similar to: Known SIP - NAT Solutions?

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

I have been experiencing a rather odd issue with my (newly connected) T100P card. It is installed in my Asterisk server, which is working great with SIP clients. The card is connected to a channelized T-1 line (24 channels). The odd bit is that outgoing calls (Asterisk --> T-1 --> PSTN) works just fine. However, incoming calls (PSTN --> T-1 --> Asterisk) are dropped after about 20

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.) >

Is there a way to record your own voice messages ("Welcome to my PBX, Press 1 for ...") using asterisk and an analog phone, or do they need to be recorded using traditional voice recording software? Also, I am confused as to why my replies to the message board are never indented. Call this the ultimate newbie question, but how should the reply be worded so that I don't screw up

I'm trying to understand IAX and whether or not it would solve my difficulties: 'The primary goals for IAX were to minimize bandwidth used in media transmissions, with particular attention drawn to control and individual voice calls, and to provide native support for NAT (Network Address Translation) transparency. Another goal is to be easy to use behind firewalls.'

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

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

I'm looking for anyone who has some vendors for rackmount PC hardware in a fairly decent configuration for Asterisk/Digium T1 termination. I am looking for well-built hardware from an integrator that understands correct component choice. Preferably, I'd like SCSI but I might be able to do Ultra-IDE platforms. I need at least two PCI slots in the chassis (so 2u/3u size factor is OK.)

Hello, Let me first start by saying Shorewall is awesome, and I use it everywhere from single box firewall, to home network firewall, even to our corporate firewall. I am experiencing a problem getting my home firewall to work with my BroadVoice VoIP connection. I use the Sipura SPA-2100 ATA (Analog Telephone Adapter) that came with my BroadVoice account. This happened when I tried to replace

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

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

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",

Recently, Windows 2000 SP4 clients have developed a problem when using a network printer connected to a Samba print share. Some applications are OK. But take, well, notepad, for example. Selecting a local printer is normal, but there is a two minute delay (!) in selecting a network printer. Once selected, all is well. All smb operations are frozen until the selection happens. This is

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

Should the following setup work? SIP UA---NAT---Internet---NAT---SIP UA If both UA's support STUN and report the external IP address in the SIP packet.. I am trying to get away from using canreinvite=no so that traffic can go directly between the UA's and not via the central server but I can't seem to get it to work.. Has anyone set this up and can give me some pointers??

hi We are using snome 200 IP phone with *. It works OK. But after a period of time we can't hear any sounds for any icoming or outgoing calls. I've got two of these phones. Same symptoms occur to both of these( not at the same time ) and the problem remains until the phone is completely rebooted. Don't know whether this's * or Snomes' prob. Any help would be appreciated.

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

Quick question re. NAT traversal. I understand how sitting behind a NAT could cause problems for a SIP UA. The SIP UA would create SIP mesages using IP addresses from inside the network (i.e. 192.#.#.# or 10.#.#.#) and these IP addresses are of course unnavigable for the recipient. What I don't get is why don't web browsers suffer the same problem? A web brower behind a NAT sends an

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

I apologize in advance for the simplicity of this question. I use R 2-3 times a year, and I seem to forget more in the intervening months than I learn during my days of panicked reading.... I HAVE tried looking at the help resources; I'm just not very good at understanding them. I have a dataframe with 18 observations of 5 different things, and a second dataframe with and estimate for those