[asterisk-users] Is this doable?

[C F]

Whats asterick?

On Wed, Feb 1, 2012 at 7:48 PM, Josh <mojo1736 at privatedemail.net> wrote:
> I am trying to configure Asterick, having the following system setup on
> the Asterick server:
>
> * eth0 faces the external Internet interface, *but* it does not have IP
> address (it has a private one given to it by my ISP's DHCP server);
> * eth1 faces my internal network (say 10.1.1.0/24);
> * tun0 serves all mobile smartphones and connects to the internal
> network (it has a different ip range, say 10.1.2.0/24) - they are all
> connected via the Internet using OpenVPN;
>
> I would like to configure Asterick for internal calls between ourselves
> (eth1<->tun0) and I think I have no problem with configuring this part.
> I would also like to use one external VOIP provider to which Asterick
> registers on startup. I think I know how to do that and use the
> "register" option in sip.conf, though I am not sure for the rest of the
> NAT-related entries (see below).
>
> The purpose of registering this external account is so that both the
> smart phones (tun0) and the internal net (eth1) users could use this
> account to make external calls (starting with "0", i.e "_0[0-9]."
> pattern in extensioins.conf). Obviously, I need these calls to be routed
> properly via the external VOIP account. In addition to that, I would
> also need to receive calls from that external account to a nominated
> internal one (say on extension 20).
>
> Is this achievable?
>
> If so, I am not completely clear on whether I need to explicitly specify
> my public IP address (via externip/externhost) or whether Asterick is
> able to find it without this option? If not, then my plan is to use
> external program to find it and then use a script in Asterick to set it
> up as an environment variable. Would that work? That external IP address
> is going to change, but only in rare circumstances and in such cases I
> have to restart a lot of stuff (including Asterick) on that server (this
> is usually triggered by a monitoring program), so it won't be a problem
> once it is setup initially. I am also not sure whether to specify
> "nat=yes" or just have "nat=route" only - any ideas?
>
> Is there a comprehensive list of all the options available in sip.conf
> and what they do, because I was unable to find such a list?
>
> If the above is doable, I would also like to add the following 2 features:
>
> 1. Secondary external VOIP account, though I have no idea how to specify
> its port in "register" (it uses port 5065 instead of the standard 5060).
> That account would need to be used on a separate interface (eth2) with a
> different public IP address. Would it be possible to use
> externip/externhost inside that external account section to specify it?
> If this is not possible, then I am thinking of running a separate
> instance of Asterick with the second VOIP account/public IP address set
> up - would that work?
>
> 2. I would like to be able to configure the following work flow: for a
> specific set of (external) calling numbers (including where no Caller ID
> is available):
> a) these callers to be prompted to specify the "reason" for their call;
> b) their response to be temporarily "recorded"/stored (a short message
> of, say no more than 10 seconds long or when they press '#' for that
> recording to stop);
> c) Asterick then rings the nominated number for external VOIP calls
> (extension 20) and play that recorded message back;
> d) then asks for one of four possible outcomes:
> - accept this call (pressing, say 1) in which case the call is connected
> as normal;
> - reject it with a message that that number/person is "unavailable"
> (say, by pressing 0);
> - ask the caller to leave a message by transferring them to a voicemail
> (say by pressing 2); or
> - end the initial call completely with a message that the caller/number
> has been "blacklisted" (say, by pressing the 9 key);
>
> Could this be achieved?
>
> One final question about binding: in order to be able to use both tun0
> and eth1 interfaces so that Asterick serves the calls from both eth1 and
> tun0, do I have to use "bind 0.0.0.0"? Is there an alternative, like
> specifying "bind 10.1.1.1" for eth1 and then "bind 10.1.2.1" for the
> tun0 interface - is this possible?
>
> Many thanks in advance!
>
>
> --
> _____________________________________________________________________
> -- 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