[Asterisk-Users] ATA call

[Eric Wieling]

I have ZERO problems with Cisco's NAT for SIP.

On Tue, 2004-01-06 at 13:42, CW_ASN - Gus wrote:
> Sometimes Cisco nat changes the port, and * can't contact to ATA. I see that
> behavior some days ago, and I can't resolve that. :(
> 
> Regards,
> 
> Gus
> 
> ----- Original Message -----
> From: "Osvaldo Mundim Junior" <osvaldo at ilinksolutions.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Tuesday, January 06, 2004 9:15 PM
> Subject: Re: [Asterisk-Users] ATA call
> 
> 
> > Some times the "sip show peers" shows me:
> > Name/username    Host                 Mask             Port     Status
> > porto/porto      (Unspecified)   (D)  255.255.255.255  0        UNKNOWN
> >
> >
> > and some times shows me:
> >
> > Name/username    Host                 Mask             Port     Status
> > porto/porto      200.167.103.219 (D)  255.255.255.255  1025     LAGGED
> (815
> > ms)
> >
> > Does the port supposed to be 5060?
> >
> > 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
> > >
> >
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> 
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-- 
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".

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