I think you might want to split your questions first. <div><br></div><div><br></div><div>this might work from local ISP network, but in my experience it might depend on provider.</div><div><br></div><div><span>1. You can't have multiple externip, but it's not necessary to run two Asterisk instances, because you can set routes to different destinations via particular interfaces.</span></div>
<div><span><br></span></div><div><span><br></span></div><div><br></div><div><br><br><div class="gmail_quote">On Thu, Feb 2, 2012 at 2:48 AM, Josh <span dir="ltr"><<a href="mailto:mojo1736@privatedemail.net" target="_blank">mojo1736@privatedemail.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I am trying to configure Asterick, having the following system setup on<br>
the Asterick server:<br>
<br>
* eth0 faces the external Internet interface, *but* it does not have IP<br>
address (it has a private one given to it by my ISP's DHCP server);<br>
* eth1 faces my internal network (say <a href="http://10.1.1.0/24" target="_blank">10.1.1.0/24</a>);<br>
* tun0 serves all mobile smartphones and connects to the internal<br>
network (it has a different ip range, say <a href="http://10.1.2.0/24" target="_blank">10.1.2.0/24</a>) - they are all<br>
connected via the Internet using OpenVPN;<br>
<br>
I would like to configure Asterick for internal calls between ourselves<br>
(eth1<->tun0) and I think I have no problem with configuring this part.<br>
I would also like to use one external VOIP provider to which Asterick<br>
registers on startup. I think I know how to do that and use the<br>
"register" option in sip.conf, though I am not sure for the rest of the<br>
NAT-related entries (see below).<br>
<br>
The purpose of registering this external account is so that both the<br>
smart phones (tun0) and the internal net (eth1) users could use this<br>
account to make external calls (starting with "0", i.e "_0[0-9]."<br>
pattern in extensioins.conf). Obviously, I need these calls to be routed<br>
properly via the external VOIP account. In addition to that, I would<br>
also need to receive calls from that external account to a nominated<br>
internal one (say on extension 20).<br>
<br>
Is this achievable?<br>
<br>
If so, I am not completely clear on whether I need to explicitly specify<br>
my public IP address (via externip/externhost) or whether Asterick is<br>
able to find it without this option? If not, then my plan is to use<br>
external program to find it and then use a script in Asterick to set it<br>
up as an environment variable. Would that work? That external IP address<br>
is going to change, but only in rare circumstances and in such cases I<br>
have to restart a lot of stuff (including Asterick) on that server (this<br>
is usually triggered by a monitoring program), so it won't be a problem<br>
once it is setup initially. I am also not sure whether to specify<br>
"nat=yes" or just have "nat=route" only - any ideas?<br>
<br>
Is there a comprehensive list of all the options available in sip.conf<br>
and what they do, because I was unable to find such a list?<br>
<br>
If the above is doable, I would also like to add the following 2 features:<br>
<br>
1. Secondary external VOIP account, though I have no idea how to specify<br>
its port in "register" (it uses port 5065 instead of the standard 5060).<br>
That account would need to be used on a separate interface (eth2) with a<br>
different public IP address. Would it be possible to use<br>
externip/externhost inside that external account section to specify it?<br>
If this is not possible, then I am thinking of running a separate<br>
instance of Asterick with the second VOIP account/public IP address set<br>
up - would that work?<br>
<br>
2. I would like to be able to configure the following work flow: for a<br>
specific set of (external) calling numbers (including where no Caller ID<br>
is available):<br>
a) these callers to be prompted to specify the "reason" for their call;<br>
b) their response to be temporarily "recorded"/stored (a short message<br>
of, say no more than 10 seconds long or when they press '#' for that<br>
recording to stop);<br>
c) Asterick then rings the nominated number for external VOIP calls<br>
(extension 20) and play that recorded message back;<br>
d) then asks for one of four possible outcomes:<br>
- accept this call (pressing, say 1) in which case the call is connected<br>
as normal;<br>
- reject it with a message that that number/person is "unavailable"<br>
(say, by pressing 0);<br>
- ask the caller to leave a message by transferring them to a voicemail<br>
(say by pressing 2); or<br>
- end the initial call completely with a message that the caller/number<br>
has been "blacklisted" (say, by pressing the 9 key);<br>
<br>
Could this be achieved?<br>
<br>
One final question about binding: in order to be able to use both tun0<br>
and eth1 interfaces so that Asterick serves the calls from both eth1 and<br>
tun0, do I have to use "bind 0.0.0.0"? Is there an alternative, like<br>
specifying "bind 10.1.1.1" for eth1 and then "bind 10.1.2.1" for the<br>
tun0 interface - is this possible?<br>
<br>
Many thanks in advance!<span><font color="#888888"><br>
<br>
<br>
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</font></span></blockquote></div><br><br clear="all"><div><br></div>-- <br>Mvh,<br>Aurimas Skirgaila<br>
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