Beim Versuch den Asterisk bei 1&1 anzumelden, wird wegen NAT die interne DSL-IP der Fritzbox 192.168.179.1 übertragen, und dies lässt 1&1 (natürlich) nicht zu.
Laut http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions ist dies prinzipiell nicht möglich, da der Asterisk über das NAT der Fritzbox ins Internet geht, und sich der Asterisk nicht über NAT an einem SIP-Server anmelden kann (Fall 1).
Ist das wirklich wahr? Oder gibt es einen Trick, es doch schaffen? Hat jemand das schon erfolgreich hinbekommen? Geht das eventuell mit einem SIP-Proxy auf der Fritzbox?
Jörg
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 212.227.15.197:5060:
REGISTER sip:sip.1und1.de SIP/2.0
Via: SIP/2.0/UDP 192.168.179.1:5061;branch=z9hG4bK13aeff4f;rport
From: <sip:[email protected]>;tag=as03c05cd0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REGISTER
User-Agent: ASTERISK_JR
Max-Forwards: 70
Expires: 120
Contact: <sip:[email protected]:5061>
Event: registration
Content-Length: 0
<-- SIP read from 212.227.15.197:5060:
SIP/2.0 403 Keine RFC1918-IPs erlaubt
Via: SIP/2.0/UDP 192.168.179.1:5061;received=84.191.34.238;branch=z9hG4bK13aeff4f;rport=5061
From: <sip:[email protected]>;tag=as03c05cd0
To: <sip:[email protected]>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8131
Call-ID: [email protected]
CSeq: 102 REGISTER
Server: UI OpenSer
Content-Length: 0
Jan 6 21:42:47 WARNING[1768]: chan_sip.c:9571 handle_response_register: Forbidden - wrong password on authentication for REGISTER for '4930xxxxxxxx' to 'sip.1und1.de'
Laut http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions ist dies prinzipiell nicht möglich, da der Asterisk über das NAT der Fritzbox ins Internet geht, und sich der Asterisk nicht über NAT an einem SIP-Server anmelden kann (Fall 1).
Asterisk, SIP and NAT
Asterisk can both act as a SIP client and a SIP server. Asterisk as a SIP client is configured with a register=> line in the [general] section of sip.conf. Asterisk as a SIP server connects clients (SIP Phones) configured with their own username, secret and other details in client sections of sip.conf.
Asterisk SIP channels in a NATed network can be generalized like this:
1. Asterisk as a SIP client behind nat, connecting to outside SIP Proxies
2. Asterisk as a SIP client behind nat, connecting to inside SIP proxies
3. Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk
4. Asterisk as a SIP server behind nat, clients on the outside behind a second NAT connecting to Asterisk
5. Asterisk as a SIP server behind nat, clients on the inside connecting to Asterisk
6. Asterisk as a SIP client outside nat, connecting to outside SIP proxies
7. Asterisk as a SIP client outside nat, connecting to inside SIP proxies
8. Asterisk as a SIP server outside nat, clients on the outside connecting to Asterisk
9. Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk
Every setup works somewhere, but it depends on the client, the NAT, the server and many other factors. In most cases, 1 and 3 is broken. SIP is a peer-to-peer protocol and a NAT can be generalized and simplified as a solution that allows clients on the inside to connect to servers on the outside and _not_ allow clients on the outside to connect to any server on the inside.
Ist das wirklich wahr? Oder gibt es einen Trick, es doch schaffen? Hat jemand das schon erfolgreich hinbekommen? Geht das eventuell mit einem SIP-Proxy auf der Fritzbox?
Jörg
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