Voip between two offices

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leonb

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We recently set up a wireless link between our two offices in town to get everything on a common network. The link is giving a stable 65mbps. There is an analogue pbx in both offices. We want to be able to call between the offices for free (not interested in external voip calls at this time). There shouldn't be more than 2 to 3 simultaneous internal calls between the 2 office at any given time.

I need a cheap and effective way to get this done, preferably routing the calls via the analogue pbx switchboard and from there to the extensions.

The pbx on both sides is a Karel ms48c. Karel does have voip cards that can be added to the system to do this seemlesly, but best qoute (including installation) is about R13000. Not sure if it is worth it at this cost - can make many local calls at this price.

Any other options I can consider?
 
You'll have to go the VoIP route. To transmit voice (or anything for that matter) over a data connection means converting the data/analogue signals into IP packets at one end, and vice versa at the other end. If your PBX doesn't support this (which sounds like it does), then you will need to put in hardware on either end to do the conversions.
 
Btw, that R13000 quote seems reasonable, and is probably your best (and possibly only) solution. The improvement in productivity and savings on call costs will pay for this amount rather quickly.
 
If you have a spare pc lying arround you could set up an asterisk pc (I would recommend elastix but there are many versions), you could then use free software phones like this or this to make calls between the offices. You could also buy actual sip phones and plug them into the network to make calls between the two sites.

If you wanted to take it a step further you could buy a tdm card for the pc and make outgoing calls via the network. You would also be able to receive outside calls and transfer them to anyone on either site.

You could do the same thing with one of these with minimal setup I suppose. again you could extend the functionality by adding FXO cards for incoming analog lines, FXS cards for analog extentions and GSM cards for Cell phone lines. It doesn't have all the bells and whistles that a full asterisk install would have, but installation would be a whole lot simpler.
 
Madman88 said:
If you have a spare pc lying arround you could set up an asterisk pc (I would recommend elastix but there are many versions), you could then use free software phones like this or this to make calls between the offices. You could also buy actual sip phones and plug them into the network to make calls between the two sites.

If you wanted to take it a step further you could buy a tdm card for the pc and make outgoing calls via the network. You would also be able to receive outside calls and transfer them to anyone on either site.

You could do the same thing with one of these with minimal setup I suppose. again you could extend the functionality by adding FXO cards for incoming analog lines, FXS cards for analog extentions and GSM cards for Cell phone lines. It doesn't have all the bells and whistles that a full asterisk install would have, but installation would be a whole lot simpler.
Click to expand...

As mentioned, we need something that can be used with the currenr pbx's, as there is about 15 telephone extensions on both sides. The pbx will need to route the voip calls.
 
leonb said:
As mentioned, we need something that can be used with the currenr pbx's, as there is about 15 telephone extensions on both sides. The pbx will need to route the voip calls.
Click to expand...



I thought you just wanted to call between offices?
If you wanted to be able to transfer calls and so on then you would need to buy the add ons specific to the pbx's you currently have or replace them with an asterisk box. The features of the asterisk system far outstrip any others when it comes to bang for your buck.
Full voip integration, ivr , call recording, pin control, full reporting, full call recording, integrated fax server, voicemail to email.. the list goes on and on and all for a fraction of the cost of an equivalent pabx.

Although you could possibly route the calls through the asterisk box using pstn carts. That depends on the functionality of your current pbx tho.
 
Like I said, go with the R13k quotation. No use reinventing the wheel when you already have a decent one that works. Perhaps try to get several quotations from different suppliers. You'll probably find a better deal.
 
Siemens use to have a device that you simply plugged into one extension and that converted it to digital, where it could be routed as normal IP traffic. We installed one at each branch and the extension simply became an extension on the other PABX.

So if you dialed that extensions number it showed at reception at the other branch who could simply route the call to the person you needed.

I dont know if they still have this as I am talking about 2001/2002 but I am sure if you look you will be able to find something like this.
 
If I would to have a dedicated analogue phone at each office for interbranch calls, what would be the minimum and most simple equipment needed to make calls. If I know this, I'll be able to do what is needed.
 
leonb said:
If I would to have a dedicated analogue phone at each office for interbranch calls, what would be the minimum and most simple equipment needed to make calls. If I know this, I'll be able to do what is needed.
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Why analogue? To have an analog phone on either side you would need a full pbx with voip on either side. Thats your 13k option above.


If you had a sip phone on either side you would need a setup as I described in post #4.

One more option I thought about now is a Vox phone on either side, some of my family use them for inter branch calling since calling from one vox phone to another vox phone is free. But again its not analogue.
 
OK my friend this is how you do this if you want the calls to go PBX to PBX

1/ VoIP enable your PBX’ s @ 13k as quoted

IF you have spare extensions or trunk ports on the PBX’s you may look at this as well, if you don’t have you can always add.

2/ Put an FXS or FXO gateway at each end and get a VoIP account with a provider that allows free on net calls and rout via DSL or whatever.

3/ / Put an FXS or FXO gateway at each end and build an Asterisk box one end to drive them, Digium or Sangoma cards for the Asterisk box will be more expensive than a gateway, so best go the gateway route.

4/ Put an FXS or FXO gateway at each end and load a free soft switch like 3CX, Snom 1, BizPBX ect on a Windows PC to drive the gateways, these free edition soft switches will allow you between 2 & 4 concurrent calls depending on which one you choose, before you have to shell out for the paid versions. The lack of g729 codec is not an issue because you have so much available bandwidth on your link so g711 is ok.

5/ Put a gateway one end and cheap Yeastar PBX the other, the Yeastar can talk to the PBX at its own end and the gateway at the other.

These are your options if you want to make your calls from your existing extensions, other options are of course using dedicated IP phones or soft phones at each end.

So some food for thought, I can give you more detail on each solution if required.
 
paulcolmer said:
OK my friend this is how you do this if you want the calls to go PBX to PBX

1/ VoIP enable your PBX’ s @ 13k as quoted

IF you have spare extensions or trunk ports on the PBX’s you may look at this as well, if you don’t have you can always add.

2/ Put an FXS or FXO gateway at each end and get a VoIP account with a provider that allows free on net calls and rout via DSL or whatever.

3/ / Put an FXS or FXO gateway at each end and build an Asterisk box one end to drive them, Digium or Sangoma cards for the Asterisk box will be more expensive than a gateway, so best go the gateway route.

4/ Put an FXS or FXO gateway at each end and load a free soft switch like 3CX, Snom 1, BizPBX ect on a Windows PC to drive the gateways, these free edition soft switches will allow you between 2 & 4 concurrent calls depending on which one you choose, before you have to shell out for the paid versions. The lack of g729 codec is not an issue because you have so much available bandwidth on your link so g711 is ok.

5/ Put a gateway one end and cheap Yeastar PBX the other, the Yeastar can talk to the PBX at its own end and the gateway at the other.

These are your options if you want to make your calls from your existing extensions, other options are of course using dedicated IP phones or soft phones at each end.

So some food for thought, I can give you more detail on each solution if required.
Click to expand...

Its a common network so there is no reason to have a gateway/pbx on each end.
 
I'm not entirely sure why you'd want to route via Analogue signals, but if you have an existing PBX on both sides, you could replace the entire system relatively cheaply (depending on the amount of analogue lines you have running into each PBX).

The way we've got Lindt setup is such:
- Elastix PBX box running on top of VMWare (You can get two of these for very cheap - and Yeastar TDM-400's in each - also not too expensive, VMWare not essential)
- VPN which connects Jozi to CPT (VPN not necessary if you have a common network already)
- IAX2 Trunk to go between CPT's PBX box and other trunks it needs to link to with a dialing rule to route calls over the internet to that box as opposed to over a paid network.

What you can do for the most cost-effective solution is setup two Elastix-based PBX boxes with as many Yeastar's as you need (Fully Elastix compatible) and setup an IAX2 Trunk between them over your wireless network. You won't need to worry about bandwidth over the trunk because it's a dedicated wireless connection.

If you only have lines coming into one office, all you need is one PBX box in one office and set it to route the calls to where they need to go.

AFAIK transferring calls and the pickup works over IAX2 (As long as the phones are in the correct call/pickup groups), but haven't tried it myself.

glhf
 
Leonb, there are too many suggestions here which will confuse you.

PaulColmer covered one of the main points:

A FXS gateway costs less than R600. www.miro.co.za
Buy two of them.
Connect one at each PBXs, tell your PBX technician to connect it as a trunk, and allocate a prefix to dial the other PBX.
Connect each gateway to your LAN network, ensuring it has internet access.
Purchase 2 SIP Accounts at www.cheapcall.co.za , at R210/sip account for the whole year covers the DiD cost only.

Simple, cost effective solution. You can do it DIY if you are familiar with routers.

Good luck!
 
mo_to said:
Leonb, there are too many suggestions here which will confuse you.

PaulColmer covered one of the main points:

A FXS gateway costs less than R600. www.miro.co.za
Buy two of them.
Connect one at each PBXs, tell your PBX technician to connect it as a trunk, and allocate a prefix to dial the other PBX.
Connect each gateway to your LAN network, ensuring it has internet access.
Purchase 2 SIP Accounts at www.cheapcall.co.za , at R210/sip account for the whole year covers the DiD cost only.

Simple, cost effective solution. You can do it DIY if you are familiar with routers.

Good luck!
Click to expand...

Hi mo_to,

Thanks, this is more or less the idea I had.
Since I dont know much about voip, can you please explain the following in you post:
1) why do I need to be connected to the internet (I have both offices on the same lan)
2) what is the sip account and why is it required for calls on the same lan. Can the 2 gateways not communicate directly.
 
1) why do I need to be connected to the internet (I have both offices on the same lan)
>>You need a medium to translate the VoIP from DATA. This can be done (as per the many good suggestions you've received above) either via an Asterisk Box, or some sort of other device which I am not aware of.
The easiest way is to run through a service provider (hence the internet connection) who will take this headache away from you, as maintaining an Asterisk box has it own challenges, so definitely worth the R210 per year for someone else to do it for you. Somehow, I doubt you have time to maintain Asterisk boxes in your already busy schedule, hence my simple suggestion.

2) what is the sip account and why is it required for calls on the same lan. Can the 2 gateways not communicate directly.
>>Half my answer here is in point one, but keep in mind that you want the convenience of having any of your users to simply dial the other PBX to speak to someone in the next building. For this, you need a gateway and a SIP Account.
If there is a device (not gateways) which can communicate directly as per your question 2, it will most likely be an IP or softphone, in which case, you will not be able to integrate in the PBX.

Hope the above makes sense.
 
The gateways cannot talk to each other over your LAN, the gateways are looking to talk to a SIP server. You only need 1 severer at either end, it does not matter which end. End point 1 PBX will communicate to the SIP server, whether it be a Asterisk, Yeastar, whatever. Endpoint 2 will communicate with a gateway that talks to the server at the other end, that will give you the communication you need. You can use additional trunks on the PBX’s if you got them or if your existing trunks are not to congested you could share these. Mo-to suggestion is very simple to implement, the set up will be very similar to the above but the server will be at O-Tel not your own on site, this option will achieve what you want with minimal capital outlay for 2 x gateways, well worth looking at ! Do you really want to set up and maintain your own server ? This was option 2 on my original post.
 
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FFS. There is no need to have two servers.

If the two locations are on the same network then the geographical location is of no consequence. It doesn't matter where on the lan the sip extensions are, as long as they're on the same lan.

In fact they dont need to be on the same lan, but thats a different kettle of fish.

Surely if you guys do this for a living you can wrap your head around that?
 
I did not say you require 2 servers, i said you require 1 only and it does not matter which site you put it on, or you use a server at a VoIP provider as sugessted by Mo-to, either way its still only 1 server !
 
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