Dear Mr. Newton
Thank you for your response. I red the wiki and sip.conf sample file and I
know about directmedia option. Actually these options are for times that
you know about your connected networks (you know which clients are behind
NAT and which are not). But my configuration is different. I
On Fri, Feb 21, 2014 at 11:13 AM, Gholamreza Sabery gr.sab...@gmail.com wrote:
Dear Mr. Newton
Thank you for your response. I red the wiki and sip.conf sample file and I
know about directmedia option. Actually these options are for times that you
know about your connected networks (you know
Anyway, thank you so much. ;-)
On Fri, Feb 21, 2014 at 9:32 PM, Rusty Newton rnew...@digium.com wrote:
On Fri, Feb 21, 2014 at 11:13 AM, Gholamreza Sabery gr.sab...@gmail.com
wrote:
Dear Mr. Newton
Thank you for your response. I red the wiki and sip.conf sample file and
I
know about
On Tue, Feb 18, 2014 at 10:53 PM, Gholamreza Sabery gr.sab...@gmail.com wrote:
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know is it possible or not?
Hi! As many others mentioned,
On Wed, Feb 19, 2014 at 2:55 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Wednesday 19 Feb 2014, Gholamreza Sabery wrote:
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know
On 19/2/14 4:53 am, Gholamreza Sabery wrote:
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know is it possible or not?
I can't help on the can Asterisk detect they're behind the same
On Wednesday 19 Feb 2014, Gholamreza Sabery wrote:
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know is it possible or not?
There is a bit of a tendency on this list to ignore
Anyway Thank you guys. ;-)
On Wed, Feb 19, 2014 at 12:25 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Wednesday 19 Feb 2014, Gholamreza Sabery wrote:
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know is it possible or not?
--
_
-- Bandwidth and Colocation Provided by
On Wed, 19 Feb 2014, Gholamreza Sabery wrote:
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know is it possible or not?
If it were only so easy...
Participation in these lists is
I have an Asterisk box with a public IP address and two SIP clients behind
the same NAT device(I also have SIP clients behind different NATs). I want
to know is it possible for Asterisk to detect if both clients are behind
the same NAT and use direct media between them and use other options for
On 7 January 2014 22:55, Adam Moffett adamli...@plexicomm.net wrote:
I'm asking about this scenario:
Asterisk(public IP) -- Internet -- Router (public IP) -- SIP client
(private IP and NAT)
What settings in sip.conf will give this the best fighting chance of
working?
We already have
On 1/8/2014 4:17 AM, Ishfaq Malik wrote:
On 7 January 2014 22:55, Adam Moffett adamli...@plexicomm.net
mailto:adamli...@plexicomm.net wrote:
I'm asking about this scenario:
Asterisk(public IP) -- Internet -- Router (public IP) -- SIP
client (private IP and NAT)
What
I'm asking about this scenario:
Asterisk(public IP) -- Internet -- Router (public IP) -- SIP
client (private IP and NAT)
What settings in sip.conf will give this the best fighting chance of
working?
We already have nat=force_rport,comedia
--
Hello
I think I finally understood the issue/solution, but I'd like to make
sure I'm correct:
- In Diana Cionoiu's famous article on Freshmeat
(http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol),
regardless of whether SIP end-points use a public IP or are behind a
NAT, RTP packets
You'd need RTP ports open for asterisk then.
Transfers and parking can be done at the SIP level, asterisk doesn't
have to be in the RTP path, as it can reinvite itself into the
callpath as necessary.
On Wed, Jan 27, 2010 at 5:23 AM, Vincent codecompl...@free.fr wrote:
Hello
I think I finally
Hi there everyone!
I use asterisk as a home pbx. My internet connection is a DSL one,
and I have a Linksys WRT54G that nat things for me in a 192.168.X.X
style network.
I've installed asterisk on my mac, and tried several examples I've
found on the net (voip-info, gizmo, etc.) about how
Joseph:
I think that was under the sip tab you have to configure the sip proxy,
just put the address of the firewall where you are doing port
forwarding to the asterisk box. Also in the nat field at the sipura
device put yes. And finally you have to open and forward the UDP ports
of the
So far I have gathered that on my NAT (where asterisk server is) I have
to port forward UDP ports: 5060 and range 1 - 2 to my asterisk
server
But I'm still stuck how to configure Sipura (behind NAT) what sip proxy
and out-bound sip proxy, under which tab to change.
--
#Joseph
Hello,
Does it make a difference for the NAT traversal capabilities of Asterisk
if the users are registered directly to Asterisk or if they are
registered to a SIP proxy which just relays the calls to Asterisk?
Are there any cases where Asterisk would be able to traverse the NAT if
the user is
Hi ALL;
I have users with Sipura/Linksysphones
regsitered behind Nat( useing STUNat phonenot
portforwarding) in my Asterisk box, when I try to call them
with another phone i got:
Got SIP response 404 "Not Found" back from
217.6.190.4
SIP/217.6.190.4:5060-666d is
circuit-busy
Isabove
I don't think the problem is NAT-related. Looks like To header in SIP INVITE message do not match to User ID in sipura settings.
On Thu, 2005-10-27 at 01:57 +0330, mohammad mirzaee wrote:
Hi ALL;
I have users with Sipura/Linksysphones regsitered behind Nat( useing
Hi ALL;
I have users with Sipura/Linksysphones
regsitered behind Nat( useing STUNat phonenot
portforwarding) in my Asterisk box, when I try to call them
with another phone i got:
Got SIP response 404 "Not Found" back from
217.6.190.4
SIP/217.6.190.4:5060-666d is
circuit-busy
Isabove
Hi, everyone. I have such a setup:
Asterisk---FireWall---internet--FireWall |--Cisco7960
|--Xlite
And I followed the instructions I got from the internet, change the
sip.conf in
Hi All,
I have a asterisk box that is now on its own static address on the
net.it was originally behind a nat firewall.
The problem I have is that the remote SPA-2000's that are behind nat
firewalls now fail.
here is relevent sip.con entry
[2001]
type=friend
username=2001
host=dynamic
I would comment out these lines in sip.conf
;externip=111.222.333.444
;localnet=192.168.1.0
;localmask=255.255.255.0
Then set nat=no
-Original Message-
From: Simon Chappell [mailto:[EMAIL PROTECTED]
Sent: Sunday, July 18, 2004 4:49 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
I am currently using Asterisk behind Belkin NAT router. With what ever
NAT router I have used, I have had difficulties in registration and
audio problems with my SIP provider (Iconnect and Nikotel)
It was suggested that I try to connect the asterisk box directly to the
internet avoiding the NAT
Here's the problem my sipura 2000 is setup on Nat
Network in my office
and my Asterisk Server is setup also on Nat Network
at home
the sipura can register and get calls but no audio
comes in and out of the sipura
and when i dial local extensions on the sipura i
get this error message. any
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