search for: stunde

Not sure if it's alright to talk about this here??? compiled the STUN server from Vovida on RedHat 7.3. Looks simple to configure. It isn't starting...it tries to for a long time and then just craps out. Here is my config:/etc/sysconfig/stund #!/bin/echo Not to execute. # Path to stund STUND=/usr/sbin/stund # Set the required args for STUND STUNDPRIMARYHOSTNAME=208.x.x.x # The hostname

Dear users, I am attempting to plot an ols() call from the Design package, by following the procedure explained by Harald Baayen in his 2008 book 'Analyzing linguistic data. A practical introduction to statistics using R', page 175-181. I've attached my data to this e-mail (I hope it's small enough that that's ok). First I paste all the commands I ran, followed by the

One for the gurus.. I have seen there has been a lot of discussion about using SER with Asterisk.. This to me seemed like an over kill becasue it would basically be doing most of what Asterisk is doing anyway unless you create some weird and wonderful config in SER.. Anyway, I decided to go and have a quick read through the SER docs and in the section about NAT they say that the best way to

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to

No, you don't need a sound card. Do you have ztdummy loaded or zaptel device in your system? Regards, Gus ----- Original Message ----- From: "Chris Hariga" <contact@techselesta.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, October 19, 2003 8:19 PM Subject: [Asterisk-Users] Music on hold... > Hi, > > I need a sound card and mpg123 for music on

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

Hi friends ! I want to add stun functionality in asterisk. can anybody give me some hint that how can i start that. thanks in advance Deepak Dhiman

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

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

...21..40 then ''eine halbe Minute'' when 41..59 then ''weniger als 1 Minute'' else ''1 Minute'' end when 2..45 then "#{distance_in_minutes} Minuten" when 46..90 then ''ca. 1 Stunde'' when 90..1440 then "ca. #{(distance_in_minutes.to_f / 60.0).round} Stunden" when 1441..2880 then ''1 Tag'' else "#{(distance_in_minutes / 1440).round} Tage" end end Maybe sometime this will be helpful for someo...

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Hi all, I'm trying to set up a vpn so we can access our asterisk server from the outside. We're using OpenVPN and the vpn portion seems to work beautifully. The problem come in when trying to use a sip softphone over the vpn. The softphones are able to register and the sip session works fine for dialing in and out until the call is established. Then -- no sound. Looking at

Is it possible to install a stun server on asterisk? joe a.

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Guys. Im reading a lot about ser, nat, stun, etc. And I noticed there are a lot of ways to get around nat but I would like to hear some success stories about handling nat users with multiple voip phones behind nat. I have my asterisk box behind but ports are forwarded (5060 5004 10000-20000 for rtp and 4569 for iax2) but still.. I can quite figure out what ser and stund have to do on this

Hi, I am trying to register SIP clients which are behind NAT on different network. In order to achieve this goal I think I need STUN Server . I downloaded STUN Server from http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz But I don't know how to install/configure it. And please advice me that STUN server is good idea for this scenario? Thanks in advance Wazb

x-post to: comp.emulators.ms-windows.wine f?up to: comp.emulators.ms-windows.wine Hi, (Suse Linux 10.0, KDE 3.4.2 "Level b", Wine 0.9.21) My oldest version is Kalorio 2.0. Starting "Kaloro.exe", i get this error message: ------------- wine: Unhandled page fault on read access to 0x836d000f at address 0x836d000f (thread 0009), starting debugger... WineDbg starting on pid

I have recently begun experimenting with Asterisk, and have been mightily impressed by its capabilities and flexibility. I have run across one problem, however, that challenges my ability to use it as a production system. My Asterisk box has a public Internet IP, and works great with SIP (ATA 186) clients that also have public IP addresses. Unfortunately, most of the locations that I would

...n><option value="1140">19</option><option value="1200">20</option><option value="1260">21</option><option value="1320">22</option><option value="1380">23</option></select>&nbsp;Stunde(n)</div><div id="DivDuration1" style="display: none;"><select class="n_Select" name="DurationDaysForShop"><option value="0">0</option><option value="1440" selected>1</option><option value="...