Whats asterick? On Wed, Feb 1, 2012 at 7:48 PM, Josh <[email protected]> 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
-- _____________________________________________________________________ -- 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
