similar to: ipPhone and ATA with UPNP

fauno <fauno at kiwwwi.com.ar> writes: > Etienne Dechamps <etienne at edechamps.fr> writes: >> (I realize that this means UPnP support could possibly be achieved >> simply by suggesting that the user spawn some standalone UPnP client >> process in the background from the tinc-up hook. That's not very >> user-friendly, though. Especially on Windows.) >

On Wed, Nov 11, 2015 at 09:04:20PM +0000, Etienne Dechamps wrote: > Unfortunately, I often notice that even with the advanced UDP hole > punching mechanisms that tinc uses, I still come across cases on my > network where two nodes behind NATs cannot talk to each other because > one of the nodes is behind a "strict" NAT [...]. This type of NAT > will stop any UDP hole

On 11 November 2015 at 21:57, David Nicol <davidnicol at gmail.com> wrote: > it is entirely possible to write code that uses threads on Win32 and forks > on POSIX by abstracting the communication bits generically. Signalling could > work over pipes on both. > > https://msdn.microsoft.com/en-us/library/windows/desktop/aa365152(v=vs.85).aspx Hum... yes of course, but I

Hi Guus, Unfortunately, I often notice that even with the advanced UDP hole punching mechanisms that tinc uses, I still come across cases on my network where two nodes behind NATs cannot talk to each other because one of the nodes is behind a "strict" NAT that won't use consistent source ports for UDP packets that are sent to different destination addresses. More formally, this

On 12 November 2015 at 21:29, Guus Sliepen <guus at tinc-vpn.org> wrote: > On Wed, Nov 11, 2015 at 09:04:20PM +0000, Etienne Dechamps wrote: >> According to an online NAT check service, around 50% of NATs in the >> wild have this problem, making this a very real issue: >> http://nattest.net.in.tum.de/results.php > > Are you referring to port restricted NAT in the

On 12 November 2015 at 01:15, fauno <fauno at kiwwwi.com.ar> wrote: > Etienne Dechamps <etienne at edechamps.fr> writes: >> (I realize that this means UPnP support could possibly be achieved >> simply by suggesting that the user spawn some standalone UPnP client >> process in the background from the tinc-up hook. That's not very >> user-friendly, though.

This week, I've been really busy with the launch of a new Swedish Voip provider, www.bbtele.se, so I haven't been able to follow the Asterisk community and haven't been very responsive either. My apologies if you've tried to contact me and I did not reply quickly or at all. So to cover up (can't report on what is happening :-) I dedicate this issue of Asterisk Sunday News to

On Wed, Nov 11, 2015 at 3:04 PM, Etienne Dechamps <etienne at edechamps.fr> wrote: > miniupnpc only provides a blocking API, there is no way to > integrate it into an event loop. That pretty much means we will need > to run the UPnP code in a separate thread. Which means using pthreads > on POSIX, and then we would either have a dependency on pthreads-Win32 > on Windows, or

Etienne Dechamps <etienne at edechamps.fr> writes: > (I realize that this means UPnP support could possibly be achieved > simply by suggesting that the user spawn some standalone UPnP client > process in the background from the tinc-up hook. That's not very > user-friendly, though. Especially on Windows.) something like this :P

Hi Ray, I was wondering if the "qualify" option is used [in sip.conf] to keep a connection (from the SIP phone inside the firewall to the Asterisk server outside the firewall) open then would the firewall not allow two way communication without incoming port mapping/NAT (providing that the SIP phone started "talking" first)? I'm not sure about that - I'm being

Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",

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

Has anyone successfully got UPnP and Shorewall working together ? If so, has anyone created documentation on how to get it working ? Thanks in advance, Mark

Hi, I''m trying let the windows clients behind my leaf ( bering rc4) talk through msn messenger. I found linux-igd.sourceforge.net that seems to be the missing piece... I used the ideas from www.wix.net.nz/LEAF/glibc.html to build a leaf cd using the redhat 7.3 glibc ( It''s alot easier to test applications because I can simply copy the binary ( and the .so) on the bering

I'm trying to run Asset UPnP media server under Wine 1.1.27 on Ubuntu 8.04. The problem is that media clients on the same network don't appear to SSDP discover this media server. The server itself *does* advertise itself fine on the network though (but as this is fairly infrequent, then any clients can be hanging around for quite some time before they spot the server). I've tried

I decided to give the whole family IP phones for christmas, all hooked into my asterisk server, so all the nephews can have their own lines. However, one of the phones I got was the SNOM 200. That's worked fine for me on my own network, but I'm having bad luck getting it to work behind NAT talking to Asterisk. It talks to my termination/origination provider, which seems to ruthlessly

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

Dear Pals I think all of us in this business (or most) looking at the Corn Flakes Back Side , Wish to find a NAT Friendly VOIP solution. Thats ridiculous I know, but what about this wish list (Personal Point of view) Upnp support for Asterisk STUN Support for Asterisk Small Embedded Asterisk (just to redirect IAX) SIP-IAX , h.323-IAX or something like that , on the End Point. As a Nat

Hello In case Asterisk is used in a private LAN behind a firewall while allowing remote SIP clients to connect from the Net, we must open UDP5060 for SIP and a range of UDP ports (as set in rtp.conf) so let incoming voice packets. Provided the user doesn't have access to the firewall (eg. corporate or hotel), and the firewall doesn't allow dynamic port opening through UPnP or NAT-PMP...

Im having a little problem with voicemail and my cisco phones i was wondering if anyone might have seen this before and let me know whats going on. it spits this out and then my cisco ip phone reboots im using the latest cvs and a cisco 7910 phone WARNING[1234379840]: File res_adsi.c, Line 205 (__adsi_transmit_messages): Unable to send CAS -- Playing 'vm-login' (language 'en')