Can you please enlighten me in layman/noob terms of what is the function of the voip feature with 2x tel. ports on the Billion 7800vdpx router?

Thanks~! Appreciate it.

Okay, here's the lowdown on the VOIP RJ-11 jacks you'll find on many a router these days - including the ones for FTTH use. Essentially, they're an integrated version of the ATA, or Analog Telephone Adapter.

What does an ATA do? Basically, the idea is that with VOIP, instead of using the copper telephone wire to carry voice signals, the sound recorded by your microphone is converted into a digital form (the Voice) and transmitted via the Internet (Over Internet Protocol). At the other end, the digital waveform is converted back into a voice signal and carried via PSTN (the telephone network). If the person on the other side is also using VOIP, then it will be converted again into a digital format until it reaches their computer, where it will undergo the final conversion back to a voice signal and piped out through their speakers. At this point, you may have noticed that a telephone handset has both a speaker and a microphone, and is, after all, meant for telephone calls. So instead of using software on your computer to do the conversion, and then you need a mic and speakers, why not just use back your existing phones? An ATA is basically a device that allows you to plug your phone in one end and an Ethernet cable to your router in the other end. It does the conversion from a phone signal to a digital form and sends it out to your VOIP provider via the Internet, and vice versa.

Think of an ATA as the exact opposite of the old 56k dialup modems. A modem converts the digital data from your computer into voice data, which it then sends across the normal telephone network to the dialup server, where another modem converts it back into digital data. The ATA takes voice data, turns it into digital data, and sends it across the Internet to the VOIP server, where another ATA converts it back into voice data.

That's it, really. In practice, with this kind of router, which has built-in ATA functions, what you need is to sign up with a VOIP provider. They may give you an actual phone number or not, but they will definitely give you a SIP username and password, as well as various config options (SIP & STUN server, codecs supported etc.) - you plug these into your router's VOIP settings. Then you plug in a phone and you're done. The point is to make it seem as if it really is a normal fixed line service.

In your case, it seems that the Billion router has a *third* RJ-11 jack for an *actual* fixed line service, according to the writeup. The idea is that you plug in your TM phone line into this jack, so that in the event your Internet goes down, you can still call out and receive normal telephone calls on the same phones you're using for VOIP. Assuming you still have a fixed line service, of course.

Just buy Linksys SPA3000 SPA-3000 as your FXO gateway to your UNIFI VOIP phone, I know that the intention is to utilizing TM's free calls to TM network.

Last time I have setup an asterisk (dial plan rules) to connect to my Linksys SPA3000 SPA-3000 for Unifi for SIP trunking (TM-to-TM free calls), International calls to voipstunt as SIP trunking (Free HK, JP, Sing etc), trunk to YES VOIP for 018-018 free calls and Google voice US number (free US-to-US dialing with 1 US #).

Now the Linux Asterisk motherboard is spoiled, still not yet bring the box up.

This post has been edited by eddie_lim: Aug 25 2014, 10:54 AM

Hello great people..

I'm totally new to this VOIP thing. Was configuring my CM11-based phone and found this SIP feature and was wondering what it is all about....

I need to key in a username,password,sip server... and then what? I can make calls to the world for free??

Outgoing call is subjected to call rate of VOIP provider. Usually lower than std call. Some voice plan got fixed rate, like xx sen/call.. even free call with fixed monthly.

Hey guys,
I'm looking for provider that able to provide Indonesian DID. Would be great if someone can point me to something.
I just need SIP trunk and DID, to add to existing PBX system.

Cool, can this Linksys SPA3000 SPA-3000 do VLAN? Thanks.

Hi NightShyamalan,

Do you mind to share who is your sip trunk provider?

99% of of my company calls are outbound to mostly African countries.

Thank you in advance...

Nope, you can do VLAN by connected to a VLAN switch

Ok, can it configured to have 2 VoIP account? Thanks.

What is your ultimate goal here?
my previous setup was to use the SPA3000 as one of a SIP trunking for 03 number (because it is free) for my asterisk

For asterisk, you can setup unlimited account (SIP extension, aka VOIP account)

FYI, SPA3000 has only 1 FXO and 1 FXS port

This post has been edited by eddie_lim: Oct 21 2014, 09:59 AM

My goal is to have free call between UniFi VoIP and Maxis Fiber (HSBB) VoIP using only one SPA3000 at each end.

Thanks.

Then u need to have 2 SPA3000, each connect to each VOIP-RJ11 port; 1 asterisk via these 2 ATA devices as SIP trunk; and create calling rules yourself inside the asterisk.

that's all.

This post has been edited by eddie_lim: Oct 21 2014, 11:52 AM

2 SPA3000 on each location?

If you wanted to to use in 2 different location u must have 2 ATA.

Thanks

is it possible in theory that i can use a sip phone with the Unifi number?

i have the password, but i do not have the sip detail

appreciate if someone here can provide a hint or a guide on this, thanks

thanks

Mitodna,

The unifi number comes to you home is already using SIP platform.

For those who wants a minimum device .. try

http://www.amazon.com/Grandstream-GS-DP715...ywords=IP+Phone

Grandstream GS-DP715

I have one, configured with ActionVOIP, I allow my maid to use it. She is from India she enjoy calling long. Its cheap, complication of VOIP avoided. No need to maintain a line. all good. Its merely 5 cents per min.

Is your company provide service to individual configuration of Asterisk or FreePBX system ? I am trying to build one but keep failing due to less knowledge