Displaying 20 results from an estimated 10000 matches similar to: "is IAX required for firewall and router?"
2004 Jan 20
0
Outbound call with Go2Call
Any got experience with these?
I couldn't fint anything in any postings...
it seems they have a h.323 on voip01.go2call.com and a sip on
sip01.go2call.com
I have tried to register with some of the same as I use for nikotel, but
Asterisk does not want to register.
I've tried to use both the user name (ingvald) and the PIN code 440.... as
authentication.
---from sip.conf----
2005 Feb 23
2
multiple sip phones behind firewall
Hello List,
Can you please point me to the right resources on making multiple sip
phones behind a firewall w/ private address work with asterisk w/c is on
a public network.
I have seen STUN on the grandstream and Xtunnels on X-lite. What is most
deployed by members here with similar setups?
Thanks.
--
Cheers,
Paul P. Pongco
2006 Dec 12
1
AGI problema
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<font face="Verdana">Hi all. I've written a AGI in C language.
2004 May 22
1
Re: Sipura and STUN (was: rejected NOTIFY requests)
At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote:
>>[snip]
>Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk
>can handled the NAT traversal all by itself with Qualify (as John points
>out) disabling the NOTIFY will not change anything.
>
>The NOTIFY will in no way affect the status - unreachable/reachable.
>
>Another problem with the SIPURA is
2004 May 22
1
Re: Sipura and STUN (was: rejected NOTIFY re quests)
Sipura does include STUN as an option. It has for quite some time. We are
using it with all of our Sipuras behind NAT'd gateways and it works great!
Try upgrading to the latest Sipura firmware rev.
Darren Nay
> -----Original Message-----
> From: John Todd [mailto:jtodd@loligo.com]
> Sent: Saturday, May 22, 2004 1:57 PM
> To: asterisk-users@lists.digium.com
> Subject:
2004 Apr 18
2
grandstream and stun
Hi,
I noticed some issues with how grandstream handles
stun test. GS is running version 1.0.4.50. First I
reset the NAT router. Then reboot GS, get results of
"restricted cone". Immediately reboot GS, get results
"full cone". I tried quite a few public and commercial
stun servers. Also tried different model/version of
linksys routers. I always got the same issue. Winstun
on
2003 Dec 05
3
GrandStream Budgetone Phone & DHCP & General Observations
Symptom: Phone after about 15mins will stop functioning
Problem: DHCP lease renewed but default route dropped
Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released
It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia
2006 Nov 13
2
STUN with one public and one private IP?
I'm finishing up deploying an Asterisk (Trixbox) box at work. Wow, I
thought Asterisk was cool by itself, but Trixbox has made just about
everything turnkey. Great stuff!
So... we're using Grandstream GXP-2000 handsets to connect to the Trixbox,
which sits on our DMZ with a single public IP. I need the phones to work
from random places behind NAT, as well as in the office. I'm using
2005 Jul 28
1
Problem with BT100 behind iptables firewall
Greetings,
I am trying to get an IP phone working through a linux based iptables
firewall. I have an asterisk server with a public IP address.
I ran netcheck from FWD. It says that it is a Port Restricted Nat.
I tried the recommended FWD approach, changing the FWD-specific settings
to the * server's. I have tried every conveivable config on the phone
(Yes to NAT traversal with STUN
2003 Jun 06
3
SIP codecs
i've been having a problem getting two SIP phones
to bridge running through asterisk, actually one is
a SIP softphone, SJ Phone, and the other is the
Go2Call calling gateway.
Someone suggested that I don't have the right codecs.
How do I find out which codecs are installed, and how
can I install further codecs? Any suggestions which
would be the right one?
I think hte problem is from the
2005 Mar 03
2
Asterisk + SIP + NAT - seriously, what's the secret?
I'm at my wit's end!
I've spent 2 days now trying to get what I thought was a very simply SIP
+ NAT arrangement working. I've trawled the web and picked brains, but
nothing anyone suggests work.
My setup is very simple. I have a * server in a datacentre, with a
public IP address. There is no firewall in place, it's completely open
(at least, as far as I'm concerned). I
2006 Nov 06
7
several behind NAT
I've got my asterisk server in the DMZ of my local LAN - I've used my
Budgetone and GXP2000's from the Internet- on direct IP connections
with no problems. However, I'm about to deploy about 5 phones
(either budgetone or GXP2000's) all on a LAN behind a NAT- on a
different network than the Asterisk server. Should I look into using
STUN servers? Will this setup be a
2009 May 11
1
Anyone with a working pfSense firewall configuration?
Other SIP clients behind the firewall (not using STUN, work).
We have a SIP client using STUN and ICE behind a pfSense firewall.
The firewall is behaving oddly.
REGISTER packets work fine.
But when the client tries to make a call, the first INVITE packet from
the client pass through the firewall and makes it to the Asterisk
server.
The Asterisk server sends back a 401 client sends ACK,
2004 Sep 21
2
Asterisk , ISA Firewall/VPN , STUN and other issues
I have just finished compiling and installing Asterisk on a test Debian
system. All is working well. We are now attempting to get remote offices
to test the system I have installed both a SIP and an IAX client at a
remote office. Then I connect to our office via Microsoft ISA firewall
and the Windows XP VPN client. Neither of the softphones will connect.
On the IAX softphone I just get a ringtone
2008 Feb 03
1
Multiple SIP phones behind a Linksys firewall
And the firewall is in between the phones and both servers or are you registering the phones on a local server and trunking to the other server through the firewall?
In terms of nat and Cisco 7960s I've never had a issue registering two of them behind nat to a server on the other side, however, if you called one phone from the other, you'd end up with one way audio.
-----Original
2005 Sep 13
2
Nat & Sip & Pain
Hi everyone,
I decided to have a look at SIP & NAT again and I've been at it for a
[quite a] few hours but typically nothing is working for me. Actually
I'm not sure if SIP and NAT can ever work but some emails on this list
do suggest that someone has got it working, once, maybe.
I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports
"Outbound Proxy",
2004 Dec 18
5
Q about IAX (and IAXy)
This is somewhat related to my other query on the list regarding NAT
traversal.
I have heard many times that IAX is "NAT-transperant". I am unsure how
it accomplishes this.
I do know that SIP works like this: your SIP device send a request to
the SIP server (usually on port 5060) with whatever command. The SIP
server respends to your device's "apparent" IP and port (this
2005 Mar 22
1
Nat and firewall port forwarding - is it really required?
I have a question which I'm sure has been asked before but my research has
yet to find it.
I have Asterisk running on a Linux server but in order to get it to connect
I needed to punch a hole in my firewall on port 5060 for it to receive the
registration responses from broadvoice.
If I run sjphone as a softclient on my home PC I do not need to punch that
same hole and it works just fine.
2003 May 27
1
please help (reposted) - re. * connecting to a commercial call service
hi,
maybe someone out there already has some experience and can help me.
I have just ordered an E100P card from Digium, I already have a basic
asterisk setup up & running.
My application is the following :
I want to accept incoming calls from the PSTN to Asterisk, and without
asking anything of the client just pass them immediately to a call gateway
in USA, actually we are planning to use
2004 Jul 22
6
D-Link DPH-80S vs *
List,
The D'Link phones are not reliable at this time. I am trying to get them fix their Firmware to my specifications. It is half done so far. However there are still hurdles. below email is self explanatory. At present if you want to use these phones, you need to buy D'Link's SIP Server and run this as one of your SIP servers in the blend to call to Asterisk.
Seshu Kanuri
"G