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Откуда: Ярославль
Сообщений: 21
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Asterisk+NAT+SIPNET
настроил транк
[NEWsipnet]
disallow=all
fromdomain=sipnet.ru
host=sipnet.ru
port=5060
secret=****
type=friend
username=0023***
fromuser=0023***
canreinvite=no
nat=yes
qualify=500
dtmfmode=rfc2833
insecure=port,invite
reinvite=no
allow=ulaw
allow=alaw
conext=from-internal
Все работает тока периодически, если позвонишь из сипнета, то транк через некоторое время поднимается, если не звонить то он перестает региться, что самое примечательное он регистрируется на сипнете с портом 1024
т.е. 0023***@80.255.179.186:1024, хотя у меня явно прописан порт 5060 в транке и в sip.conf, кстати вот он:
[general]
language=ru
nat=yes
canreinvite=no
natip=Внешний_IP
externip=Внешний_IP
localnet=192.168.0.0/255.255.0.0
maxexpiry=3600
minexpiry=60
defaultexpiry=270
externrefresh=30
port=5060
realm=(DNS-имя)
команда
sip set debug peer NEWsipnet
---
[Nov 27 08:35:15] Retransmitting #1 (NAT) to 212.53.40.40:5060:
OPTIONS sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 80.255.179.186:5060;branch=z9hG4bK431a1031;rport
From: "Unknown" <sip:Unknown@80.255.179.186>;tag=as4d9890f8
To: <sip:sipnet.ru>
Contact: <sip:Unknown@80.255.179.186>
Call-ID: 2868132b68fc0de00e61feaf52b70d11@80.255.179.186
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 27 Nov 2009 05:35:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Nov 27 08:35:15] Really destroying SIP dialog '2868132b68fc0de00e61feaf52b70d11@80.255.179.186' Method: OPTIONS
[Nov 27 08:35:15] Reliably Transmitting (NAT) to 212.53.40.40:5060:
OPTIONS sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 80.255.179.186:5060;branch=z9hG4bK79ca2ac3;rport
From: "Unknown" <sip:Unknown@80.255.179.186>;tag=as074e7f43
To: <sip:sipnet.ru>
Contact: <sip:Unknown@80.255.179.186>
Call-ID: 0737801a6dabf4d90dc9254306dd3e31@80.255.179.186
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 27 Nov 2009 05:35:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Nov 27 08:35:15] REGISTER 12 headers, 0 lines
[Nov 27 08:35:15] Reliably Transmitting (NAT) to 212.53.40.40:5060:
REGISTER sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 80.255.179.186:5060;branch=z9hG4bK77ae0c90;rport
From: <sip:0023904231@sipnet.ru>;tag=as71644e0b
To: <sip:0023904231@sipnet.ru>
Call-ID: 02fb96f706f2e83371222d2460002cff@127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 270
Contact: <sip:0023904231@80.255.179.186>
Event: registration
Content-Length: 0
---
[Nov 27 08:35:15] Retransmitting #1 (NAT) to 212.53.40.40:5060:
OPTIONS sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 80.255.179.186:5060;branch=z9hG4bK535ca124;rport
From: "Unknown" <sip:Unknown@80.255.179.186>;tag=as15c00166
To: <sip:sipnet.ru>
Contact: <sip:Unknown@80.255.179.186>
Call-ID: 20dcdbca42cf97b0437f8dc3008c5955@80.255.179.186
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 27 Nov 2009 05:35:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Nov 27 08:35:15] Really destroying SIP dialog '20dcdbca42cf97b0437f8dc3008c5955@80.255.179.186' Method: OPTIONS
[Nov 27 08:35:15] Reliably Transmitting (NAT) to 212.53.40.40:5060:
OPTIONS sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 80.255.179.186:5060;branch=z9hG4bK0a575fb5;rport
From: "Unknown" <sip:Unknown@80.255.179.186>;tag=as0c8b62aa
To: <sip:sipnet.ru>
Contact: <sip:Unknown@80.255.179.186>
Call-ID: 62ace4d330fe054051dae10b14aa78bb@80.255.179.186
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 27 Nov 2009 05:35:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
[Nov 27 08:35:16] Retransmitting #1 (NAT) to 212.53.40.40:5060:
REGISTER sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 80.255.179.186:5060;branch=z9hG4bK09e11baf;rport
From: <sip:0024184301@sipnet.ru>;tag=as37a733fa
To: <sip:0024184301@sipnet.ru>
Call-ID: 21354baa0c03c8e20cdf04387916ce19@127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 270
Contact: <sip:0024184301@80.255.179.186>
Event: registration
Content-Length: 0
---
[Nov 27 08:35:16] Retransmitting #1 (NAT) to 212.53.40.40:5060:
OPTIONS sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 80.255.179.186:5060;branch=z9hG4bK79ca2ac3;rport
From: "Unknown" <sip:Unknown@80.255.179.186>;tag=as074e7f43
To: <sip:sipnet.ru>
Contact: <sip:Unknown@80.255.179.186>
Call-ID: 0737801a6dabf4d90dc9254306dd3e31@80.255.179.186
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 27 Nov 2009 05:35:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Да и еще очень раздрожжает " <sip:Unknown@" - я думаю это связано с тем что не может зарегистрироваться.
Помогите пожалуйста, куда крутить не знаю мозг уже вспух. В файерволе для астериска все открыто, а вот почем он не хочет регится?, помогите дельным советом...
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