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 <fr...@efirehouse.com>
To: ch...@acsdi.com, Asterisk Users Mailing List - Non-Commercial
Discussion <asterisk-users@lists.digium.com>,
Date: 02/07/2013 12:06 PM
Subject: Re: [asterisk-users] Asterisk calls between 2 private networks
Sent by: asterisk-users-boun...@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
> <jkil...@allamericanasphalt.com
<mailto:jkil...@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-boun...@lists.digium.com
> <mailto:asterisk-users-boun...@lists.digium.com>
> [mailto:asterisk-users-boun...@lists.digium.com
> <mailto:asterisk-users-boun...@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-boun...@lists.digium.com
> <mailto:asterisk-users-boun...@lists.digium.com>
> [mailto:asterisk-users-boun...@lists.digium.com
> <mailto:asterisk-users-boun...@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-boun...@lists.digium.com
> <mailto:asterisk-users-boun...@lists.digium.com>
> [mailto:asterisk-users-boun...@lists.digium.com
> <mailto:asterisk-users-boun...@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).
> >>
> >>
> >
> > --
> >
_____________________________________________________________________
> > -- Bandwidth and Colocation Provided by
> http://www.api-digital.com <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 <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 <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 <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 <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
<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
<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
