[Asterisk-Users] ATA call

[Doug Shubert]

Oz,
Is the * server on a "real" Internet IP?

NAT traversal may require you to redirect (PAT)
SIP port 5060 to the inside IP or your ATA 186.

try something like this,

extensions.conf

exten => 3000,1,Dial(SIP/3000,20,tr)
exten => 3000,2,Voicemail,u3000
exten => 3000,102,Voicemail,b3000

sip.conf

[3000]
type=friend
username=3000
secret=xxxx
host=dynamic
mailbox=3000
qualify=200
nat=yes

Doug


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

--
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/