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Well, a SIP authorization does not require a registration (in
fact, registration should be primarily used to inform a registrar about
the whereabouts of a UA with dynamic IP address in order to handle incoming
calls _for_ that UA).
CS can just create for his Asterisk a "type=user"
entry in sip.conf containing "username" (equal to the section's title) and
"secret" both matching the remote peer's own: his Asterisk will then react
to an INVITE from that peer with a "401" reply containing a nonce
as challenge; the peer will then retry the INVITE with valid credentials
based on the shared secret and the nonce.
Enzo
----- Original Message -----
Sent: Wednesday, September 14, 2005 2:49
PM
Subject: Re: [Asterisk-Users] Anyone
knows how to receive a SIP call withoutregistering gateway?
What they're asking you to do is quite insecure to be doing over
public IP. At the very least, you should confirm that there is a static IP
that these calls will be coming from and only accept calls from that IP, but
that's still not quite as secure as digest authentication that would be
available via registration.
If you know what IP the calls are coming from, you simply insert a
host=XX.XX.XX.XX instead of host=dynamic in your sip.conf for that peer and
calls should then come in as they did before without them having to register.
If they are pre-pending digits on to the front of what you're interpreting as
the dialed number/extension, you may choose to lop them off in
extensions.conf, but aside from that this is fairly straight
forward.
On 9/14/05, C.
Savinovich <[EMAIL PROTECTED]>
wrote:
Hello
everyone, I am pulling my hair here because a carrier threw me curve early
today.
They want to send calls to my asterisk server
using SIP. Then they said that their gateways don't have to
register with my server, that all they have to do is send a prefix for
validation. Whereas I can think of several ways to authenticate
their incoming number string, I am only used to the orthodox SIP way which
is: client registers to my proxy. Guess what, I can't find any
samples on this!!, Can anyone please help?, I will probably need a sample
sip.conf. and then, to make a test call, I can use another
asterisk box and try asterisk to asterisk sip calls (without register) via
the cli prompt. But I have no idea.... and I am
intrigued.
Thanks
CS
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