Re: [asterisk-users] STUN re-evalutation every 2 minutes ??
On Saturday 01 September 2018 at 22:12:50, sean darcy wrote: > 13.21.0 > > Every 2-3 minutes: Does it really vary, or is it more like "every 150 seconds"? > Sep 1 16:00:57] WARNING[150257]: res_stun_monitor.c:140 > stun_monitor_request: STUN poll got no response. Re-evaluating STUN > server address. What relevant firewall rules have you got? > [Sep 1 16:02:18] NOTICE[150257]: res_stun_monitor.c:151 > stun_monitor_request: Old external address/port :42562 now > seen as :33904. What sort of Internet connectivity router are you using? > IAX, got a network change message, renewing all IAX registrations. > SIP, got a network change message, renewing all SIP registrations. > > Always just for a different port number. Sounds to me like your router has got a very short term connection tracking table, or else your firewall rules aren't allowing the required replies. > I've tried a number of STUN servers with the same result. Now using > counterpath : > > /etc/asterisk/res_stun_monitor.conf:stunaddr = stun.counterpath.net > > Probably harmless, but odd. > > sean Regards, Antony. -- "I estimate there's a world market for about five computers." - Thomas J Watson, Chairman of IBM Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STUN re-evalutation every 2 minutes ??
13.21.0 Every 2-3 minutes: Sep 1 16:00:57] WARNING[150257]: res_stun_monitor.c:140 stun_monitor_request: STUN poll got no response. Re-evaluating STUN server address. [Sep 1 16:02:18] NOTICE[150257]: res_stun_monitor.c:151 stun_monitor_request: Old external address/port :42562 now seen as :33904. IAX, got a network change message, renewing all IAX registrations. SIP, got a network change message, renewing all SIP registrations. Always just for a different port number. I've tried a number of STUN servers with the same result. Now using counterpath : /etc/asterisk/res_stun_monitor.conf:stunaddr = stun.counterpath.net Probably harmless, but odd. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stun Server
We have been running a windows stun server for 5 years now and I would like to change to either a linux of freebsd based unit to phase out the old XP box in our datacenter. What should I look at that would be a good replacement. The windows box has worked but the hardware is at end of life and I want to move it to a vm and I don't want Windows. Any advise is apperciated. Thanks zktech -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stun Server
I like Xen. It's free and rock solid. VMWare is great but their money greedy. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Wednesday, July 27, 2011 9:30 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Stun Server We have been running a windows stun server for 5 years now and I would like to change to either a linux of freebsd based unit to phase out the old XP box in our datacenter. What should I look at that would be a good replacement. The windows box has worked but the hardware is at end of life and I want to move it to a vm and I don't want Windows. Any advise is apperciated. Thanks zktech -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STUN
Hello, i want to instal STUN Server in my Asterisk-PC. is it possible ? if yes, how kann i do it ? where can i find STUN Program? Thanks for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STUN setting in Asterisk 1.6.X
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]: 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 No matter which STUN server I point to I get those messages. Am I missing some other setting? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STUN setting in Asterisk 1.6.X
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed No matter which STUN server I point to I get those messages. Am I missing some other setting? Hey Carlos, That just means the stun request failed, there are several reasons for that, I won't even try to guess. So, first try this on the Asterisk CLI: stun set debug on That should give you (and us) more information to troubleshoot why the stun request failed (also enable debug and verbosity as usual). -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stun clients and canreinvite
Howdy, Scenario: Asterisk server Customer connected over internet using nat Customer phones are Linksys 942 with Stun enabled Issue: Inbound and Outbound calls work fine. But when phones call each other internally we have to carry the voice stream ie using t on dial commands. Question: Is there a better way of doing this or another way to get the media to stream internally on the customer network rather than us carrying it? We have to keep Stun on the phones to get the media to flick off on outbound calls. Cheers, Taff.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stun with hosted asterisk solution???
Howdy, I have the following issue and would like to know if anyone has got around this before. IP Phones - Linksys 942 Sip server - Asterisk 1.4.13 Stun server - Vovida Ok heres the issue. We have multiple client phones on their own network behind a natted connection. We have setup the phones to be natted and also pointing to our stun server. Now when the phones make an outside call to the PSTN stun kicks in and their rtp streams are carried from the phones to the sip provider without any issues. Now when the phones dial each other internally the rtp stream is still carried via stun and therefore fails as its pointing to the same ip on the same router. Now by adding t to the asterisk dial commands for each internal phone the inbound calls work fine but the rtp streams are carried through asterisk rather than between themselves on their network. Also in this scenario when you try conference an outside phone with an inside phone it fails due to stun and outside address problems. So my question is can we set up or change something on the phones or asterisk to allow the phones rtp to go across the local network on internal calls and via stun for outbound pstn calls? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STUN
Hi guys, I'm trying to implement STUN support in *, is there anyone here which have any experience in something like that? I've got the STUND and I'll try to buld a patch or something for sip. Any ideas or existing implementation would be nice. I know openpbx have it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
On Tue, 3 Apr 2007, Joe Acquisto wrote: Is it possible to install a stun server on asterisk? You can install a stun server on the same PC that asterisk is running on. No need for it to be part of asterisk itself, it's a totally separate program and will exist happily on the same server. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
Gordon Henderson [EMAIL PROTECTED] Wrote: 4/4/2007 3:32 AM: On Tue, 3 Apr 2007, Joe Acquisto wrote: Is it possible to install a stun server on asterisk? You can install a stun server on the same PC that asterisk is running on. No need for it to be part of asterisk itself, it's a totally separate program and will exist happily on the same server. Gordon now for the next DA question, where to find it (one)? Google has not been my friend. An alleged spot on sourceforge turned up blank. joe a. +++ www.j4computers.com 845-687-4563 Stone Ridge, NY 12484 +++ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
On Wed, 4 Apr 2007, Joe Acquisto wrote: Gordon Henderson [EMAIL PROTECTED] Wrote: 4/4/2007 3:32 AM: On Tue, 3 Apr 2007, Joe Acquisto wrote: Is it possible to install a stun server on asterisk? You can install a stun server on the same PC that asterisk is running on. No need for it to be part of asterisk itself, it's a totally separate program and will exist happily on the same server. Gordon now for the next DA question, where to find it (one)? Google has not been my friend. An alleged spot on sourceforge turned up blank. http://sourceforge.net/projects/stun/ Which is linked from: http://www.vovida.org/applications/downloads/stun/ That's what I'm running. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
. . . http://sourceforge.net/projects/stun/ Which is linked from: http://www.vovida.org/applications/downloads/stun/ That's what I'm running. Gordon Thanks. Looking there, why would I need a stun client if the device/softdevice already has STUN support? All I should need is the linux daemon thing-let, correct? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
On Wed, 4 Apr 2007, Joe Acquisto wrote: . . . http://sourceforge.net/projects/stun/ Which is linked from: http://www.vovida.org/applications/downloads/stun/ That's what I'm running. Gordon Thanks. Looking there, why would I need a stun client if the device/softdevice already has STUN support? No. All I should need is the linux daemon thing-let, correct? Yes. AIUI, You'll need 2 IP addresses to run it on though. Gordon joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stun
Joe Acquisto wrote: . . . http://sourceforge.net/projects/stun/ Which is linked from: http://www.vovida.org/applications/downloads/stun/ That's what I'm running. Gordon Thanks. Looking there, why would I need a stun client if the device/softdevice already has STUN support? All I should need is the linux daemon thing-let, correct? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The linux daemon is also downloadable there i think ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] stun
Is it possible to install a stun server on asterisk? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STUN and SNMP
22 jan 2007 kl. 07.38 skrev Thomas Deillon: Hi all, I read somewhere that asterisk v 1.4 can make Stun and SNMP. I tried to find more information on these features but I didn’t find any clues. Someone find a way to use it? There's a module called res_snmp that implements an SNMP agent or an NetSNMP agent plugin. You need to have Netsnmp installed for this to be compiled, as well as have it enabled in menuselect. The stun support is only implemented in the google talk/jingle channel driver. /O___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STUN and SNMP
Hi all, I read somewhere that asterisk v 1.4 can make Stun and SNMP. I tried to find more information on these features but I didn't find any clues. Someone find a way to use it? Thanks, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STUN in Asterisk 1.4
Browsing through the developers documentation and 1.4 source, I see references to STUN in the code and documentation. Does 1.4 have support for STUN, if so how is it configured? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STUN with one public and one private IP?
Are you kidding? Lighten up people! Al made a friendly recommendation based on the comments regarding TrixBox. Go have a beer... take a load off... enjoy the holidays. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STUN with one public and one private IP?
You said voxbox is better, but even the link you gave for them didn't work. I googled, and apparantly links are broken on their website. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STUN with one public and one private IP?
I never said voxbox is better than trixbox. I said You like trixbox Should try voxbox. The link is: http://www.easyvoxbox.org/ Trixbox has good and bad points (loads from RPM's) Voxbox has good and bad points (Loads from source) I like source better than RPM's -- Thats me, but I do programming Both are ISO and run asterisk You tell me the better one. If you try both... Good thing that I only told you about two out of (Well god only knowns) I know ten diff installs off hand. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Zeeshan Zakaria wrote: You said voxbox is better, but even the link you gave for them didn't work. I googled, and apparantly links are broken on their website. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0649-0, 11/15/2006 - 11/16/2006 11:30:14 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STUN with one public and one private IP?
On Thu, 16 Nov 2006, Zeeshan Zakaria wrote: You said voxbox is better, but even the link you gave for them didn't work. I googled, and apparantly links are broken on their website. Not to mention that the humongous ad for voxbox, in response to my TECHNICAL QUESTION, was completely out of line. If I still had the message, I'd suggest that the poster's email address be removed from this mailing list. -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STUN with one public and one private IP?
On Thu, 16 Nov 2006, Al Bochter wrote: I never said voxbox is better than trixbox. I said You like trixbox Should try voxbox. Yes. You didn't answer my question, and then you posted all of THIS crap... Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email No content, nothing but ad. -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STUN with one public and one private IP?
On Mon, 13 Nov 2006, Steve Sobol wrote: So I'm wondering... I'm using stund from SourceForge. Is there any reason I couldn't give the Trixbox's public IP address as the primary and 127.0.0.1 as the secondary? I believe Asterisk is listening on the loopback interface... Following up to myself. 127.0.0.1 didn't work, but setting up an alias on my eth0 interface and configuring it as 192.168.1.1 did. -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 STUN, and I understand I need a primary IP and an alternate IP to make STUN work. Well, I got STUN working here on amethyst.justthe.net, which has a bunch of available public IPs, but the Trixbox only has one public IP, and I have to request (and pay for) more IPs from the phone company if I need any more. And I'd really prefer that STUN be running in the office, and not on my personal server. So I'm wondering... I'm using stund from SourceForge. Is there any reason I couldn't give the Trixbox's public IP address as the primary and 127.0.0.1 as the secondary? I believe Asterisk is listening on the loopback interface... Thanks in advance, Steve -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STUN with one public and one private IP?
You like trixbox Should try voxbox. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Steve Sobol wrote: 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 STUN, and I understand I need a primary IP and an alternate IP to make STUN work. Well, I got STUN working here on amethyst.justthe.net, which has a bunch of available public IPs, but the Trixbox only has one public IP, and I have to request (and pay for) more IPs from the phone company if I need any more. And I'd really prefer that STUN be running in the office, and not on my personal server. So I'm wondering... I'm using stund from SourceForge. Is there any reason I couldn't give the Trixbox's public IP address as the primary and 127.0.0.1 as the secondary? I believe Asterisk is listening on the loopback interface... Thanks in advance, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN?
Hi all, Could someone point at resources for running Asterisk behind a firewall. STUN keeps coming up but, alas, Im easily confused. J Ray ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN?
On Jun 26, 2006, at 9:32 AM, Raymond Tant wrote: x-tad-smallerHi all,/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerCould someone point at resources for running Asterisk behind a firewall./x-tad-smallerx-tad-smallerSTUN keeps coming up but, alas, I’m easily confused. /x-tad-smallerx-tad-smallerJ /x-tad-smaller STUN is just a way to discover the true address of a machine behind a NAT. Firewalls aren't really an issue per se, other then needing to open particular ports for asterisk to use. For example, udp port 4569 for IAX2 traffic, and 5060 for SIP signaling, as well as ports in the 1-2 range RTP traffic. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN?
please type in google.com: STUN server ALG The fourth result is a good and small explanation. On 6/26/06, Martin Joseph [EMAIL PROTECTED] wrote: On Jun 26, 2006, at 9:32 AM, Raymond Tant wrote: Hi all, Could someone point at resources for running Asterisk behind a firewall. STUN keeps coming up but, alas, I'm easily confused. J STUN is just a way to discover the true address of a machine behind a NAT. Firewalls aren't really an issue per se, other then needing to open particular ports for asterisk to use. For example, udp port 4569 for IAX2 traffic, and 5060 for SIP signaling, as well as ports in the 1-2 range RTP traffic. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] stun server
Hi list, I want to setup a stun server. I tried stun server from vovida and mystun, but my voip phone said that logon failed. When I use stun.xten.com works fine. This is vovida stun in verbose mode: ***received on A1:P1 Got a request (len=28) from xxx.186.145.226:5060 Received stun message: 28 bytes ChangeRequest = 0 Request parsed ok BindRequest does not contain MessageIntegrity Request is valid: flags=0 changeIp=0 changePort=0 from = xxx.186.145.226:5060 respond to = xxx.186.145.226:5060 mapped = xxx.186.145.226:5060 Encoding stun message: Encoding MappedAddress: xxx.186.145.226:5060 Encoding SourceAddress: xxx.96.148.138:3478 Encoding ChangedAddress: xxx.96.148.139:3479 Encoding XorMappedAddress: xxx.101.192.77:16411 Encoding ServerName: Vovida.org 0.96 Thanks -- Serghei. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun server
I seriously doubt that message has something to do with the stun server. Have you read what the STUN server does? The message you are getting is most likely to be because of wrong registration user name or password in your voip phone. Any way, if you are still interested in having the stun properly installed and you have Gentoo Linux, I have made an ebuild that will install it right with a configuration file. Just let me know. Regards On 5/11/06, Serghei Amelian [EMAIL PROTECTED] wrote: Hi list, I want to setup a stun server. I tried stun server from vovida and mystun, but my voip phone said that logon failed. When I use stun.xten.com works fine. This is vovida stun in verbose mode: ***received on A1:P1 Got a request (len=28) from xxx.186.145.226:5060 Received stun message: 28 bytes ChangeRequest = 0 Request parsed ok BindRequest does not contain MessageIntegrity Request is valid: flags=0 changeIp=0 changePort=0 from = xxx.186.145.226:5060 respond to = xxx.186.145.226:5060 mapped = xxx.186.145.226:5060 Encoding stun message: Encoding MappedAddress: xxx.186.145.226:5060 Encoding SourceAddress: xxx.96.148.138:3478 Encoding ChangedAddress: xxx.96.148.139:3479 Encoding XorMappedAddress: xxx.101.192.77:16411 Encoding ServerName: Vovida.org 0.96 Thanks -- Serghei. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun server
On Thursday 11 May 2006 16:51, Moises Silva wrote: I seriously doubt that message has something to do with the stun server. Have you read what the STUN server does? The message you are I known what do a stun server. getting is most likely to be because of wrong registration user name or password in your voip phone. Any way, if you are still interested in having the stun properly installed and you have Gentoo Linux, I have made an ebuild that will install it right with a configuration file. Just let me know. I'm interested, I use gentoo on my asterisk/stun server. Thanks a lot. -- Serghei. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun server
I was thinking in sending you an attachment, but I have decided to put it on the web, you can get it in http://phpmexic.u33.0web-hosting.com/wordpress/ebuilds/vovida-stun-0.96-ebuild.tar.bz2 Let me know if you it have bugs Regards On 5/11/06, Serghei Amelian [EMAIL PROTECTED] wrote: On Thursday 11 May 2006 16:51, Moises Silva wrote: I seriously doubt that message has something to do with the stun server. Have you read what the STUN server does? The message you are I known what do a stun server. getting is most likely to be because of wrong registration user name or password in your voip phone. Any way, if you are still interested in having the stun properly installed and you have Gentoo Linux, I have made an ebuild that will install it right with a configuration file. Just let me know. I'm interested, I use gentoo on my asterisk/stun server. Thanks a lot. -- Serghei. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun server
On Thursday 11 May 2006 18:32, Moises Silva wrote: I was thinking in sending you an attachment, but I have decided to put it on the web, you can get it in http://phpmexic.u33.0web-hosting.com/wordpress/ebuilds/vovida-stun-0.96-ebu ild.tar.bz2 Let me know if you it have bugs I'm not lucky, my voip phone do not want to work with vovida-stun. The phone is Perfectone IP-300. The command is: /usr/sbin/stunserver -h xxx.96.148.138 -a xxx.96.148.139 -p 3478 -o 3479 In attachment is tcpdump runned stun server. It's very strange, with stun.fwdnet.net it works perfectly. Question: in dns both ip's must be registered? In this moment only xxx.96.148.138 is registered as stun.blabla [...] Thanks -- Serghei. 20:19:56.711700 IP (tos 0x0, ttl 122, id 4, offset 0, flags [none], proto: UDP (17), length: 56) xxx.186.145.226.5060 xxx.96.148.138.3478: SIP, length: 28 \000\001\000\010*\323q\307.\244~S\012 \015\035\014'Ev\000\003\000\004\000\000\000\000 20:19:56.711746 IP (tos 0x0, ttl 64, id 8, offset 0, flags [DF], proto: UDP (17), length: 116) xxx.96.148.138.3478 xxx.186.145.226.5060: SIP, length: 88 \001\001\000D*\323q\307.\244~S\012 \015\035\014'Ev\000\001\000\010\000\001\000\000\010\236\353\267!\000\000\000udp and host devel.thel.ro\000\267\021\000\000\0003478\000H\352\267host\021\000\000\0005060 20:19:56.756315 IP (tos 0x0, ttl 122, id 5, offset 0, flags [none], proto: UDP (17), length: 411) xxx.186.145.226.5060 xxx.96.148.138.5060: SIP, length: 383 REGISTER sip:sip.datadrill\000\000\010\236\353\267!\000\000\000udp and host devel.thel.ro\000\267\021\000\000\0003478\000H\352\267host\021\000\000\0005060\000H\352\267\020\000\000\000)\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000PV\027\010(V\027\010dns\0001\000\000\000\200F\027\010\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000\000\000\000\000\000\000\000\000files\000\000\000\000\000\000\000)\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000\000\000\000\000\000\000\000\000dns\000\031\000\000\000\030G\027\010\300F\027\010services\000\000\000\000)\000\000\000\350F\027\010\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000\000\000\000\000\000\000\000\000db\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000\000\000\000\000\000\000\000\000files\000\000\000\000\000\000\000\031\000\000\000xA\027\0100G\027\010protocols\000\000\000)\000\000\000\330V\027\010\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000 20:19:56.756474 IP (tos 0x0, ttl 64, id 18140, offset 0, flags [DF], proto: UDP (17), length: 453) xxx.96.148.138.5060 xxx.186.145.226.5060: SIP, length: 425 SIP/2.0 100 Trying Via: S\000\000\010\236\353\267!\000\000\000udp and host devel.thel.ro\000\267\021\000\000\0003478\000H\352\267host\021\000\000\0005060\000H\352\267\020\000\000\000)\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000PV\027\010(V\027\010dns\0001\000\000\000\200F\027\010\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000\000\000\000\000\000\000\000\000files\000\000\000\000\000\000\000)\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000\000\000\000\000\000\000\000\000dns\000\031\000\000\000\030G\027\010\300F\027\010services\000\000\000\000)\000\000\000\350F\027\010\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000\000\000\000\000\000\000\000\000db\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000\000\000\000\000\000\000\000\000files\000\000\000\000\000\000\000\031\000\000\000xA\027\0100G\027\010protocols\000\000\000)\000\000\000\330V\027\010\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000\000\000\000\000\000\000\000\000db\000\006!\000\000\000\270W\027\010`W\027\010ethers\\000\000\000\000\000 20:19:56.756515 IP (tos 0x0, ttl 64, id 18141, offset 0, flags [DF], proto: UDP (17), length: 535) xxx.96.148.138.5060 xxx.186.145.226.5060: SIP, length: 507 SIP/2.0 401 Unauthorized \000\000\010\236\353\267!\000\000\000udp and host
Re: [Asterisk-Users] stun server
Thanks for the fix :) About DNS, I think only the primary stun address should be registered. Try using as stun server grievous.ivsol.net, is our stun server and is installed with the ebuild I sent you. That will give us more hints about where the problem might be. Try using the stunclient application from inside the network where the phone is, it will help you to debug. On 5/11/06, Serghei Amelian [EMAIL PROTECTED] wrote: On Thursday 11 May 2006 18:32, Moises Silva wrote: I was thinking in sending you an attachment, but I have decided to put it on the web, you can get it in http://phpmexic.u33.0web-hosting.com/wordpress/ebuilds/vovida-stun-0.96-ebu ild.tar.bz2 Let me know if you it have bugs I'm not lucky, my voip phone do not want to work with vovida-stun. The phone is Perfectone IP-300. The command is: /usr/sbin/stunserver -h xxx.96.148.138 -a xxx.96.148.139 -p 3478 -o 3479 In attachment is tcpdump runned stun server. It's very strange, with stun.fwdnet.net it works perfectly. Question: in dns both ip's must be registered? In this moment only xxx.96.148.138 is registered as stun.blabla [...] Thanks -- Serghei. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun server [SOLVED]
On Thursday 11 May 2006 21:10, Moises Silva wrote: Thanks for the fix :) no problem About DNS, I think only the primary stun address should be registered. Try using as stun server grievous.ivsol.net, is our stun server and is installed with the ebuild I sent you. That will give us more hints about where the problem might be. Try using the stunclient application from inside the network where the phone is, it will help you to debug. When i've seen that you stun works, I've got an ideea: the stun and asterisk cannot share the same ip. I swapped the primary ip with secondary ip, and voila :-) Thanks a lot for help. -- Serghei. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN Server info
Hi, Do we need STUN server with Asterisk(1.2.6) for SIP phones which are using NAT on different networks ??? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN vs NAT Helper
What is this sip-nat-helper thing, is there a website were we can get some info on it, partly thinking as the question before was relating to open source software, I would assume that I could download this thing. Dan On Wed, 14 Sep 2005 [EMAIL PROTECTED] wrote: If you have a linux box, then u can try sip-nat-helper for netfilter. Cheers. Mensaje citado por: Waldo Rubinstein [EMAIL PROTECTED]: 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 wealth of features it can add to the SIP services (e.g. voicemail, ivr, call queueing, etc). All of our clients are behind NATs, mainly basic NATs such as linksys routers behind DSL modems. I read on the wiki that STUN is not readily supported by most clients, so I don\'t know if its worth the effort or if we should just concentrate on getting SER working with Asterisk. Any ideas or suggestions? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Registrate desde http://servicios.arnet.com.ar/registracion/registracion.asp?origenid=9 y particip? de todos los beneficios del Portal Arnet. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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 wealth of features it can add to the SIP services (e.g. voicemail, ivr, call queueing, etc). All of our clients are behind NATs, mainly basic NATs such as linksys routers behind DSL modems. I read on the wiki that STUN is not readily supported by most clients, so I don't know if its worth the effort or if we should just concentrate on getting SER working with Asterisk. Any ideas or suggestions? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN vs NAT Helper
If you have a linux box, then u can try sip-nat-helper for netfilter. Cheers. Mensaje citado por: Waldo Rubinstein [EMAIL PROTECTED]: 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 wealth of features it can add to the SIP services (e.g. voicemail, ivr, call queueing, etc). All of our clients are behind NATs, mainly basic NATs such as linksys routers behind DSL modems. I read on the wiki that STUN is not readily supported by most clients, so I don\'t know if its worth the effort or if we should just concentrate on getting SER working with Asterisk. Any ideas or suggestions? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Registrate desde http://servicios.arnet.com.ar/registracion/registracion.asp?origenid=9 y participá de todos los beneficios del Portal Arnet. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN vs NAT Helper
I think STUN is quite widely supported by hardphones. I'd be interested to know if STUN is a magic fix to SIP NAT - I've a feeling that its not. Derek Waldo Rubinstein wrote: 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 wealth of features it can add to the SIP services (e.g. voicemail, ivr, call queueing, etc). All of our clients are behind NATs, mainly basic NATs such as linksys routers behind DSL modems. I read on the wiki that STUN is not readily supported by most clients, so I don't know if its worth the effort or if we should just concentrate on getting SER working with Asterisk. Any ideas or suggestions? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Derek Conniffe Rivertower Ltd Ireland: (Freephone) 1800 719 400 Ireland: (Local) 01 244 9719 United Kingdom: 0870 068 2368 International: 00 353 1 244 9719 Derek Conniffe DDI: 01 201 0146 (International: 00 353 1 201 0146) Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823) Fax: 01 201 0085 (International: 00 353 1 201 0085) Email: [EMAIL PROTECTED] Web: http://www.rivertowerhosting.com begin:vcard fn:Derek Conniffe n:Conniffe;Derek org:Rivertower Ltd;IT adr:Dublin 2;;46 Upper Mount Street;Dublin;Dublin;Dublin 2;Ireland email;internet:[EMAIL PROTECTED] tel;work:+353 1 201 0146 tel;fax:+353 1 201 0085 tel;cell:+353 86 856 3823 note;quoted-printable:Ireland: (Freephone) 1800 719 400=0D=0A= Ireland: (Local) 01 244 9719=0D=0A= United Kingdom: 0870 068 2368=0D=0A= International: 00 353 1 244 9719=0D=0A= url:http://www.rivertowerhosting.com version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN on PAP2-NA 2.0.12(LS)
Hello, I'm having intermittent STUN trouble. Every one out of perhaps 5 reboots the PAP2 contacts STUN ... on the other attempts it just skips that step all together. I have been verifying this using ethereal which shows the distinctive STUN server DNS lookup followed by about 10 STUN queries (when it works - when it doesn't it skips all that including the initial DNS lookup ... apparently it doesn't even try) Has anyone else had this trouble? It's hard to find new firmware for PAP2-NA ... but maybe that's the problem ... is 2.0.12 very obsolete? The newest I saw on the web was 2.0.13 and even that one was hard to find. Thank you, Tomas ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stun support
Hi Eric, How one can make outgoing call to a SIP user sitting in the Internet when Asterisk is not configured with Outbound Proxy to some SIP Proxy server on the Internet for this simple scenario ? SIP UA A --- Asterisk ---NAT---Internet--- SIP UA B I know if Asterisk is configured with Outbound proxy sitting in the Internet, it will work. What will happen when it is not configured with that? Thanks in advance Rajeew === -Original Message- From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED] Sent: Monday, August 08, 2005 8:05 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Stun support someshwarak wrote: Hi * users, I want to know if STUN suport is available with Asterisk. Kindly let me know. I have posted this also in DEV list but none replied to me. Short Answer: No. Longer Answer: No, and most people that think they need STUN don't actually need it. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stun support
Rajeew Kumar Singh wrote: Hi Eric, How one can make outgoing call to a SIP user sitting in the Internet when Asterisk is not configured with Outbound Proxy to some SIP Proxy server on the Internet for this simple scenario ? SIP UA A --- Asterisk ---NAT---Internet--- SIP UA B I know if Asterisk is configured with Outbound proxy sitting in the Internet, it will work. What will happen when it is not configured with that? This is just a standard home/SME NAT setup with Asterisk. Nothing special about it. Heck, the SIP devices are not even behind NAT! Use externip= and localnet= in sip.conf then port forward 5060/UDP and the RTP ports on the NAT router. The only significant is issue is making sure the remote SIP devices use the RTP ports you are expecting them to and making sure that the SIP device does not have any NAT options enabled. I use an even more complicated configuration where I have this setup: SIP UA A -- Asterisk -- NAT Internet My SIP UA A can roam between the local network, the internet with public IP and the internet with a NAT IP. No config changes at all to Asterisk or the SIP UA when I move bewteen networks. Just unplug the device from my home network and go to another network and plug it in. There are two significant limitations to my setup. The first is that all audio that goes between remote SIP devices that are behind NAT must go thru Asterisk. i.e. Reinvites won't work. My response to this issue is Who cares?. A VoIP service provider will have most of their calls going from the SIP UA to the PSTN. Assuming Asterisk is acting as the PSTN gateway, then the audio will have to go thru Asterisk anyway, so reinvites not working is a non-issue. The second limitation is that the NAT IP should not be dynamic. Asterisk has significant issues with ANY transient DNS issue. I've been told that this issue has been addressed in CVS-HEAD, but have not personally tested this. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stun support
Hi * users, I want to know if STUN suport is available with Asterisk. Kindly let me know. I have posted this also in DEV list but none replied to me. thanks, Somesh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stun support
someshwarak wrote: Hi * users, I want to know if STUN suport is available with Asterisk. Kindly let me know. I have posted this also in DEV list but none replied to me. Short Answer: No. Longer Answer: No, and most people that think they need STUN don't actually need it. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stun codec
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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stun codec
I uses to have this when I enabled stun and did not need it On Tue, 2005-05-10 at 16:55, Ronald Wiplinger wrote: 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stun codec
You can use Ethereal to see what your phone (stun) is sending. Of this way you can see the RTP ports and IP public that your phones are going to use. You can see that information in INVITE and OK packets. For other hand, If you use one router with symmetrical NAT then Stun won't work http://www.networkmagazine.com/shared/article/showArticle.jhtml?articleId=17602009classroom= __ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN Server
Hi, Does asterisk have in itself an STUN server built in? Or do I need to set one up seperately? And if that is the case, what is recommended for use with asterisk (to allow VOIP users behind nats to connect to my VOIP servers) Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN Server
Hi! Asterisk supports NAT! http://www.voip-info.org/wiki-Asterisk+Avoid+SIP+NAT+Traversal http://www.voip-info.org/wiki-Asterisk+sip+nat+solutions /Madhawa On Thu, 17 Mar 2005 18:17:14 -0500, Matt [EMAIL PROTECTED] wrote: Hi, Does asterisk have in itself an STUN server built in? Or do I need to set one up seperately? And if that is the case, what is recommended for use with asterisk (to allow VOIP users behind nats to connect to my VOIP servers) Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended addressee, or the person responsible for delivering it to them, you may not copy, forward disclose or otherwise use it or any part of it in any way. To do so may be unlawful. If you receive this e-mail by mistake, please advise the sender immediately. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN
I have a SER server and an * server, both have private addresses and have static nat's on the router to the internet. I have installed STUN (by vovida) on the SER server by giving the SER server a second private address on a sub interface (which is probably not right). I understand I need a public address on the SER box, however is this the correct approach to getting it working for clients behind a router e.g broadband users ? Thanks ___ ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN and Asterisk? (Was: SER is a better NAT solution?)
STUN requires 2 NIC interfaces on the machine running the server right? And both interfaces need seperate public IP's right? 'And' the phones/ATA's need to support STUN right? I don't think the Cisco phones support STUN. -Matthew - Original Message - From: Brian Wilkins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, November 23, 2004 4:25 AM Subject: Re: [Asterisk-Users] SER is a better NAT solution? Addendum:LinksysWRT54G If you had 100,000 phones registering to the Asterisk server, I would think you would have at least two or three more Asterisk servers for people to point to their devices to. Who is to say that SER won't crash with 100,000 registrations either? You could always use a STUN server on each Asterisk box and that will work perfectly. On Tuesday 23 November 2004 03:13 pm, Matthew Boehm wrote: Yes, exactly! I think 100,000 phones all regeristing every 60 seconds would put quite a load. And if 50% of them are all behind NAT/FW, asterisk wouldn't play nice, would it? So, is SER a better option for this? -Matthew - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 22, 2004 5:38 PM Subject: Re: [Asterisk-Users] SER is a better NAT solution? Addendum: LinksysWRT54G Tracy R Reed wrote: On Mon, Nov 22, 2004 at 10:00:48AM -0500, Paul Rodan spake thusly: I am quite interested in this as well. I didn't realize registrations are the #1 cause of load on an asterisk server, we haven't gotten to that kind of usage just yet. I don't think they are, are they? How could a few registration packets per phone once an hour come anywhere near the load of a single sip session? I guess if you have 100,000 customers. Say 5-10% are making calls and yet all of them register? Maybe. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN and Asterisk? (Was: SER is a better NAT solution?)
You need two public ip addresses. I am running STUN on one nic. I don't know about the Cisco phones however. I have a Gradnstream and a couple of soft phones bouncing off my STUN server. Lyle - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, November 23, 2004 9:51 AM Subject: [Asterisk-Users] STUN and Asterisk? (Was: SER is a better NAT solution?) STUN requires 2 NIC interfaces on the machine running the server right? And both interfaces need seperate public IP's right? 'And' the phones/ATA's need to support STUN right? I don't think the Cisco phones support STUN. -Matthew - Original Message - From: Brian Wilkins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, November 23, 2004 4:25 AM Subject: Re: [Asterisk-Users] SER is a better NAT solution? Addendum:LinksysWRT54G If you had 100,000 phones registering to the Asterisk server, I would think you would have at least two or three more Asterisk servers for people to point to their devices to. Who is to say that SER won't crash with 100,000 registrations either? You could always use a STUN server on each Asterisk box and that will work perfectly. On Tuesday 23 November 2004 03:13 pm, Matthew Boehm wrote: Yes, exactly! I think 100,000 phones all regeristing every 60 seconds would put quite a load. And if 50% of them are all behind NAT/FW, asterisk wouldn't play nice, would it? So, is SER a better option for this? -Matthew - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 22, 2004 5:38 PM Subject: Re: [Asterisk-Users] SER is a better NAT solution? Addendum: LinksysWRT54G Tracy R Reed wrote: On Mon, Nov 22, 2004 at 10:00:48AM -0500, Paul Rodan spake thusly: I am quite interested in this as well. I didn't realize registrations are the #1 cause of load on an asterisk server, we haven't gotten to that kind of usage just yet. I don't think they are, are they? How could a few registration packets per phone once an hour come anywhere near the load of a single sip session? I guess if you have 100,000 customers. Say 5-10% are making calls and yet all of them register? Maybe. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN and Asterisk? (Was: SER is a better NAT solution?)
Sure they do. I have a bunch of Cisco phones that support STUN. On Tuesday 23 November 2004 03:51 pm, Matthew Boehm wrote: STUN requires 2 NIC interfaces on the machine running the server right? And both interfaces need seperate public IP's right? 'And' the phones/ATA's need to support STUN right? I don't think the Cisco phones support STUN. -Matthew - Original Message - From: Brian Wilkins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, November 23, 2004 4:25 AM Subject: Re: [Asterisk-Users] SER is a better NAT solution? Addendum:LinksysWRT54G If you had 100,000 phones registering to the Asterisk server, I would think you would have at least two or three more Asterisk servers for people to point to their devices to. Who is to say that SER won't crash with 100,000 registrations either? You could always use a STUN server on each Asterisk box and that will work perfectly. On Tuesday 23 November 2004 03:13 pm, Matthew Boehm wrote: Yes, exactly! I think 100,000 phones all regeristing every 60 seconds would put quite a load. And if 50% of them are all behind NAT/FW, asterisk wouldn't play nice, would it? So, is SER a better option for this? -Matthew - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 22, 2004 5:38 PM Subject: Re: [Asterisk-Users] SER is a better NAT solution? Addendum: LinksysWRT54G Tracy R Reed wrote: On Mon, Nov 22, 2004 at 10:00:48AM -0500, Paul Rodan spake thusly: I am quite interested in this as well. I didn't realize registrations are the #1 cause of load on an asterisk server, we haven't gotten to that kind of usage just yet. I don't think they are, are they? How could a few registration packets per phone once an hour come anywhere near the load of a single sip session? I guess if you have 100,000 customers. Say 5-10% are making calls and yet all of them register? Maybe. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN and Asterisk? (Was: SER is a better NAT solution?)
Matthew Boehm wrote: STUN requires 2 NIC interfaces on the machine running the server right? And both interfaces need seperate public IP's right? ' Why ever for? I realize that in order to set up a STUN server you need a public IP, but why two of them and why two different interfaces? Dazed and confused, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN and Asterisk? (Was: SER is a better NATsolution?)
I tried to setup MySTUN (http://developer.berlios.de/projects/mystun/) and it said STUN servers require 2 seperate public IPs on the same machine. (and yes, I realize I made a mistake when I said 2 NIC cards) -Matthew - Original Message - From: Gilad Ben-Yossef [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, November 23, 2004 10:24 AM Subject: Re: [Asterisk-Users] STUN and Asterisk? (Was: SER is a better NATsolution?) Matthew Boehm wrote: STUN requires 2 NIC interfaces on the machine running the server right? And both interfaces need seperate public IP's right? ' Why ever for? I realize that in order to set up a STUN server you need a public IP, but why two of them and why two different interfaces? Dazed and confused, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] stun and only one external ip
hi, i want to use mystun because off nat problems by more than one device behind one nat gw. i think it is the only solution to solve the nat problem. what i do not understand is why needs the stun server two ip addresses? thx for any hints. -- Thomas Küpper 01063 Telecom GmbH Co. KG Mottmannstr. 2 53842 Troisdorf Telefon: 02241-9434-506 Telefax: 02241-9434-846 E-Mail: [EMAIL PROTECTED] E-Mail: [EMAIL PROTECTED] Homepage: http://www.01063telecom.de --- Diese Nachricht ist vertraulich. Sie ist ausschliesslich fuer den im Adressfeld ausgewiesenen Adressaten bestimmt. Sollten Sie nicht der vorgesehene Empfaenger sein, so bitten wir um eine kurze Nachricht. Jede unbefugte Weiterleitung oder Fertigung einer Kopie ist unzulaessig. Da wir nicht die Echtheit oder Vollstaendigkeit der in dieser Nachricht enthaltenen Informationen garantieren koennen, schliessen wir die rechtliche Verbindlichkeit der vorstehenden Erklaerungen und Aeusserungen aus. Wir verweisen in diesem Zusammenhang auch auf die fuer uns geltenden Regelungen ueber die Verbindlichkeit von Willenserklaerungen mit verpflichtendem Inhalt, die in den bank- bzw. unternehmensueblichen Unterschriftenverzeichnissen bekannt gemacht werden. --- This message is confidential and may be privileged. It is intended solely for the named addressee. If you are not the intended recipient please inform us. Any unauthorised dissemination, distribution or copying hereof is prohibited. As we cannot guarantee the genuineness or completeness of the information contained in this message, the statements set forth above are not legally binding. In connection therewith, we also refer to our governing regulations of concerning signatory authority published in the standard bank or company signature lists with regard to the legally binding effect of statements made with the intent to obligate us. --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun and only one external ip
Thomas Kuepper wrote: hi, i want to use mystun because off nat problems by more than one device behind one nat gw. i think it is the only solution to solve the nat problem. what i do not understand is why needs the stun server two ip addresses? It needs 2 IPs because the server will attempt to contact the phone from these 2 IPs. If the phone responds from the 2 IPs it means that a 3rd IP (the phone you are calling) will also be able to communicate...hence do the whole NAT traversal thing. If the second IP is unable to communicate it means the phone is behind a Symmetric NAT. -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun server
AJ Grinnell wrote: What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary? I don't know which server is the best one, I'm using the one from IPtel.org. You need two IP addresses for STUN to work, with a proper implementation you will be able to run both on the same NIC. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun server
http://developer.berlios.de/projects/mystun/ works fine for me. Tested on debian sarge with one NIC and two IP addresses for this NIC. Klaus AJ Grinnell wrote: What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] stun server
What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun server
AJ Grinnell wrote: What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary? Asterisk does not require STUN. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun server
STUN can be nice when connecting to Asterisk behind NAT in some situations. X-Lite/Pro softphones, Grandstream Budgetones and a few other clients make great use of STUN. That being said, the only good (free) STUN server I've seen is the Vovida one that requires two NICs. It works very well, if that is any consolation. Brian --- Jeremy McNamara [EMAIL PROTECTED] wrote: AJ Grinnell wrote: What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary? Asterisk does not require STUN. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stun server
I just put multiple IPs on the same interface and use -a eth0:1 ip. Seems to work fine. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN command line client?
Hi...I'm trying to figure out the famous 3 tests that a STUN client uses for determining the kind of NAT that it is behind. Is there a command line client available to send binding requests to a known STUN server? I'm aware of the SourceForge ones. Either Linux or Windows is fine. Best wishes, -- Larry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN and Asterisk
OK, I've breifly looked at STUN and what it is and can do. First off it is NOT a way to punch UDP through a firewall. STUN offers a method to determine the firewall environment and find out just what is out there. But leaves it to Asterisk to determine what to do. The way it could be used within Asterisk: You would link in the STUN client library from www.vovida.org/ and then when Asterisk first fires up it would call the STUN library to see what kind, if any fire wall is up. It would store this information globally. Later inside chan_sip.c Asterisk could set up the packets correctly with pulic IP address if required. This would be VERY much like the two current patches do except that we would no longer need the new lines in sip.conf as STUN would figure this out for us. The other thing we could do is detect hopeless caes and rather then let the audio fall on the floor we could issue an error message saying something like UDP is 100% blocked no way to make this call and not even attemp it. Bottom line: STUN could save the user much configuration hassel but does noting that a very knowagable person could not figure out and then put into a *.conf file. But most people don't know if their NAT firewall for symetric for restricted cone. STUN can figure this out automatically. Notice that xten X-Lite already does the above. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN and Asterisk
Chris, snip OK, I've breifly looked at STUN and what it is and can do. First off it is NOT a way to punch UDP through a firewall. snip Bottom line: STUN could save the user much configuration hassel but does noting that a very knowagable person could not figure out and then put into a *.conf file. But most people don't know if their NAT firewall for symetric for restricted cone. STUN can figure this out automatically. Excellent analysis!!! Can I buy you a beer? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN server from Vovida
Sorry to answer a question with a question..
Can stund and * be loaded on the same server and run at the same time?
Later..
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 where another stund server is running on port and
alternate
# port.
STUNDALTERNATEHOSTNAME=127.0.0.1
# The primary response port to user
STUNDPRIMARYPORT=3478
# The alternate port to use
STUNDALTERNATEPORT=3479
STUNDARGS=-h ${STUNDPRIMARYHOSTNAME} \
-p ${STUNDPRIMARYPORT} \
-a ${STUNDALTERNATEHOSTNAME} \
-o ${STUNDALTERNATEPORT}
Any ideas? Any suggestion for another STUN server?
--
Paul
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Re: [Asterisk-Users] STUN server from Vovida
On Wed, 2003-09-03 at 09:01, WipeOut . wrote: Sorry to answer a question with a question.. Can stund and * be loaded on the same server and run at the same time? I've also never been able to figure out stund, if that is possible wouldn't it be the answer to most of the SIP difficulties. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN server from Vovida
The client device has to support stun. Bugetones do, ATA do, 7960 don't..etc Dave Cotton wrote: On Wed, 2003-09-03 at 09:01, WipeOut . wrote: Sorry to answer a question with a question.. Can stund and * be loaded on the same server and run at the same time? I've also never been able to figure out stund, if that is possible wouldn't it be the answer to most of the SIP difficulties. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN server from Vovida
I would suggest you use the Vovida STUN CVS Version. It worked fine on our
RedHat 7.3. Look though the Vovida mail archives to learn where it is
exactly. And no...this is not a fix for all the NAT troubles, STUN does not
work with Symmetric NATs. But the good point is that Symmetric NATs are a
minority.
Regards,
Andres
On Wednesday 03 September 2003 00:51, Paul Lambert wrote:
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 where another stund server is running on port and
alternate
# port.
STUNDALTERNATEHOSTNAME=127.0.0.1
# The primary response port to user
STUNDPRIMARYPORT=3478
# The alternate port to use
STUNDALTERNATEPORT=3479
STUNDARGS=-h ${STUNDPRIMARYHOSTNAME} \
-p ${STUNDPRIMARYPORT} \
-a ${STUNDALTERNATEHOSTNAME} \
-o ${STUNDALTERNATEPORT}
Any ideas? Any suggestion for another STUN server?
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Re: [Asterisk-Users] STUN server from Vovida
Only hard phones with STUN support that I am aware of are Grandstream and SNOM. I am sure the Cisco ATA186 does not support STUN. On Wednesday 03 September 2003 09:11, James Sizemore wrote: The client device has to support stun. Bugetones do, ATA do, 7960 don't..etc Dave Cotton wrote: On Wed, 2003-09-03 at 09:01, WipeOut . wrote: Sorry to answer a question with a question.. Can stund and * be loaded on the same server and run at the same time? I've also never been able to figure out stund, if that is possible wouldn't it be the answer to most of the SIP difficulties. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN server from Vovida
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 where another stund server is running on port and
alternate
# port.
STUNDALTERNATEHOSTNAME=127.0.0.1
# The primary response port to user
STUNDPRIMARYPORT=3478
# The alternate port to use
STUNDALTERNATEPORT=3479
STUNDARGS=-h ${STUNDPRIMARYHOSTNAME} \
-p ${STUNDPRIMARYPORT} \
-a ${STUNDALTERNATEHOSTNAME} \
-o ${STUNDALTERNATEPORT}
Any ideas? Any suggestion for another STUN server?
--
Paul
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