[Aaug] Delayed ring

Thu Feb 8 17:38:40 EST 2007

On Thu, 2007-02-08 at 17:15 -0500, Bill Learning wrote:
> You have to know it or it wont work *SMILE* if you do know it, it will
> work. 

How so?  Is it a requirement of SIP?

Here is the device I'm currently using:
http://www.voip-info.org/wiki-Packet8+DTA310+and+Asterisk
I'm using that device with the original P8 firmware and I pay them for
service.

I can take that device _anywhere_ and get a line out.  I can be double
natted and it still works no problem.  I do not know how it does this
but based on your comments I will assume it is not using SIP?  My
experience with the device is that the only time it failed me was when
the site's firewall was blocking it.  I would then fire up my laptop
convert it to a router and VPN back into our office.  In effect NAT #1
was my laptop and NAT #2 was our firewall at the office.  I've never
even connected to the DTA310 via a web browser to configure any setting.

Possibly this is what someone was talking about earlier about the
difference between IAX2 and SIP.  Something about IAS2 does not have the
NAT issues as SIP.  Is this what I'm experiencing?  If so would
purchasing the Digium IAX2 adapter allow me to use Asterisk in virtually
any environment even if I'm 2x natted or even 3x natted?

That is what I'm testing.  I'm testing the ability to attach the
Sipura-2000 to _any_ network regardless of design and connect.  In most
cases I'll have no clue what that network looks like and if a firewall
blocks me I can tunnel back to our office.  In that case I'm screwed
because I'll be 2x natted and this device will not work unless I
specifically configure it for that configuration.  Each time I move it I
may have to configure it again?

> That is what the 301 message is. 

301 is the extension I've configured for the ATA
> 
> I think it is page 15 in the manual *LOL* 
> 
> Bill Learning
> Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like
> excess."
> 
> -----Original Message-----
> From: Christopher Fowler [mailto:cfowler at outpostsentinel.com] 
> Sent: Thursday, February 08, 2007 5:11 PM
> To: blearning at speakeasy.net; ATLANTA * USERS GROUP
> Subject: RE: [Aaug] Delayed ring
> 
> On Thu, 2007-02-08 at 16:56 -0500, Bill Learning wrote:
> > If you haven't got it yet
> > 
> > Set the gateway on the ata to the gateway on the DSL not the Router.
> It
> > will pass through correctly on the routers route table.  Nat has to be
> > on both the * server and the ATA, but don't put a NAT address in.  The
> > issue is that it is using the router route table rather then where it
> > should be on the Modem, but really in sip speak your are double
> natted.
> > 
> 
> what if I do not know the gateway.  Read the paragraph below and you'll
> see what I'm trying to test.
> 
> The server has a public address so I've not even considered mucking with
> sip_nat.conf on the server.  I assumed since the server is public and is
> not behind a firewall then I would not have too.
> 
> > 
> > BTW I wouldn't implement this on a Large scale, it would be nasty
> 
> 
> I don't plan on it.  I'm testing what I can do compared to services like
> Vonage.  
> When I travel it may be that I will be double natted.  I will not
> control what I plug into.  
> I get what I can get.
> 
> When I normally travel with my Packet8 adapter I plug it into eth1 on my
> laptop.  I then run MASQ on my laptop with dhcp, named, etc.  Just like
> If my laptop was a router.  In those cases the ATA will be at least
> double natted.  One at the hotel and the second my laptop.
> 
> 
> 
> > 
> > Bill Learning
> > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like
> > excess."
> > 
> > -----Original Message-----
> > From: aaug-bounces at atlaug.com [mailto:aaug-bounces at atlaug.com] On
> Behalf
> > Of Christopher Fowler
> > Sent: Thursday, February 08, 2007 3:48 PM
> > To: ATLANTA * USERS GROUP
> > Subject: Re: [Aaug] Delayed ring
> > 
> > I see this in the PBX log:
> > 
> > Feb  8 15:44:43 NOTICE[2831] chan_sip.c: Peer '301' is now
> UNREACHABLE!
> > Last qualify: 69
> > 
> > 
> > On Thu, 2007-02-08 at 15:30 -0500, Christopher Fowler wrote:
> > > I've plugged my ATA into a Linksys 802.11B router.  The router is
> > > plugged into my home network.  So to make a call the ATA needs to go
> > out
> > > of 2 firewalls.  The Linksys and then my Linux routers.
> > > 
> > > When I call the DID line and dial 301 to get my extension I
> > immediately
> > > go into voice mail.  I get a message that "The person at extension
> 301
> > > is on the phone".  My ATA does not ring.  About 60(s) later after
> I've
> > > hung up it rings.  When I answer it I only hear a dial tone.
> Trixbox
> > > panel is showing the phone on hook.  Maybe dual firewalls is causing
> > an
> > > issue?  
> > > 
> > > _______________________________________________
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> > 
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> > 
> > 
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> 
> 