I'm using Digium Phones.
I still do not understand why it's not possible to do it the way the
networks are right now.
If the options I mentioned in my sip.conf are enough, then both phones
should use Asterisk as a proxy, and Asterisk should handle all the media.
I will run tcpdump traces tonight and will check it out.
My router has a bug and won't let me mirror port. From tech support I
need to reflash it. I'll do it and try it again.
F.
On 2/7/13 12:59 PM, Christopher Harrington wrote:
Digium phones, which (as far as I can tell with my experience) do not
support VPN yet.
On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen
<[email protected] <mailto:[email protected]>>
wrote:
Or if it's just a couple phones, you might be able to setup a vpn
connection directly on the phone itself - have it vpn into 'HQ' and
get an address on that network. I'm not sure which phones you're
using though or what phones support that setup.
Justin Killen
-----Original Message-----
From: [email protected]
<mailto:[email protected]>
[mailto:[email protected]
<mailto:[email protected]>] On Behalf Of
Justin Killen
Sent: Thursday, February 07, 2013 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk calls between 2 private networks
I don't see how that would really solve anything - instead of the
server sending the 192.168.x.x packets onto the local network, it
will send them up toward the internet and get black-holed. What
probably makes more sense would be to switch the subnet on one of
the networks, AND put up a vpn between them, adding the routes for
the private networks to cross thru the tunnels.
Justin Killen
-----Original Message-----
From: [email protected]
<mailto:[email protected]>
[mailto:[email protected]
<mailto:[email protected]>] On Behalf Of Frank
Sent: Thursday, February 07, 2013 9:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Eric Wieling
Subject: Re: [asterisk-users] Asterisk calls between 2 private networks
I thought about that.
I will give it a shot tonight and will post back my results in here.
Thanks
On 2/7/13 12:39 PM, Eric Wieling wrote:
> The easiest thing to is renumber one of the networks so they are
not using the same address block.
>
> -----Original Message-----
> From: [email protected]
<mailto:[email protected]>
[mailto:[email protected]
<mailto:[email protected]>] On Behalf Of Frank
> Sent: Thursday, February 07, 2013 12:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk calls between 2 private
networks
>
> AJS,
>
> That is a solution that I am envisaging.
> But I would really love to try to work out with my issue first.
It will allow me to deploy more phones in separates buildlings in
the future. If I do the IAX solution, it means that for every
building, I need a box..
> Which I would like to prevent.
>
>
>
> On 2/7/13 10:46 AM, A J Stiles wrote:
>> On Thursday 07 February 2013, Frank wrote:
>>> My apologies if this topic was already discussed in the past.
>>>
>>> Here is my scenario:
>>> Network A - 192.168.1.0
>>> 1 Asterisk
>>> 1 Digium phone
>>> Router does NAT from the public IP to asterisk, and forward ports
>>> 5060tcp/udp and 10k-20k udp
>>>
>>> Network B - 192.168.1.0
>>> 1 Digium phone, registering to the public IP of network A
>>>
>>>
>>> My SIP.CONF has:
>>> nat=yes
>>> localnet=192.168.1.0/255.255.255.0
<http://192.168.1.0/255.255.255.0>
>>> externaddr=public_ip_of_network_a
>>> directmedia=no
>>
>> My (lazy) solution to this problem was to throw hardware at it
.....
>>
>> Bearing in mind that Asterisk will run on just about any old
scrapper
>> (or even a Raspberry Pi, if you feel so inclined), there's little
>> point even trying to send SIP over the Internet. Just have an
>> Asterisk box at each end, and then you only need a much
simpler-to-configure IAX trunk between the two.
>> The routers at each end then just need one port -- UDP 4569 --
>> forwarded to the Asterisk box (if it isn't configured as the
default DMZ machine).
>>
>>
>
> --
> _____________________________________________________________________
> -- 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
>
--
_____________________________________________________________________
-- 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
--
_____________________________________________________________________
-- 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
--
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
--
_____________________________________________________________________
-- 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
