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  #1 (permalink)  
Old July 15th, 2007, 09:04 AM
vya vya is offline
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Default Planning the implementation to interconnect 3 stations. Looking for advice.

Dear all:

After reading diverse posts on this forum - and as a newbie on the world of the VOIP - I would like to get your advise in order to check I'm planning correctly my solution.

I want to connect three places in different cities. Each is equipped with the following:

* Linksys SPA-3102
* PSTN line with unlimited local calls (inside the city only)
* High speed Internet access with dynamic IP

What I would like to achieve:

1. Every location can call any of the other locations via the Internet (great if all three can join in a conference).

2. Every location can receive calls from other standard SIP phones/stations on the internet (duh!)

3. One station can reach any of the others to place phone calls using the remote PSTN lines.

4. (Optional) The stations - using Skype or Yahoo accounts - can place international calls using the attached old telephone to the SPA-3102 (So no computers are required to do this and I can share the same account for all the stations)

Here is what I have thought I should do in order to allow the above described:

1. The devices should be registered somewhere on the net, so every other can known how to reach them. I guess that for this, I should create a SIP account for every location and when I put this information on the SPA-3102 it will register itself on the internet at the time it connects to the internet or when the connection is lost for some reason (which changes its IP on the net).
For this, I have alternatives like sipbroker.com, freeworlddialup.com and voxalot.com.

2. Optionally, I can register every SIP number related to the PSTN number. e164.org is a good place to do it.

3. I should look at the device manual, how to allow remote calls. But to prevent unauthorized calls, I should configure a PIN code.

4. Regarding the Skype/Yahoo use, I have no idea how to do it. There is a service that allows to call users of many popular IM: gtalk2voip.com But only for the users already on their lists, not PSTN numbers. Seems like I should use the computer if I want to use only those services, or purchase credit for the SIP account I register on every device.

Please, any feedback will be highly appreciated. I want to be sure I'm pointing in the right direction. I'll post my findings here.

Thank you!
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Old July 17th, 2007, 04:57 AM
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DracoFelis DracoFelis is offline
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Default Re: Planning the implementation to interconnect 3 stations. Looking for advice.

Quote:
Originally Posted by vya View Post
I want to connect three places in different cities. Each is equipped with the following:

* Linksys SPA-3102
* PSTN line with unlimited local calls (inside the city only)
* High speed Internet access with dynamic IP

What I would like to achieve:

1. Every location can call any of the other locations via the Internet (great if all three can join in a conference).

2. Every location can receive calls from other standard SIP phones/stations on the internet (duh!)

3. One station can reach any of the others to place phone calls using the remote PSTN lines.

4. (Optional) The stations - using Skype or Yahoo accounts - can place international calls using the attached old telephone to the SPA-3102 (So no computers are required to do this and I can share the same account for all the stations)
1, 2, and 3 are "easy" (relatively speaking) to accomplish. However, #4 is going to be a royal PITA to do, because both Skype and Yahoo are designed (by the companies that run them) to be "closed systems" that don't interact with other systems well at all!!!

In any event, here's how I would set it up if I was doing it:

1) Sign up for SIX "free" VoIP accounts (two for each adapter). Pretty much any of the "open" free VoIP systems will work, so choose the service you like the best. Or if you want ideas, you might try either Voxalot - Home or FWD | Home (both of which I personally use).

2) Setup one free VoIP account as your main "Line 1" provider on the SPA-3102, and the other account as your "PSTN provider".

3) Hook up the PSTN line to the "Line" jack of your SPA-3102

4) Configure the dial plan and "ring through" to allow you to use the PSTN line locally.

5) Configure the VoIP to PSTN features of each adapter, to allow you to call the PSTN side VoIP provider, enter a pin/password, and then dial out via the PSTN line.

6) I would also setup outbound dialing via SIPBroker - Login , to give you the maximum flexibility to make free VoIP to VoIP calls from these adapters.

7) (optionally) setup the adapter's "gateway" accounts (and properly configure the dial plan) to allow you to use some other commercial VoIP provider for making low cost outbound VoIP calls. As already mentioned this won't work with either Skype or Yahoo (due to the limits imposed by the companies who run those two services), but it will work with a lot of VoIP companies out there.


If you set this up correctly then you get most of what you want:

A) You can call each other VoIP to VoIP (for "free") by just calling the "number"/account assigned on the "Line 1" side of the adapter you want to reach. Likewise others on the internet (who know the "number" of your free VoIP account, and can dial it VoIP to VoIP) can also reach you for free.

NOTE: You could also use ENUM to make the above a little more "user friendly". However, ENUM is not needed to do this (it just makes the VoIP to VoIP "dialing" a little more "intuitive" to people who already know your PSTN number).

B) If the local PSTN line isn't already in use (remotely), your dial plan should let you dial out on it. And when the local PSTN line "rings", you can answer it. So you have full access to your local PSTN line.

C) If your PSTN line isn't already in use, it's possible for any of the other 2 adapters to use it "remotely", by dialing the SECOND VoIP account on each adapter, entering the proper pin code, and then dialing out normally. And you also have the option to restrict the dialing pattern so only the local/free calls can be dialed remotely...

D) For 3 person/adapter "conferences", the easiest solution would probably be for one person to call (for free using VoIP) each of the other two people/adapters as part of a 3-way call (warning: the station connecting the two others needs TWICE normal bandwidth, because they essentially have 2 calls at once). Or, you could just have everyone "call" a free conference/chat service (for example, the free "conference rooms" run by SIPphone/Gizmo).

E) For free VoIP to VoIP (outbound) calling, SIP Broker gives you a HUGE amount of flexibility. Alternatively Voxalot - Home has SIP Broker dialing features already built in.

F) And finally, while it's major work to get Skype or Yahoo to integrate to this, there are still MANY VoIP companies/plans that would integrate very easily, with various costs and features. For really light usage, pay by the minute plans make a lot of sense (and many of them don't have any fixed monthly costs, just the per minute usage costs). Conversely "unlimited" plans make sense for heavy phone users. And there are even "fixed minutes per month" plans in the middle. The point is, get the plans (that will work on your 3102s) that make sense given your situation and budget, and add those account(s) to your VoIP adapter (so they are yet one more dialing option).
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Old August 18th, 2007, 04:20 PM
rookiebot rookiebot is offline
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Default Re: Planning the implementation to interconnect 3 stations. Looking for advice.

Hi,

Im in a mea country where voip is not permitted by the telecom monopoly- but has no written law about it. In this scenario I need to set up a system connecting to my home country to make inexpensive calls. Since I'm a noob at voip this is what I came up with - Get dsl connections >=256kbps in either countries, set up a router to router vpn access between either locations(isp will be giving only dynamic ips), use draytek vigour 2800vgi and use the voip hotlines functionality to call from one point to another.

Problem being I will be able to call only one location. I saw the earlier post about SPA3102. How can I make use of this product to dial out on the local pstn in my home country. I cannot use any voip/sip service providers in mena as they are all blocked regularly. Hence the VPN, VoipHotline to router scenario.

Do advice

Regs
rookie
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Old August 18th, 2007, 04:31 PM
hwittenb hwittenb is offline
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Default Re: Planning the implementation to interconnect 3 stations. Looking for advice.

Quote:
Originally Posted by rookiebot View Post
set up a router to router vpn access between either locations(isp will be giving only dynamic ips)

Problem being I will be able to call only one location.
If you have a vpn connecting your two routers you should be able to access a 3102 behind the distant router using the local NAT ip address of the 3102. This will allow you to dial out on a pstn line attached to the 3102. You could also call a handset attached to the 3102's line 1 to talk to someone at that location.

Some people find it challenging to make the necessary settings on the device. Actually they are pretty straight forward, however I would recommend that you get it working at your home location first in order to familiarize yourself with what is necessary to make it work.

You can deal with the changing ip addresses between the two routers by using a DDNS service such as DynDns or no-ip.com.

Last edited by hwittenb : August 18th, 2007 at 04:47 PM.
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Old August 18th, 2007, 06:36 PM
rookiebot rookiebot is offline
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Default Re: Planning the implementation to interconnect 3 stations. Looking for advice.

more than the challenging factor, its which is the right product to invest in that kept me looped.

Quote:
Originally Posted by hwittenb View Post
This will allow you to dial out on a pstn line attached to the 3102. You could also call a handset attached to the 3102's line 1 to talk to someone at that location.
This is what I can figure regarding your expln & im partly ok with it:
??-->VPNRouter1<-->VPNRouter2-->3102-->Route to PSTN/Analog Phone.
The part i dint get is - how does voice or using what device(without 3rdparty sip provider) do I call the 3102 behind Router2 ? Do I need another 3102 at router1 pointing/routed directly to 3102 behind router2?
I'm not much into voip/3102 hence the confusion.

Do recommend some strong vpn/dsl modem/router as well for long term unattended usage. So far I was stuck up with Draytek Vigor 2800vgi coz of its voip hotline feature with 2 units.
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Old August 18th, 2007, 06:36 PM
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  #6 (permalink)  
Old August 19th, 2007, 01:51 AM
hwittenb hwittenb is offline
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Default Re: Planning the implementation to interconnect 3 stations. Looking for advice.

Quote:
Originally Posted by rookiebot View Post
The part i dint get is - how does voice or using what device(without 3rdparty sip provider) do I call the 3102 behind Router2 ? Do I need another 3102 at router1 pointing/routed directly to 3102 behind router2?
Behind your home router you need a Sip device that you can configure yourself. Most any of the unlocked Linksys/Sipura adapters (3102, PAP2T, etc) would do the job (stay away from the PAP2v2 which is sold on EBay, that's a special case). You could also use a softphone on a PC such as SJPhone, XLite or Ekiga. Looking at the Draytek unit that you referenced, I believe you could setup the voip adapter on that unit to call a remote 3102.

Quote:
Do recommend some strong vpn/dsl modem/router as well for long term unattended usage.
I'm not that familiar with the off the shelf routers that support vpn. I know there are a number of them. I use 3-party DD-wrt software on my router and it works well for voip. DD-wrt supports vpn, but I have no experience with it. You can install dd-wrt on a variety of routers with one of the Buffalo brand models being the current favorite. If you used one of these routers, though, you would also need to have an adsl modem.

The Draytek unit you mentioned is manufactured and sold for the European market and I have no experience with it. I looked at the specs and the user guide for the Draytek router. It looks like it is a combined ADSL modem, router, and voip adapter all in one unit which would be an advantage. It also supports Dynamic DNS (DDNS) which you will need to use if your ip address is changing over time. In its internal voip adapter, the 2800vgi will support two FXS interfaces to attached analog telephones. It does not have an FXO interface which is needed if you wish to interface a pstn line. If you want to use the Draytek unit and also the 3102 you should be able to attach the 3102 to this router and use the 3102's voip-to-pstn gateway feature. The 3102 would have its own unique address on the local NAT.
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Old August 19th, 2007, 12:29 PM
rookiebot rookiebot is offline
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Default Re: Planning the implementation to interconnect 3 stations. Looking for advice.

so i guess the following will be the final setup ive to go with -
AnaloguePhone ---FXSPort---> DrayTekRouter1 <--- VPNTunnel ---> DraytekRouter2 ---> SPA3102 ---FXOPort ---> PSTN

That's by configuring the FXS Port of Draytek1 to Call the SPA3102. Can you check the live configuration demo page of draytek 2800 to be sure i can reach/configure the fxo of spa3102 from the fxs of draytek? Like you said I'm not so confident abt this particular configuration.
Live demo : DrayTek Vigor2800 series

The whole setup should cost around 700$.. so I wanted to be sure before buying all the equipment.

Btw whats the difference between locked/unlocked SPA3102 or similiar devices?
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Old August 20th, 2007, 02:44 AM
hwittenb hwittenb is offline
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Default Re: Planning the implementation to interconnect 3 stations. Looking for advice.

Quote:
Originally Posted by rookiebot View Post
Btw whats the difference between locked/unlocked SPA3102 or similiar devices?
Voip Providers often sell and frequently require that you use "locked" ata adapters. Sometimes the adapters are provided at a price below their cost. When an adapter is "locked" you cannot access the settings to make any changes. This is done to reduce the provider's customer support costs and to inhibit your using the device with a competitor. Providers that do not require you to use a locked adapter are typically called "BYOD" providers where you can make the required settings to your own adapter. In the example we are discussing you are not using a voip provider.

You should know at the outset that what you are doing is not just plug it in, power it up, and go. If you are not comfortable using computers, setting parameters, reading user guides, using diagnostic tools, then you will either become comfortable or my guess is you will find the whole experience very frustrating. You apparently have a strong incentive to proceed or else you would not even consider the endeavor. The internet service provider that presents this barrier does not want you to succeed and may present additional roadblocks.

I cannot say that it will work because I have never seen a Draytek router and I haven't setup direct ip dialing over a VPN.

There are two different ways to configure the 3102 using direct ip dialing to access its voip-to-pstn gateway. You can configure it with no or pin authentication and it will return a dial tone to the voip caller. The other way to configure it is with "http digest" authentication and in this mode you dial the outgoing pstn number directly on the calling telephone. The latter method is more reliable because you do not have to send the dtmf digits over the broadband network which sometimes is a problem. With the latter method you send the digits as part of a sip invite using the sip signalling protocol.

With both techniques you need to determine the local network address of the attached 3102. You need to develop a technique whereby the router that the 3102 is attached to will always give the 3102 the same (static) address. I am not referring to the external internet address, I am referring to the local network address.

With the no/pin authentication technique you make a sip uri call to user@3102ipaddress : port#. The distant 3102 will return a dial tone to you where you can enter a pin number and then enter the pstn number you wish to dial.

With the "http digest" technique you setup the distant 3102 as a provider that does not require registration and then you place a call using that provider to the pstn number that you wish to dial. Your local adapter will contact the 3102 using the sip uri address, send authentication, and the 3102 will dial the number on the PSTN port.

If I was going to do this, I would start out with a single router and a 3102 and I would learn how to call that 3102 from the voip adapter attached to the router and access the pstn line. Not until I had that working satisfactorily and understood what was going on would I change things to adapt to a VPN. In the learning process I would learn how to access the graphic user interfaces for the router and the 3102. I would become familiar with the 3102 debug trace tool in case I need to use it.
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Old August 20th, 2007, 07:09 AM
rookiebot rookiebot is offline
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Default Re: Planning the implementation to interconnect 3 stations. Looking for advice.

thanks for the detailed info and i dont mind troubleshooting as long as it works after quite some time . so i too am not confident about using the draytek voip to call the 3102. however u did resolve my qs on how to call the box - which in a way was my main issue.

but i assume you are confident that a 3102 can be configured to call out of another 3102's pstn line? If thats so I'll get 2 x 3102 & regular VPN Routers with DDNS for this.

For starters takin in your advice - I'll purchase the 3102 and call it up locally using X-Lite and see the outcome. As for VPN i'll set it up across a PC and a router for testing.
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Old August 20th, 2007, 06:20 PM
hwittenb hwittenb is offline
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Default Re: Planning the implementation to interconnect 3 stations. Looking for advice.

Quote:
Originally Posted by rookiebot View Post
For starters takin in your advice - I'll purchase the 3102 and call it up locally using X-Lite and see the outcome.
That will get you started. I just looked at XLite and SJPhone.

The free version of XLite apparently only allows 1 "profile" setup. For direct ip dialing of the 3102 to access the pstn line you setup a profile with the UserName, password, authorization user name matching the values you setup on the pstn tab of the 3102. The domain is the local network ip address of the 3102, and the proxy is the ip address : SipPort (without the spaces between the address and the SipPort). You do not check "Register with domain and receive calls".

On the 3102 you enable the pstn tab, set Register NO, Make Call Without Reg YES, Ans Call Without Reg YES, enable the VoIP-To-PSTN Gateway, set VoIP Caller Auth Method HTTP Digest, set VoIP User 1 Auth ID (under VoIP Users and Passwords (HTTP Authentication) your-userid, set VoIP User 1 Password your-password. You should be able to just dial the pstn number on XLite.

The free version of SJPhone allows multiple profiles and more flexibility on the settings. The settings are somewhat similiar to XLite for a profile for the direct ip dialing over the pstn. To setup a profile on SJPhone to call the handset attached to the 3102 you setup a "new" profile of the type "Direct SIP Calls". In your SJPhone Address book you setup a direct sip url which is userid@ipadress : SipPort (without the spaces between the address and the SipPort).

If you wish to receive a call to a phone attached to the 3102 Line 1 tab you enable Line 1, set Register NO, Make Call Without Reg YES, Ans Call without Reg Yes, and put a user id under UserID. The sip uri for the call is userid@ipaddress : SipPort. If you don't change the sip port, the default is 5060 and you don't have to enter it.
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Old August 20th, 2007, 06:20 PM
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