similar to: How to configure a stun server for a sip peer

Hi,all I configured two Asterisk PBXs: 1.4.X and 1.6.X. All relevant ports are opened, externip is configured in sip.conf. I think, that all relevant configurations are checked. But, no voice hear between local and remote extension. What I need to check, configure in router and PBX for resolving this issue ? How I can to install and configure STUN server ? Thanks, Oleg . -------------- next part

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

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

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

You have two options for dealing with an IP change during the registration period: 1) set the registration time to shorter period of time to minimize the downtime 2) detect that the IP address has changed via whatever method available, and then issue a "sip reload" CLI command to asterisk, which will cause it to resend registrations immediately. On Tue, Mar 31, 2015 at 1:36 PM, Daniel

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

If I correctly understand what the problem is, what I did was write a script that runs out of CRON every 15 minutes. It checks the outside IP address by querying http://checkip.dyndns.org and compares it to the IP address stored in the parameter ?externip? in the [general] section of sip.conf. If the two values are the same, the script exits quietly. If they are different, the script updates

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:

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

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

I have an asterisk 13 server behind NAT on a dynamic IP Address. It is running the PJSIP Stack It is registering to another asterisk 13 server that is on a Static IP outside of the firewall at a different location (also on the PJSIP Stack). How do we implement STUN/ICE on the server behind the dynamic Address. It does not appear to be registering properly without knowing the NAT pubic

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

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

Hello I was wondering if someone knew of an application that could check that the user has a firewall and a broadband connection that will work OK with Asterisk and VoIP. The app would first perform some bandwith + jitter tests, and will then call a STUN server to check that the firewall isn't symetric. BTW, is Asterisk now STUN-capable, or is it still to map ports manually on the firewall

Hi people! Do you guys know where to find a STUN Server Howto?! Why?! We all know, to get Asterisk behind an NAT Router to run, is a bit tricky, and you might have to fire a lot of holes in your firewall. However, I would appreciate it very much if somebody could give me great links of how to set up a STUN Server. Tamer

Dear all, I have enabled stun module and configured it in asterisk , but asterisk not using stun returned public ip address for any of the sip requests going out of my network. i have done settings as below res_stun_monitor.conf settings: [general] stunaddr = stun.ideasip.com stunrefresh = 30 stun show status Hostname Port Period Retries Status ExternAddr

Hi ? I ?wonder if?anyone could give some light on SIP NAT. I've having a friken headache with SIP NAT 1 way audio. Client - NAT? - NAT - Server Client can hear users from server side but server cant hear client. ? Ive tried every possible settings externip set localip set NAT= yes / route directmedia yes/ no ? Ive check the sip headers in the debug mode and its using the external address in

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

I couldn't be happier with the simplicity of this - but it's a hack! Hi all, I'm currently using a SIP client (BT101) to connect via DSL to a remote instance of Asterisk. - Asterisk has a private IP behind my OFFICE router. - The SIP client has a private IP behind my HOME router. I'm doing this _without_ the use of STUN or proxy servers. Here's how it works: -