[asterisk-users] Asterisk calls between 2 private networks

Frank frank at efirehouse.com
Thu Feb 7 12:22:52 CST 2013


i think canreinvite is not part of Asterisk 1.8 anymore.

Asterisk 1.8 added directmediapermit and directmediadeny to limit which 
peers can send direct media to each other.

On 2/7/13 1:15 PM, Kevin Larsen wrote:
> Did you set canreinvite=no in sip.conf on the phone in network B? A
> phone that can connect but loses audio is almost a sure sign that it is
> reinviting and your rtp packets are not making it to the phone. By
> turning canreinvite off, it will keep asterisk in the middle of your
> sessions and should give you the audio.
>
> Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
>
>
>
> From: Frank <frank at efirehouse.com>
> To: chris at acsdi.com, Asterisk Users Mailing List - Non-Commercial
> Discussion <asterisk-users at lists.digium.com>,
> Date: 02/07/2013 12:06 PM
> Subject: Re: [asterisk-users] Asterisk calls between 2 private networks
> Sent by: asterisk-users-bounces at lists.digium.com
> ------------------------------------------------------------------------
>
>
>
> 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
>  > <jkillen at allamericanasphalt.com <mailto:jkillen at allamericanasphalt.com>>
>  > 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: asterisk-users-bounces at lists.digium.com
>  >     <mailto:asterisk-users-bounces at lists.digium.com>
>  >     [mailto:asterisk-users-bounces at lists.digium.com
>  >     <mailto:asterisk-users-bounces at lists.digium.com>] 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: asterisk-users-bounces at lists.digium.com
>  >     <mailto:asterisk-users-bounces at lists.digium.com>
>  >     [mailto:asterisk-users-bounces at lists.digium.com
>  >     <mailto:asterisk-users-bounces at lists.digium.com>] 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: asterisk-users-bounces at lists.digium.com
>  >     <mailto:asterisk-users-bounces at lists.digium.com>
>  >     [mailto:asterisk-users-bounces at lists.digium.com
>  >     <mailto:asterisk-users-bounces at lists.digium.com>] 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).
>  >      >>
>  >      >>
>  >      >
>  >      > --
>  >      >
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