<font size=2 face="sans-serif">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.</font>
<br><font size=2 face="sans-serif"><br>
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208</font>
<br>
<br>
<br>
<br><font size=1 color=#5f5f5f face="sans-serif">From:
</font><font size=1 face="sans-serif">Frank <frank@efirehouse.com></font>
<br><font size=1 color=#5f5f5f face="sans-serif">To:
</font><font size=1 face="sans-serif">chris@acsdi.com, Asterisk
Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>,
</font>
<br><font size=1 color=#5f5f5f face="sans-serif">Date:
</font><font size=1 face="sans-serif">02/07/2013 12:06 PM</font>
<br><font size=1 color=#5f5f5f face="sans-serif">Subject:
</font><font size=1 face="sans-serif">Re: [asterisk-users]
Asterisk calls between 2 private networks</font>
<br><font size=1 color=#5f5f5f face="sans-serif">Sent by:
</font><font size=1 face="sans-serif">asterisk-users-bounces@lists.digium.com</font>
<br>
<hr noshade>
<br>
<br>
<br><tt><font size=2>I'm using Digium Phones.<br>
I still do not understand why it's not possible to do it the way the <br>
networks are right now.<br>
<br>
If the options I mentioned in my sip.conf are enough, then both phones
<br>
should use Asterisk as a proxy, and Asterisk should handle all the media.<br>
<br>
I will run tcpdump traces tonight and will check it out.<br>
My router has a bug and won't let me mirror port. From tech support I <br>
need to reflash it. I'll do it and try it again.<br>
<br>
F.<br>
<br>
<br>
On 2/7/13 12:59 PM, Christopher Harrington wrote:<br>
> Digium phones, which (as far as I can tell with my experience) do
not<br>
> support VPN yet.<br>
><br>
><br>
> On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen<br>
> <jkillen@allamericanasphalt.com <</font></tt><a href=mailto:jkillen@allamericanasphalt.com><tt><font size=2>mailto:jkillen@allamericanasphalt.com</font></tt></a><tt><font size=2>>><br>
> wrote:<br>
><br>
> Or if it's just a couple phones, you might be able to
setup a vpn<br>
> connection directly on the phone itself - have it vpn
into 'HQ' and<br>
> get an address on that network. I'm not sure which
phones you're<br>
> using though or what phones support that setup.<br>
><br>
> Justin Killen<br>
><br>
> -----Original Message-----<br>
> From: asterisk-users-bounces@lists.digium.com<br>
> <</font></tt><a href="mailto:asterisk-users-bounces@lists.digium.com"><tt><font size=2>mailto:asterisk-users-bounces@lists.digium.com</font></tt></a><tt><font size=2>><br>
> [</font></tt><a href="mailto:asterisk-users-bounces@lists.digium.com"><tt><font size=2>mailto:asterisk-users-bounces@lists.digium.com</font></tt></a><tt><font size=2><br>
> <</font></tt><a href="mailto:asterisk-users-bounces@lists.digium.com"><tt><font size=2>mailto:asterisk-users-bounces@lists.digium.com</font></tt></a><tt><font size=2>>]
On Behalf Of<br>
> Justin Killen<br>
> Sent: Thursday, February 07, 2013 9:55 AM<br>
> To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
> Subject: Re: [asterisk-users] Asterisk calls between
2 private networks<br>
><br>
> I don't see how that would really solve anything - instead
of the<br>
> server sending the 192.168.x.x packets onto the local
network, it<br>
> will send them up toward the internet and get black-holed.
What<br>
> probably makes more sense would be to switch the subnet
on one of<br>
> the networks, AND put up a vpn between them, adding
the routes for<br>
> the private networks to cross thru the tunnels.<br>
><br>
> Justin Killen<br>
> -----Original Message-----<br>
> From: asterisk-users-bounces@lists.digium.com<br>
> <</font></tt><a href="mailto:asterisk-users-bounces@lists.digium.com"><tt><font size=2>mailto:asterisk-users-bounces@lists.digium.com</font></tt></a><tt><font size=2>><br>
> [</font></tt><a href="mailto:asterisk-users-bounces@lists.digium.com"><tt><font size=2>mailto:asterisk-users-bounces@lists.digium.com</font></tt></a><tt><font size=2><br>
> <</font></tt><a href="mailto:asterisk-users-bounces@lists.digium.com"><tt><font size=2>mailto:asterisk-users-bounces@lists.digium.com</font></tt></a><tt><font size=2>>]
On Behalf Of Frank<br>
> Sent: Thursday, February 07, 2013 9:49 AM<br>
> To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
> Cc: Eric Wieling<br>
> Subject: Re: [asterisk-users] Asterisk calls between
2 private networks<br>
><br>
> I thought about that.<br>
> I will give it a shot tonight and will post back my
results in here.<br>
> Thanks<br>
><br>
> On 2/7/13 12:39 PM, Eric Wieling wrote:<br>
> > The easiest thing to is renumber one of the
networks so they are<br>
> not using the same address block.<br>
> ><br>
> > -----Original Message-----<br>
> > From: asterisk-users-bounces@lists.digium.com<br>
> <</font></tt><a href="mailto:asterisk-users-bounces@lists.digium.com"><tt><font size=2>mailto:asterisk-users-bounces@lists.digium.com</font></tt></a><tt><font size=2>><br>
> [</font></tt><a href="mailto:asterisk-users-bounces@lists.digium.com"><tt><font size=2>mailto:asterisk-users-bounces@lists.digium.com</font></tt></a><tt><font size=2><br>
> <</font></tt><a href="mailto:asterisk-users-bounces@lists.digium.com"><tt><font size=2>mailto:asterisk-users-bounces@lists.digium.com</font></tt></a><tt><font size=2>>]
On Behalf Of Frank<br>
> > Sent: Thursday, February 07, 2013 12:27 PM<br>
> > To: Asterisk Users Mailing List - Non-Commercial
Discussion<br>
> > Subject: Re: [asterisk-users] Asterisk calls
between 2 private<br>
> networks<br>
> ><br>
> > AJS,<br>
> ><br>
> > That is a solution that I am envisaging.<br>
> > But I would really love to try to work out
with my issue first.<br>
> It will allow me to deploy more phones in separates
buildlings in<br>
> the future. If I do the IAX solution, it means that
for every<br>
> building, I need a box..<br>
> > Which I would like to prevent.<br>
> ><br>
> ><br>
> ><br>
> > On 2/7/13 10:46 AM, A J Stiles wrote:<br>
> >> On Thursday 07 February 2013, Frank wrote:<br>
> >>> My apologies if this topic was already
discussed in the past.<br>
> >>><br>
> >>> Here is my scenario:<br>
> >>> Network A - 192.168.1.0<br>
> >>> 1 Asterisk<br>
> >>> 1 Digium phone<br>
> >>> Router does NAT from the public IP
to asterisk, and forward ports<br>
> >>> 5060tcp/udp and 10k-20k udp<br>
> >>><br>
> >>> Network B - 192.168.1.0<br>
> >>> 1 Digium phone, registering to the
public IP of network A<br>
> >>><br>
> >>><br>
> >>> My SIP.CONF has:<br>
> >>> nat=yes<br>
> >>> localnet=192.168.1.0/255.255.255.0<br>
> <</font></tt><a href=http://192.168.1.0/255.255.255.0><tt><font size=2>http://192.168.1.0/255.255.255.0</font></tt></a><tt><font size=2>><br>
> >>> externaddr=public_ip_of_network_a<br>
> >>> directmedia=no<br>
> >><br>
> >> My (lazy) solution to this
problem was to throw hardware at it<br>
> .....<br>
> >><br>
> >> Bearing in mind that Asterisk will run
on just about any old<br>
> scrapper<br>
> >> (or even a Raspberry Pi, if you feel
so inclined), there's little<br>
> >> point even trying to send SIP over the
Internet. Just have an<br>
> >> Asterisk box at each end, and then you
only need a much<br>
> simpler-to-configure IAX trunk between the two.<br>
> >> The routers at each end then just need
one port -- UDP 4569 --<br>
> >> forwarded to the Asterisk box (if
it isn't configured as the<br>
> default DMZ machine).<br>
> >><br>
> >><br>
> ><br>
> > --<br>
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><br>
><br>
> --<br>
> -Chris Harrington<br>
> ACSDi Office: 763.559.5800<br>
> Mobile Phone: 612.326.4248<br>
><br>
><br>
><br>
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