[Asterisk-Users] ATA call

[Eric Wieling]

Your ATA is not registering.  I have a sample ATA-186 at
http://www.fnords.org/~eric/asterisk/


On Tue, 2004-01-06 at 17:41, Osvaldo Mundim Junior wrote:
> Hi Doug,
> 
> I do use the SIP 2.16x on my ATA 186. But I can not see the IP address of my
> ATA on show sip peers. What I can see is:
> 
> Name/username    Host                 Mask             Port     Status
> porto/porto      (Unspecified)   (D)  255.255.255.255  0        UNKNOWN
> 
> Just one thing which I did not mention on the last email is that my ATA is
> behing NAT.
> 
> Oz
> 
> ----- Original Message -----
> From: "Doug Shubert" <doug at accessgate.net>
> To: <asterisk-users at lists.digium.com>
> Sent: Tuesday, January 06, 2004 9:09 AM
> Subject: Re: [Asterisk-Users] ATA call
> 
> 
> > Is your ATA running SIP if so, what version (2.16?)
> >
> > With SIP, then * extensions.conf and sip.conf files are configured
> > you should see the following
> >
> > asterisk3*CLI> sip show peers
> > Name/username    Host                 Mask             Port     Status
> > 3000/3000        10.0.0.30       (D)  255.255.255.255  5060     OK (15 ms)
> > 9000/9000        10.0.0.90       (D)  255.255.255.255  5060     OK (47 ms)
> >
> > ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960
> >
> > to test an extension from the CLI
> > CLI>dial <ext. #>
> > you should hear your ATA ring
> >
> > Doug
> >
> > Osvaldo Mundim Junior wrote:
> >
> > > Hey all!
> > >
> > > I'm having problems trying to set up an ATA 186 with my Asterisk box.
> When I
> > > get the phone to place the call, I type the extension and I only get
> busy
> > > signal after 5 seconds. So I can't call my Asterisk box from my ATA and
> > > either call from my Asterisk to my ATA.
> > >
> > > Does anybody know what can be happing?
> > >
> > > Log is attached..
> > >
> > > tks
> > > regards
> > > Oz
> > >
> >
>   ------------------------------------------------------------------------
> > >                   Name: ast_log.txt
> > >    ast_log.txt    Type: Plain Text (text/plain)
> > >               Encoding: quoted-printable
> >
> > --
> > FREE Unlimited Worldwide Voip calling
> > set-up an account and start saving today!
> > http://www.voippages.com ext. 7000
> > http://www.pulver.com/fwd/ ext. 83740
> > free IP phone software @
> > http://www.xten.com/
> > http://iaxclient.sourceforge.net/iaxcomm/
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Go to http://www.digium.com/index.php?menu=documentation and look at
the "Unofficial Links" section.  This section has links to a wide
variety of 3rd party Asterisk related pages.  My page is the
"Asterisk Resource Pages".

BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643