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:[EMAIL PROTECTED]
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 [EMAIL PROTECTED], 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...):
[EMAIL PROTECTED] 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."
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