[asterisk-users] action at registering or de-registering
Sherwood McGowan
sherwood.mcgowan at gmail.com
Wed Nov 24 15:47:02 CST 2010
On Wed, Nov 24, 2010 at 3:08 PM, Hans Witvliet <hwit at a-domani.nl> wrote:
> On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote:
>> On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
>> a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but
>> should be easy enough to test.
>>
>> Here is an example of what I see on the manager interface during a
>> register/unregister:
>>
>> Event: PeerStatus
>> Privilege: system,all
>> ChannelType: SIP
>> Peer: SIP/twinkle
>> PeerStatus: Registered
>> Address: 192.168.56.1:5068
>>
>> Event: PeerStatus
>> Privilege: system,all
>> ChannelType: SIP
>> Peer: SIP/twinkle
>> PeerStatus: Unregistered
>>
>> I think that should work for whatever you need to do.
>>
>
> I'm doing a fresh install, so 1.8 is what i'm going to use.
>
> What i want to check, is whether to person who is doing a register, is
> realy the person at the other end of a VPN-tunnel.
> With openvpn i'm absolutely sure which person is at a certain
> vpn-ip-addres. I must check if the registering is faked or not.
>
> As ong as linphone (or for that matter any other softphone) does not
> have a possibility for using the libraries from opensc, there is no
> other way...
>
> So next couple of weeks i'll start exploring AMI,
>
> Thanks!
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
Well, if that's all you need (restricting registrations for a SIP
endpoint to a specific IP address), try one of the following
methods...
Method 1:
In the endpoint definition, set the host to the vpn ip address, rather
than setting it to dynamic. This disallows registrations. Then, use
qualify=yes so Asterisk "knows" when the endpoint is available
(responding to OPTIONS requests).
Method 2:
Use the permit,deny, and mask settings to define what ip address
and/or network the endpoint should be at, thereby locking out use from
another address.
(http://www.voip-info.org/wiki/view/Asterisk+sip+permit-deny-mask)
Either of those should resolve your needs
More information about the asterisk-users
mailing list