>
> At the risk of sounding stupid. what's CSU/DSU ? *i'm googling it
> right now, but it's nice to have convo on the list!*
A CSU/DSU is Channel Service Unit (CSU) this terminates T1 connections
from the phone company. This information is then passed to the Data
Service Unit which turns th
I am new to * and I have been attempting to solve this same issue, but
have come to the conclusion that they only way to make it work is for *
to have a real reachable IP address or place another * box at the second
site and use IAX trunking. This second * box, unfortunately is
unsuitable for my sc
> ClientServer
>
> XTEN <--> */Firewall(NAT) <---IAX---> Firewall(NAT)/*
>
If you are going to use IAX, I don't think you have to put * on the
firewall boxes, only if you wish to use SIP.
Steve
___
Asterisk-Users mail
t; For IAX, just open up the IAX ports on the firewall (the exact
> numbers escape me right at the moment), and let it fly?
>
> -cj
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Stephen Varga
Try this link:
http://gnophone.com/directory/createAccount.php
You will find it on the setup page. The first line says: To sign up for
free IAXtel access, go here.
The word 'here' is the hyperlink to account creation form.
As for top posting, it is what I just did here.
You should also setup y
On Sat, 2003-09-20 at 11:48, Steve Totaro wrote:
> routers or a t1 cards can do that without a csu/dsu. a csu/dsu is really
> just something that the telco can use for trouble shooting and loopback
> testing. i think the fcc also requires it by law.
First off CSU/DSU is not required by FCC law
Unfortunetly this setup does not work, when * sends SDP info in the
INVITE process on how to establish the audio session *'s real IP address
is in the packet and the outside phone tries to connect to this IP
address, which of course is unreachable because of the firewall. For
this to work you need
it work in your setup. Let me
know.
Steve
> Brad
>
>
> Stephen Varga wrote:
>
> > Unfortunetly this setup does not work, when * sends SDP info in the
> > INVITE process on how to establish the audio session *'s real IP address
> > is in the packet and the outside phon
On Mon, 2003-09-22 at 13:51, Bartosz Jozwiak wrote:
>
> When I call directly to X-Lite from ATA it doesn't work but when I
> call to X-lie with queue from ATA is works...
> It is really strange.
Try adding 'canreinvite=no' to the sip definition for both the X-lite
phone and the ATA.
__
On Tue, 2003-09-23 at 09:44, Bartosz Jozwiak wrote:
> Right now it works great!
> Thanks so much.
>
> Could you tell me what is that:
> 'canreinvite=no' in sip.conf ?
>
When SIP initiates the call, the INVITE message contains the information
on where to send the media streams. * uses itself as
On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote:
> Adam:
> in reference to my first message, the NAT on the SIP/GS (a D-Link router)
> has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being
> forwarded to the Sip/GS.
> The Asterisk server, also behind another NAT (Linksys), has
On Thu, 2003-09-25 at 10:42, Michael Koehler wrote:
> Sorry, but my * is behind NAT and i have no problems with SIP, and it
> even works with NAT to NAT and without forwarding ports or similar
> effords.
>
>
> Michael
What kinda box/device is doing the NAT?
___
On Thu, 2003-09-25 at 12:54, Michael Koehler wrote:
> A plain wireless dlink dsl router.
Do you know the model number and the software version?
I am trying to understand how it is making the appropriate adjustments
to allow the connection to work.
Thanks,
Steve
On Thu, 2003-09-25 at 15:41, Michael Koehler wrote:
> It is not a feature of the router, it is the way SIP is handled with
> nikotel.com
>
> I recently wrote that i'm using just a plain router with my natted
> asterisk because "Stephen Varga" wrote that SIP behind
On Monday 08 March 2004 07:31 pm, Brian Capouch wrote:
> Well, maybe. The Grandstreams use an extended version, and at least
> right now, the hpa tftpd does NOT work with them. I spent many hours
> playing around last night, and the GS phones for whatever reason will
> not download files that ar
On Thursday 11 March 2004 02:39 am, admin wrote:
> Ping times (latency) and bandwidth are really not related unless you are
> filling the pipe. Your ping times are too high. My understanding is that
> anything over 100ms is not good. Your problem probably lies with too many
> hops or slow or ove
On Friday 12 March 2004 09:28 pm, AstGrp wrote:
> Do I need to associate the outside interface of the PIX with the phone
> on the inside.. I don't remember doing this before...
>
> Setup
>
> * Server ---> PIX FW ---> WWW CLOUD > PIX FW ---> IP Phone
>
> Again the only difference than befor
On Saturday 13 March 2004 08:21 pm, AstGrp wrote:
> Thank you... I found that document last night.. And I have the pix
> configured this way with fixup sip... But still no go.. I am going to
> try and upgrade the pix tonight and see if that helps.
>
The only suggestion I have now is to start doing
18 matches
Mail list logo
