How to set up calls: India--VOIP-->California--POTS-->

I want to allow our single consultant in India to use our single standard telephone line in California to place outgoing calls in the U.S. We both have broadband internet access. I am not interested in paying any additional subscription fees such as monthly fees. Independence from any proprietary or non-standard service providers is important to me as I have noticed many of them going out of business recently. Ideally the solution is still useful in 10 years. We currently own 2 linksys RV042 on our end, and gigabit linksys cards on our internal network. We can probably scrounge up some modems from the pre-broadband days if they are of any use, but don't mind paying a little more in fixed costs if it means a higher quality voice connection. Finally, it would be nice if any required hardware parts were available to us quickly - ideally from a local store.

Please advise!

Greetings dramastic and welcome to VoIPUser forum.

What you want can be achieved with two Analag Telephone Adapters (ATA) where one will have a built-in FXO port to be put in California as your PSTN gateway and the other with an FXS port in India where you can connect a (cordless) phone unit to it. A typical ATA with a built-in FXO port is Linksys SPA3K and a typical ATA with a built-in FXS port is Linksys PAP2T-NA.

A Linksys 3102NA will do the job at the USA end and seems to be reliable (I have one doing roughly what you are asking). On the other ends I have Grandstream HT286 ATA or BTone 102 desk phone. The PAP2 could be used at the distant end but I have no experience of them.

Thanks for the feedback guys, I'll definitely look into these suggestions as possible options for us. I'm fairly good with computers in general, but have no experience at all with this particular technology. All the acronyms and specialized usage of words are going to require some effort on my part to understand. Now I know where to start.

Without loss of quality, are there also implementations available that are more software based on the India side? I mean like using a standard computer headset and piping through Skype or Microsoft's telephony service or some other software? Our consultant lives in a pretty remote area (despite having broadband) and it could take some time to get specialty hardware to him.

dramastic :

I mean like using a standard computer headset and piping through Skype or Microsoft's telephony service or some other software?

You can try Asterisk PBX system.

I guess he could just use a soft client such as X-Lite or the Pro version but a computer would have to be running.

First off, thanks for the tips so far guys!

Ok, so I've purchased a couple of SPA3102-NA "Unlocked" to play with. For this scenario, I'm only using one of them. The other is still in its box unpacked. I have one set up on my LAN now. I've also installed X-lite on a computer on the LAN.

SPA3102-NA ("WAN") IP address is 192.168.1.50
Computer with X-lite IP address is 192.168.1.10

(Note the SPA3102 WAN port is actually connected to my LAN at the moment).

The telephone line is connected to the "line" and phone to "phone".

I'm trying to dial the SPA3102 from X-lite. I set up the X-lite dial plan to read:
123T;match=1;replace=@192.168.1.50:5060;

When I dial 123, and pause for it to dial out, it just gives me the "beep Beep BEEP.. we're sorry.." message.

I'm sure if I play with it long enough, I'll figure it out, but thought I would check with you guys incase I missed anything obvious.


I've referenced bits and pieces of the following sources:

http://www.provu.co.uk/pdf/sipura/spa_backtoback_2x_spa3000.pdf
http://www.provu.co.uk/pdf/sipura/LinksysSPAAdminGuidev2.0.11-16.pdf
http://www.counterpath.com/docs/X-Lite3.0_UserGuide.pdf
http://www.voipuser.org/index.php?name=PNphpBB2&file=viewtopic&t=9639&postdays=0&postorder=asc&highlight=spa3102+xlite&start=0


Dramastic

Probably the simlest way to get some success is to first make sure that X-Lite will log into voipuser.org and see that the echo test works.
If that works and you have registered for outgoing pstn then you could try an outgoing call. Shut down X-lite and do the same test for one of the 3102 devices. Next get an account with another free carrier, say, Freeworld Dialup or Voxalot(Basic service) and try calling with X-lite registered to voipuser and the 3102 registered to the other.
I expect there will be problems and you will then have to set about forwarding ports from your server to the 3102 and/or set up STUN. This is the set up that I have running now. My nephew has a 3102 in anther town and we are able to call eachother directly although we normally go through Voxalot for simplicity.

I have sent you a pm.

Wow,

I had no idea that I would need to involve any outside service providers to do this. What if I didn't have internet access? I was hoping to keep this confined to my LAN for the moment at least!

Can't I just do direct IP dialing from X-lite instead of one of the SPA3102 as suggested by the "Sipura back-to-back Config" link? I guess I just don't see what the third parties have to do with my phonecalls..

Dramastic

dramastic :

What if I didn't have internet access? I was hoping to keep this confined to my LAN for the moment at least!

I believe what you want is to configure an Asterisk PBX System within your LAN and have both your SPA devices and/or softphone registered to this system. This way, you can fasilitate the PSTN line from your softphone through this Asterisk PBX System.

You can call IP to IP with the 3102 but it is normally best to make sure that everything is working correctly first. That is what I was suggesting you do. When that works you can worry about IP to IP then. Voipuser is handy for doing these tests!

Actually I ended up finding this set of instructions:

http://forum.voxilla.com/linksys-sipura-voip-support-forum/spa-softphone-20117.html

That allowed me to use SJphone to connect to the SPA3102 using ip dialing. Still not sure how its done with x-lite, but at least I got something working now

I guess both your suggestions are ways to go. I just felt uncomfortable adding several layers of complexity (Asterix or NAT port forwarding + 3rd party providers) in order just to prove this on my local LAN.

Anyway, I really appreciate all your help. Without it I would not have been able to make this a reality.

Dramastic

dramastic :

I just felt uncomfortable adding several layers of complexity (Asterix or NAT port forwarding + 3rd party providers) in order just to prove this on my local LAN.

For what you want, you don't need to deal with any NAT because everything is within your LAN, except if you want to include I/O from VoIP line.