Displaying 20 results from an estimated 900 matches similar to: "SIP behind NAT to pub Asterisk, best solution?"
2005 Jul 05
2
Remote SIP Connections
Hello all, I have my * server setup behind a Linksys WRT54G on
Adelphia cable. I have forwarded 5060,10000-10020, and another port
set can't remember off the top of my head but I can't seem to connect
to the * server from any locations that are direct connects to the
Internet. Am I missing a portset for forwarding?
If I use the name service (voip.*****.com) from my home connection on
the
2016 Oct 13
15
[Bug 98240] New: Kernel module fails to load on HP Pavilion V3A33AV laptop.
https://bugs.freedesktop.org/show_bug.cgi?id=98240
Bug ID: 98240
Summary: Kernel module fails to load on HP Pavilion V3A33AV
laptop.
Product: xorg
Version: unspecified
Hardware: x86-64 (AMD64)
OS: Linux (All)
Status: NEW
Severity: normal
Priority: medium
Component:
2006 Feb 15
4
SIP and firewalls?
Hi
We are currently using Asterisk 1.2.4 with IAX and app_meetme for
conferencing, but are looking to move to SIP because of issues with an IAX
control we're using.
The reason we moved from SIP to IAX in the first place was because of the
poor NAT traversal with SIP. At that stage we were using Asterisk 1.0.*. How
does Asterisk 1.2.4 handle NAT traversal and firewalls compared to the older
2006 Apr 17
24
Sip Traffic
Hi.
there is a way to MARK udp VOIP (SIP) traffic,
in order to put in a highest prio class ?
Traffic flow seems start on udp 5060 port, but
next both server and client seems jump to a
random(?) port.
I can''t use CONNMARK because is udp traffic.
I only see a pattern for L7 patch in order to
SIP traffic identification , but I run 2.4
kernel series .
When you patch 2.4 kernel with
2005 Sep 14
2
STUN vs NAT Helper
I'm wondering if anyone can recommend one over the other. I'm mostly
interested in running open source solutions, so I would prefer if
your recommendations are within the open source arena.
Basically, I contemplated the idea of using SER as a NAT Helper and
possibly as a SIP server for a portion of our user base. We prefer to
have Asterisk in the mix because of the additional
2012 Jul 13
4
R-squared with Intercept set to 0 (zero) for linear regression in R is incorrect
Hi,
I have been using lm in R to do a linear regression and find the slope
coefficients and value for R-squared. The R-squared value reported by R
(R^2 = 0.9558) is very different than the R-squared value when I use the
same equation in Exce (R^2 = 0.328). I manually computed R-squared and the
Excel value is correct. I show my code for the determination of R^2 in R.
When I do not set 0 as the
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",
2014 Dec 30
3
status - Unmonitored, how to change it
How to change status of peers "Unmonitored" to monitored?
Home users showing "Unmonitored" some display timing.
Name/Username Host Mask Port Status
zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored
clinic_server (null) (D) 255.255.255.255 0 Unmonitored
voip
2011 Oct 27
5
Asterisk Executing outbound dial number twice
Hello,
I noticed Asterisk 1.8.4.1 execute number dial twice
Log
== Using SIP RTP CoS mark 5
-- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920",
"CALLERID(num)=2066604") in new stack
== Extension Changed 4773[sipphones] new state InUse for Notify User 4701
-- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",
2013 Jul 24
1
Alpha channel in colorRamp() and colorRampPalette()
Hi all,
I had the need to create a colorbar considering the alpha channel of the colors, but colorRamp() and colorRampPalette() ignored the alpha argument in rgb(). So I performed some minor modifs. in their codes, as to support the interpolation using the alpha channel.
I guess that those simple modifications might be useful for other people, so perhaps it would be worth to add them to
2004 Dec 06
1
iax2 nativ bridge question?
hallo all,
i would like to know, as i would suspect, nativ bridiging should work also,
if only one iax party is behind an nat router and the other has a public
ip. when i connect to iax clients, which have both pubic ip's nativ
bridging is working. if one of the clients is behind an nat, the iax2
channels always get routed through the asterisk server (latest stable
version from cvs) ?? i
2005 Mar 04
2
Broadvoice + incoming call works only for ~2 minutes
Hi, all.
The asterisk setup is working fine, receiving calls via broadvoice "initially". ?
When call comes in via broadvoice number, asterisk picks it up and routes
correctly, as long as the call came in within ~2 min from the previous one.
In other words, as long as a call comes in within ~2 min since the previous one,
asterisk will answer the call. However, if the call comes in
2010 Jan 11
2
Extension Status
Hello,
I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know
how can i monitor the extension status?
when i wrote sip show peers on asterisk
Extension Domain port Status
111/111 (Unspecified) D 0 Unmonitored
1300/1300 192.168.50.111 D 5060 Unmonitored
222/222
2005 May 17
4
multiple sip accounts from same sip registrar
Dear all,
I have an asterisk sip issue which I don't believe is unique.
I use a registrar (sipgate.co.uk) where I have 3 different accounts.
These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users.
By using just one of these accounts I can set asterisk up to send and receive calls no problem.
However, when I start to introduce an
2004 Jun 15
3
Grandstreams randomly go busy with Asterisk?
I've searched the lists but I didn't find anything exactly like this.
I have two Grandstream BT101 phones connected to an Asterisk.
Periodically, for reasons that I can't determine, one or the other (or
both) of the BT101s decide(s) to go on permanent busy. Dialing that
phone gives:
-- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack
2015 May 28
4
Peer is UNREACHABLE
Hi list!
I have a problem and I hope someone can help me...
I configured an Asterisk on a VM to serve more accounts and act as a proxy to
other SIP-providers.
The first account running on my phone works without any problem.
A second account, running on the phone of my wife, is always UNREACHABLE.
I can just see in the log:
[May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer
2007 Jul 08
2
Auto Fall Through when kicking users in MeetMe
Hi all,
My scenario is such that I have three users connected to a conference.
CLI> meetme list 1234
User #: 01 9176502096 <no name> Channel: Zap/23-1
(unmonitored)00:00:32
User #: 02 john john Channel: SIP/john-b7800468
(unmonitored) 00:00:28
User #: 03 6463875998 <no name> Channel: Zap/22-1
(unmonitored)00:00:19
3 users in that
2010 Jun 15
4
can't seem to register, status unmonitored
Hi everybody,
I am trying to register my softphone(twinkle) on an asterisk server.
Everything seems to be fine.
Here is the output on show registrations in twinkle:
Tue 18:57:51
nikhil: you have the following registrations
<sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013
208 is ip of the asterisk server.
on the server on doing 'sip show peers' , it
2007 Dec 02
4
get SIP extension status without calling it
Hi,
I am trying to get a SIP extension's status without
actually making a call.
I am using sofia-sip's "options" example utility and
the sip clients are SJphone softphones.
2004 Jul 08
2
Cisco 7960 NAT question
I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The
asterisk box is on a WAN connection on the other end of a DS3, the
phones connect fine to the Asterisk server as you can see from the
output of show sip peers below.
tp3/tp3 <firewall-ip> D N 255.255.255.255 60665
Unmonitored
tp2/tp2 <firewall-ip> D N