[Asterisk-Users] ATA call
CW_ASN - Gus
cw_asn at fibertel.com.ar
Tue Jan 6 12:42:16 MST 2004
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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