Tom Rymes
2005-Nov-26 12:25 UTC
[Asterisk-Users] Re: SIP Extension behind NAT, Asterisk on NAT (DMZ)
On Nov 26, 2005, at 12:22 PM, Manny A. Wise wrote:>> On Nov 25, 2005, at 7:00 PM, Manny A. Wise wrote: >> >>> Great!!, this did the trick, now we have audio... >>> We are using a Sipura 2000 for testing.... >>> The Sipura now can call out and have audio...the only problem left >>> is that >>> the sipura can't receive calls, when the extension is dialed, the >>> recording >>> says, the person is on the phone.....any ideas??? >>> >>> I changed the externip=, localnet= and nat=yes in sip.com and in the >>> extension setup in amp nat=1...... missing anything???? >>> >>> THANKS!!!!!!!!!!!!!!!! >>> >>> Manny >> >> -----Original Message----- >> From: Tom Rymes [mailto:trymes@cascadelinksystems.com] >> Sent: Friday, November 25, 2005 10:48 PM >> To: Manny A.Wise >> Subject: Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk >> on a public >> domain >> >> It sounds as if your extension isn't registered. Make sure that the >> extension is configured as dynamic in sip.conf (or AMP) and as >> nat=yes. Also, make sure that the Sipura is configured through its >> web interface to register and it has the right user and password >> entered. Once this is done, when you type 'sip show peers' from the >> CLI your Sipura's extension should be listed, and show a 'D' and an >> 'N' for dynamic and nat. >> >> Also, it sounds like you are using AMp and or A@H, so make sure that >> you put the nat, externip, and localnet parameters in the >> sip_nat.conf file, *NOT* the sip.conf file, as that is likely to get >> overwritten by AMP. From my installation (obviously, substitute your >> external IP for the xxx.xxx.xxx.xxx below...): >> >> [root@mercury root]# cat /etc/asterisk/sip_nat.conf >> nat=yes >> externip=xxx.xxx.xxx.xxx >> localnet=10.0.0.0/255.255.255.0 >> >> Other than that, I recommend further google and voip-info spelunking >> expeditions to track down your problem. > > Tom > We found the trouble... > What is happening is that my friend connection is changing IP > address way to > often.....he is on a DSL line in Peru...and is chaging when we less > expect > it, now that we know, we are monitoring it.... > Anyway...the problem left is that if we use the IP, it work > perfectly both > ways ...but if we use his name.dyndns.org name audio doesn't work... > I though that externip=201.240.220.123 or externip=name.dyndns.org > Should do the same work.....? > The dyndns.org support is build on his Ztxel modem, and so far I > never had a > single problem accessing the site by webrowser... > > Any thought??Manny, Two things: 1.) please send your replies to the list and not to me personally! That way the discussion ends up in the list archives and will be there to help the next person who has this problem. 2.) I am not certain if the externip= option can take a FQDN as an argument. The page I looked at on the wiki (see below) was not clear on this. I suggest using trial and error to see if works and then update the wiki page once you know. 3.) If your external IP is constantly changing, I searched for "site:voip-info.org externip" on Google and found this page on the wiki: http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+externip which has a link to this message in the list archives: http://lists.digium.com/pipermail/asterisk-users/2004-November/ 071349.html which discusses this problem. Basically, every time that your IP address changes you need to update the IP with dyndns, and you need to change the externip= option. Anyhow, if your address changes and you do not update the dyndns service, the SIP client will be looking for the wrong IP (the old IP.). HTH, Tom -------------------- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 "Intelligent technology solutions for small businesses."