similar to: stun

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

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

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 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

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

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

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

Getting "no service" display on aastra 480i. Sip debug shows an "unathorized" blub when the aastra tries to register. Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in /tftpboot/. There are none. Anyone have basic config files? Or can point me to a good link? All links I have tried, that purport to have config files, are either dead or error out.

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

Any ideas on how asterisk could accept an email (such as an email to SMS or "number at mybox.org" sort of thing) and make a phone call to a specific number and make an announcement? I imagine the first part is the big question. joe a.

I have been trying out several stun servers with Asterisk 1.6.0.9 and 1.6.1.0 and I keep getting the following message: [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]:

I have two phones, one does not need stun, the other one needs. All settings are identically, except the number/password and said above stun - not stun I use codec in the order: g729 g711u g711a Any ideas, why the user can hear me, but I cannot hear him (stun) while the other user without stun has no problem. bye Ronald

Since the beginning (of my Asterisk life) I have an install that is, supposedly, set up for call waiting. Using a TDM400p, with FXO and FXS modules. On the Analog phones, I can hear the Incoming call (call waiting) tone, but the system does not respond to a "hook flash", to place the current call on hold and answer the incoming call. I have not attempted, nor research how/if this can

HI all! I'm looking for some infos to configure stun server support for a SIP peer. I've installed Asterisk 1.4.3, but searching for stun support in chan_sip (sip.conf) i've found nothing, only a "misterious" externip = stun... But where i have to put the ip of stun server? No infos around Google and forum! :-) Thank all, regards -- Marco Ciacci Asterisk Admin Windows

My Polycom phones are displaying time, off by one hour. Seems they are on the old DST rules. How do I fix this? joe a.

Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ"): The main points listed on Asterisk's "CONS" that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT

I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040715/b37cee4b/attachment.htm -------------- next part -------------- ? Hi friends I have some doubt in connecting my firefly3rd party softphone from windows machine to asterisk server in linux . My asterisk is behind the "Port Restricted NAT". I am using STUN server to cross the

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: