto UDP (17), length 1175)
85.XXX.XXX.218.sip > 10.XXX.XXX.234.5660: SIP, length: 1147
INVITE sip:57XXXX22@81.17.138.30:8972 SIP/2.0
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK32b921d0;rport
Max-Forwards: 70
From: "050XXXXX06" <sip:050XXXXX06@85.XXX.XXX.218>;tag=as541b1432
To: <sip:57XXXX22@81.17.138.30:8972>
Contact: <sip:050XXXXX06@85.XXX.XXX.218:5060>
Call-ID: 049b7fad06daf12d686add36415bfe32@85.XXX.XXX.218:5060
CSeq: 103 INVITE
User-Agent: Alcatel 5020
Authorization: Digest username="velton", realm="asterisk", algorithm=MD5, uri="sip:57XXXX22@81.17.138.30:8972", nonce="4d16bf93", response="956ed7286ee5dd6aee1d81500bb22eea"
Date: Tue, 31 Jan 2017 07:44:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 380

v=0
o=root 171416626 171416627 IN IP4 85.XXX.XXX.218
s=Alcatel 5020
c=IN IP4 85.XXX.XXX.218
t=0 0
m=audio 16406 RTP/AVP 8 18 111 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

09:44:22.199080 IP (tos 0x60, ttl 64, id 42483, offset 0, flags [none], proto UDP (17), length 532)
10.XXX.XXX.234.5660 > 85.XXX.XXX.218.sip: SIP, length: 504
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK32b921d0;received=85.XXX.XXX.218;rport=5060
From: "050XXXXX06" <sip:050XXXXX06@85.XXX.XXX.218>;tag=as541b1432
To: <sip:57XXXX22@81.17.138.30:8972>;tag=as46f707cb
Call-ID: 049b7fad06daf12d686add36415bfe32@85.XXX.XXX.218:5060
CSeq: 103 INVITE
Server: FPBX-AsteriskNOW-13.0.190.7(11.21.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


09:44:22.203573 IP (tos 0x0, ttl 59, id 62829, offset 0, flags [none], proto UDP (17), length 447)
85.XXX.XXX.218.sip > 10.XXX.XXX.234.5660: SIP, length: 419
ACK sip:57XXXX22@81.17.138.30:8972 SIP/2.0
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK32b921d0;rport
Max-Forwards: 70
From: "050XXXXX06" <sip:050XXXXX06@85.XXX.XXX.218>;tag=as541b1432
To: <sip:57XXXX22@81.17.138.30:8972>;tag=as46f707cb
Contact: <sip:050XXXXX06@85.XXX.XXX.218:5060>
Call-ID: 049b7fad06daf12d686add36415bfe32@85.XXX.XXX.218:5060
CSeq: 103 ACK
User-Agent: Alcatel 5020
Content-Length: 0


[|sip]
09:44:33.893191 IP (tos 0x0, ttl 59, id 62830, offset 0, flags [none], proto UDP (17), length 582)
85.XXX.XXX.218.sip > 10.XXX.XXX.234.5660: SIP, length: 554
OPTIONS sip:577XXXX46@81.17.138.30:5660 SIP/2.0
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK49d224db;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@85.XXX.XXX.218>;tag=as131f8f50
To: <sip:577XXXX46@81.17.138.30:5660>
Contact: <sip:asterisk@85.XXX.XXX.218:5060>
Call-ID: 233d17682283b357444ef90d09e4619d@85.XXX.XXX.218:5060
CSeq: 102 OPTIONS
User-Agent: Alcatel 5020
Date: Tue, 31 Jan 2017 07:44:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


09:44:33.893690 IP (tos 0x60, ttl 64, id 42484, offset 0, flags [none], proto UDP (17), length 581)
10.XXX.XXX.234.5660 > 85.XXX.XXX.218.sip: SIP, length: 553
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK49d224db;received=85.XXX.XXX.218;rport=5060
From: "asterisk" <sip:asterisk@85.XXX.XXX.218>;tag=as131f8f50
To: <sip:577XXXX46@81.17.138.30:5660>;tag=as324b24e6
Call-ID: 233d17682283b357444ef90d09e4619d@85.XXX.XXX.218:5060
CSeq: 102 OPTIONS
Server: FPBX-AsteriskNOW-13.0.190.7(11.21.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:81.17.138.30:5660>
Accept: application/sdp
Content-Length: 0


09:44:33.895156 IP (tos 0x0, ttl 59, id 62831, offset 0, flags [none], proto UDP (17), length 582)
85.XXX.XXX.218.sip > 10.XXX.XXX.234.5660: SIP, length: 554
OPTIONS sip:57XXXX22@81.17.138.30:5660 SIP/2.0
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK33bb6f1f;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@85.XXX.XXX.218>;tag=as32e2a38b
To: <sip:57XXXX22@81.17.138.30:5660>
Contact: <sip:asterisk@85.XXX.XXX.218:5060>
Call-ID: 294cacfe26b6309a67e7617344d9c10c@85.XXX.XXX.218:5060
CSeq: 102 OPTIONS
User-Agent: Alcatel 5020
Date: Tue, 31 Jan 2017 07:44:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


09:44:33.895481 IP (tos 0x60, ttl 64, id 42485, offset 0, flags [none], proto UDP (17), length 581)
10.XXX.XXX.234.5660 > 85.XXX.XXX.218.sip: SIP, length: 553
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK33bb6f1f;received=85.XXX.XXX.218;rport=5060
From: "asterisk" <sip:asterisk@85.XXX.XXX.218>;tag=as32e2a38b
To: <sip:57XXXX22@81.17.138.30:5660>;tag=as66226184
Call-ID: 294cacfe26b6309a67e7617344d9c10c@85.XXX.XXX.218:5060
CSeq: 102 OPTIONS
Server: FPBX-AsteriskNOW-13.0.190.7(11.21.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:81.17.138.30:5660>
Accept: application/sdp
Content-Length: 0


09:45:03.897506 IP (tos 0x0, ttl 59, id 62832, offset 0, flags [none], proto UDP (17), length 582)
85.XXX.XXX.218.sip > 10.XXX.XXX.234.5660: SIP, length: 554
OPTIONS sip:577XXXX46@81.17.138.30:5660 SIP/2.0
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK3a3c2267;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@85.XXX.XXX.218>;tag=as67491399
To: <sip:577XXXX46@81.17.138.30:5660>
Contact: <sip:asterisk@85.XXX.XXX.218:5060>
Call-ID: 336915be44d3db77183fdc725653b53a@85.XXX.XXX.218:5060
CSeq: 102 OPTIONS
User-Agent: Alcatel 5020
Date: Tue, 31 Jan 2017 07:45:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


09:45:03.897668 IP (tos 0x0, ttl 59, id 62833, offset 0, flags [none], proto UDP (17), length 582)
85.XXX.XXX.218.sip > 10.XXX.XXX.234.5660: SIP, length: 554
OPTIONS sip:57XXXX22@81.17.138.30:5660 SIP/2.0
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK02c696c8;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@85.XXX.XXX.218>;tag=as4fa13c6a
To: <sip:57XXXX22@81.17.138.30:5660>
Contact: <sip:asterisk@85.XXX.XXX.218:5060>
Call-ID: 5e232ba80624d5a052d2c9643972a75e@85.XXX.XXX.218:5060
CSeq: 102 OPTIONS
User-Agent: Alcatel 5020
Date: Tue, 31 Jan 2017 07:45:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


09:45:03.937066 IP (tos 0x60, ttl 64, id 42486, offset 0, flags [none], proto UDP (17), length 585)
10.XXX.XXX.234.5660 > 85.XXX.XXX.218.sip: SIP, length: 557
OPTIONS sip:85.XXX.XXX.218 SIP/2.0
Via: SIP/2.0/UDP 81.17.138.30:5660;branch=z9hG4bK1c089df0;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@81.17.138.30:5660>;tag=as43947649
To: <sip:85.XXX.XXX.218>
Contact: <sip:Unknown@81.17.138.30:5660>
Call-ID: 6e23bd9e2c35c1ca15da463b378e5a89@81.17.138.30:5660
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-13.0.190.7(11.21.2)
Date: Tue, 31 Jan 2017 07:45:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


09:45:03.937938 IP (tos 0x60, ttl 64, id 42487, offset 0, flags [none], proto UDP (17), length 581)
10.XXX.XXX.234.5660 > 85.XXX.XXX.218.sip: SIP, length: 553
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.XXX.XXX.218:5060;branch=z9hG4bK3a3c2267;received=85.XXX.XXX.218;rport=5060
From: "asterisk" <sip:asterisk@85.XXX.XXX.218>;tag=as67491399
To: <sip:577XXXX46@81.17.138.30:5660>;tag=as5c227681
Call-ID: 336915be44d3db77183fdc725653b53a@85.XXX.XXX.218:5060
CSeq: 102 OPTIONS
Server: FPBX-AsteriskNOW-13.0.190.7(11.21.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:81.17.138.30:5660>
Accept: application/sdp
Content-Length: 0


Текущие настройки транка:
type=user
qualify=yes
nat=no
username=57ХХХХХ22
secret=ХХХХХ
host=dynamic
dtmfmode=rfc2833
disallow=all
context=from-internal
allow=ulaw&alaw

Я так понимаю, что проблема в регистрации?

Re: Нет входящих звонков

Добавлено: 31 янв 2017, 12:16
ded
Я так понимаю, проблема в понимании типичности проблемы.
А разделе для новичков есть блок-схемы поиска и решений проблем.
Изображение

Re: Нет входящих звонков

Добавлено: 31 янв 2017, 13:00
Zavr2008
настройки от старого астера и есть подозрительные вещи типа:

Код: Выделить всё

insecure=invite,port
canreinvite=no
Сами такое придумали?
и нет directmedia=no

Что в externip? Что за девайс делает NAT, что с его SIP ALG?

Re: Нет входящих звонков

Добавлено: 31 янв 2017, 13:29
virus_net
Во-первых, дампы, рулоны логов и т.п. нужно убирать под спойлер. Это элементарное уважение к читающим данный форум.
Во-вторых, вы сами дамп, который выложили, смотрели ? Какие мысли ?
На чем основано мнение:
Legat писал(а):Я так понимаю, что проблема в регистрации?
?
В-третьих,
Legat писал(а):09:44:22.199080 IP (tos 0x60, ttl 64, id 42483, offset 0, flags [none], proto UDP (17), length 532)
10.XXX.XXX.234.5660 > 85.XXX.XXX.218.sip: SIP, length: 504
SIP/2.0 403 Forbidden
Ни на какие мысли не наводит ?
В-четвертых, замазывание внутренних IP-адресов только подчеркивает ваш уровень непонимания.

Посему приговор остается без изменений или дорога в платный суппорт.
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