[asterisk-users] Asterisk calls between 2 private networks
Frank
frank at efirehouse.com
Thu Feb 7 12:25:09 CST 2013
And actually I did not have directmediadeny=0.0.0.0
But I had directmedia=no.
So I will add the directmediadeny line, and will check it out again
tonight.
On 2/7/13 1:22 PM, Frank wrote:
> 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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>> >
>> >
>> > --
>> > -Chris Harrington
>> > ACSDi Office: 763.559.5800
>> > Mobile Phone: 612.326.4248
>> >
>> >
>> >
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