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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > 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" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > 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 > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users