Anyone here work with VoIP?

[dombooth]

I've got a client asking for advice from me, but not sure where to start.

I'll start by saying I know very little about VoIP.

Their current situation is they have one office with two desks, attached to their house.

Each desk has it's own pc and phone line.
Owner works at one, Staff1 works at the other.

Owner is now taking on another member of staff (Staff2), and wants both Staff1 and Staff2 working in the office, and Owner will work in the house.

Owner wants a way of communicating between the house and office, I flouted the idea of VoIP and said I'd get more details.

The idea I have is as follows:

2 standard phone lines (01909 111111 & 01909 111112)
|
VoIP system
|
3 phones (two in office, 1 in house)

Basically, what do I need for this to work? Or am I dreaming and need to go to bed?

Dom

 

[Instant Housecall - Corey]

It looks to me like what you're describing is more of a PBX than VoIP.

The PBX will let you "pool" the two phone lines between the three phones, add extensions, and let you set up a "Press 1 for Mary, 2 for Joe..." autoattendant.

Your customer can use his POTS (Plain Old Telephone System) lines to a PBX via a special card called an "Analogue Interface Card" (if you build it into a PC), but if you buy a PBX off the shelf it'll have that already built-in.

There are a million ways you can set this up. I think if you want to continue using POTS, the easiest thing to do is buy a PBX off the shelf and move on with life.

If you're really interested in learning the nuts and bolts you can buy an analogue interface card (this is a card that plugs the POTS line into the PC that runs the PBX software) for each phone and wire it into the PC/PBX, install Asterisk (industry standard), Trixbox or PBX in a Flash (based on Asterisk but easier)

The "VoIP" way of doing the same thing, and a way that will save your customer a ton of money, is to replace the POTS lines with SIP trunks.

If your customer needs inbound, you would either buy a new phone number or arrange to port the old numbers to a VoIP provider. Then buy however many SIP trunks you'll need (1 or 2 probably), and then attach a VoIP PBX to it. SIP trunks are usually billed on a per-minute basis, and they're scandalously cheap. You'd then either (a) buy SIP phones; or (b) buy an ATA that converts standard phones into SIP and continue using the existing phones

Barely scratched the surface here, and if I were in your shoes, I wouldn't bother learning all this stuff unless you're really interested. It's possible, and there will be lots of people who will show you how you can do it for free, but I think the effort isn't worth the time of learning/building/maintaining/upgrading the thing if you have a business to run.

A couple SIP trunking providers to consider:
- 2600hz.com - a very good UI and very easy to use with every feature imaginable including a reseller platform
- unlimitednet.us - small provider (Jason Canady, who I think frequents Technibble) with personal service

HTH.

 

[Altster]

From my experience with VoIP I can tell you that the better the bandwidth (ie; a dedicated internet connection for EACH phone) is the best way to go. Before we went with Cable Internet (Comcast Business) we had a "crappy" (shared) DSL connection with the phone company (AT&T). I hated to answer the VoIP phone because of the lousy bandwidth / speed. I still don't like to answer (any) phone calls from that phone because it is on the other side of the room.

Oh well, just my 2 cents here.

 

[Moltuae]

From my experience with VoIP I can tell you that the better the bandwidth (ie; a dedicated internet connection for EACH phone) is the best way to go. Before we went with Cable Internet (Comcast Business) we had a "crappy" (shared) DSL connection with the phone company (AT&T). I hated to answer the VoIP phone because of the lousy bandwidth / speed. I still don't like to answer (any) phone calls from that phone because it is on the other side of the room.

Oh well, just my 2 cents here.

I believe it's more about QoS than bandwidth. It's important to have sufficient bandwidth, but the amount of bandwidth required for each concurrent call is actually pretty low; generally less than 100Kbps per call I think. It does require a consistent, uninterrupted connection though, so the traffic needs to be prioritised above all other internet traffic.





I've done a little work with VoIP in the past, but I'm by no means an expert. I am soon to install two fairly large VoIP PBX-based systems though .... so ask me again in a few months!

 

[dombooth]

It looks to me like what you're describing is more of a PBX than VoIP.

The PBX will let you "pool" the two phone lines between the three phones, add extensions, and let you set up a "Press 1 for Mary, 2 for Joe..." autoattendant.

Your customer can use his POTS (Plain Old Telephone System) lines to a PBX via a special card called an "Analogue Interface Card" (if you build it into a PC), but if you buy a PBX off the shelf it'll have that already built-in.

There are a million ways you can set this up. I think if you want to continue using POTS, the easiest thing to do is buy a PBX off the shelf and move on with life.

If you're really interested in learning the nuts and bolts you can buy an analogue interface card (this is a card that plugs the POTS line into the PC that runs the PBX software) for each phone and wire it into the PC/PBX, install Asterisk (industry standard), Trixbox or PBX in a Flash (based on Asterisk but easier)

The "VoIP" way of doing the same thing, and a way that will save your customer a ton of money, is to replace the POTS lines with SIP trunks.

If your customer needs inbound, you would either buy a new phone number or arrange to port the old numbers to a VoIP provider. Then buy however many SIP trunks you'll need (1 or 2 probably), and then attach a VoIP PBX to it. SIP trunks are usually billed on a per-minute basis, and they're scandalously cheap. You'd then either (a) buy SIP phones; or (b) buy an ATA that converts standard phones into SIP and continue using the existing phones

Barely scratched the surface here, and if I were in your shoes, I wouldn't bother learning all this stuff unless you're really interested. It's possible, and there will be lots of people who will show you how you can do it for free, but I think the effort isn't worth the time of learning/building/maintaining/upgrading the thing if you have a business to run.

A couple SIP trunking providers to consider:
- 2600hz.com - a very good UI and very easy to use with every feature imaginable including a reseller platform
- unlimitednet.us - small provider (Jason Canady, who I think frequents Technibble) with personal service

HTH.

Corey, that is just the explanation I was looking for! Thank you!

They don't want to go full VoIP at the moment, so want to keep the two POTS lines.

Is this what I'm needing:

ZYCOO ZX20-A202 Asterisk IP PBX; 2-Port FXO

And some IP phones and headsets:

Snom 300?

Dom

 

[jmitservices]

I've got an office of 10 all using Snom300 and a few Snom821. The 821s are very nice!

Should be able to pickup a snom300 for £45ish and the 821 around £75-100.

 

[layoric]

Voip.ms is another. Just as mentioned, port the number to them, then you can buy a few voip phones, setup ring groups where all sub accounts (trunks) ring at once, set up a voice menu system, rules, even call each phone w/o using the outside line. Some things to tweak like dialing rules, but they can help you with that.

 

[tek9]

Some voip providers don't have the ability to display caller id names on outgoing calls (voip.ms for example).
Also the router you're using could cause a problem for some setups if they're using NAT and SIP ALG. You might have to disable SIP ALG on some routers, while on some routers it can't be disabled.

 

[Instant Housecall - Corey]

Corey, that is just the explanation I was looking for! Thank you!

They don't want to go full VoIP at the moment, so want to keep the two POTS lines.

Do they eventually want to move over to VoIP (specifically replacing their POTS with SIP)? If not, what are their goals in making this change? If it's just to get a PBX, there are better and cheaper ways to do this.

The answer to that questions affects the equipment you want to buy at this stage. There is a little benefit to having SIP handsets with a POTS line via a PBX, but not much.

If they already have multi-line analogue phones, and they plan to stay on POTS, you can buy an ATA to connect the analogue phones to the PBX. Then you don't have to re-wire their office with Ethernet: you can re-use the existing wiring for analogue.

That's cheaper buying SIP phones, easier, and they won't know the difference. But again, which route you choose depends on what the long-term plan is.

Is this what I'm needing:

ZYCOO ZX20-A202 Asterisk IP PBX; 2-Port FXO

Yes, that's exactly what you're looking for. The PoE is a nice touch. But refer to caveat above and determine whether it's worth it.

I don't have any experience with that device. My experience lesser known devices is that they tend to bog down over time and require reboots. Also, they don't specify and I doubt it's vanilla Asterisk (probably Trixbox if I had to guess), but it's not specified. If it is vanilla, setting it up can get complex.


/*
EDIT: None of this applies to you. POTS lines wouldn't be affected by these, only SIP
There are also other wildcards. As another poster mentioned, QoS matters a lot. Latency/jitter makes a big difference too. Bandwidth, not so much if they're on anything reasonably high speed provided you have good QoS/latency.

If the customer's router doesn't have QoS, you can put the PBX in its own VLAN and reserve bandwidth for it that way. If you can't do either, you'll need to buy a new router.

A good router makes a BIG difference. Stuff you can't see when surfing the web can affect a phone call like you wouldn't believe. So make sure your customer has a good router, and stress test your setup by putting a big file onto OneDrive and making a call while it's uploading (the upstream is going to be your Achilles heel, not the downstream).
*/

And some IP phones and headsets:

Snom 300?

Dom

I've never used a Snom. I use Cisco/LinkSys, and couldn't be happier.

If you describe your goals in this experiment a little better, I can probably offer a recommendation. For example, is it just to get additional lines (in which case you could use SIP for outbound and keep the POTS for inbound for a short term experiment)? Is it to get the features of the PBX like call transfer and autoattendant? Do you want to eventually go all-VoIP to save money? Are there features of VoIP that you want to implement that you can't on POTS? Do you want to be able to roam? ... etc.

 

[dombooth]

Do they eventually want to move over to VoIP (specifically replacing their POTS with SIP)? If not, what are their goals in making this change? If it's just to get a PBX, there are better and cheaper ways to do this.

The answer to that questions affects the equipment you want to buy at this stage. There is a little benefit to having SIP handsets with a POTS line via a PBX, but not much.

If they already have multi-line analogue phones, and they plan to stay on POTS, you can buy an ATA to connect the analogue phones to the PBX. Then you don't have to re-wire their office with Ethernet: you can re-use the existing wiring for analogue.

That's cheaper buying SIP phones, easier, and they won't know the difference. But again, which route you choose depends on what the long-term plan is.



Yes, that's exactly what you're looking for. The PoE is a nice touch. But refer to caveat above and determine whether it's worth it.

I don't have any experience with that device. My experience lesser known devices is that they tend to bog down over time and require reboots. Also, they don't specify and I doubt it's vanilla Asterisk (probably Trixbox if I had to guess), but it's not specified. If it is vanilla, setting it up can get complex.

There are also other wildcards. As another poster mentioned, QoS matters a lot. Latency/jitter makes a big difference too. Bandwidth, not so much if they're on anything reasonably high speed provided you have good QoS/latency.

If the customer's router doesn't have QoS, you can put the PBX in its own VLAN and reserve bandwidth for it that way. If you can't do either, you'll need to buy a new router.

A good router makes a BIG difference. Stuff you can't see when surfing the web can affect a phone call like you wouldn't believe. So make sure your customer has a good router, and stress test your setup by putting a big file onto OneDrive and making a call while it's uploading (the upstream is going to be your Achilles heel, not the downstream).



I've never used a Snom. I use Cisco/LinkSys, and couldn't be happier.

If you describe your goals in this experiment a little better, I can probably offer a recommendation. For example, is it just to get additional lines (in which case you could use SIP for outbound and keep the POTS for inbound for a short term experiment)? Is it to get the features of the PBX like call transfer and autoattendant? Do you want to eventually go all-VoIP to save money? Are there features of VoIP that you want to implement that you can't on POTS? Do you want to be able to roam? ... etc.

Basically they want to be able to transfer calls between phones, and call each other from two different offices.

Currently one number goes to each phone, so two totally separate lines.

Office is already ethernet cabled, no problems there.

I don't think they'll want to upgrade to SIP just yet, so all the VoIP bit will be internal for the time being.

Dom

 

[Instant Housecall - Corey]

What you've described will work, but it's far from ideal. Troubleshooting the off-site phone (the one that's not in the office with the PBX), port forwarding/dynamic IP, figuring out which network is causing jitter/echo/whatever is going to be a lot of work. Plus you'd ideally want to set up QoS on the off-site phone's network which means more equipment... and a lot of labour and possibly travel. Yuck.

For the purposes you've described, I'd do the following as a proof-of-concept, with the medium/long-term goal of moving them over to full-on SIP:
- Go with a cloud SIP provider (probably 2600hz in this case)
- Attach throw-away local phone numbers to the SIP lines
- Forward the POTS line to the throwaway on your SIP so your incoming goes through the cloud PBX.
- Buy two ATAs (one for "in-office", one for the person who's out of office), then connect their analogue phones to the cloud PBX. This avoids re-wiring their office
- Strongly consider getting a router with QoS or VLANs for both in- and out-of-office.

If everything goes well, you can port the POTS number to the SIP provider and do away with the POTS, then buy SIP phones and re-wire.

If it doesn't work out, you can unforward the POTS line and revert back very easily, and you didn't have to go to all the trouble to find out that your customer doesn't like it

Easy, cheap, fast, very little to manage.

 

[4ycr]

Dom,

If you want to mange this yourself try contacting Voipfone, I use them and they are good. 2 phones will be fine on DSL, I have a client with 12 phones with 1.5Mb up (they are waiting on a leased line being installed)

If you just want to service the client and take a back seat on Voip then partner with someone. I have a connection that I use for clients that want more complicated setups or want to transfer existing things to Voip as I don't know enough about that side.

He is based in Edinburgh but I can pass on details if you want.

 

[SAG]

I mucked with one of those Zycoo boxes before at a trade show.
If I recall correctly, it had a really dumbed down web based interface (I don't recall it being a FreePBX spinoff, likely some inhouse stuff). It didn't have all the fancier or cool features of a full FPBX box either, but DEFINITELY had what you were looking for and more - runs an older Asterisk though, which is really no big deal for this application.
Grab that and a couple inexpensive Grandstream phones and your client could be happy on the cheap.

 

[layoric]

Some voip providers don't have the ability to display caller id names on outgoing calls (voip.ms for example).
Also the router you're using could cause a problem for some setups if they're using NAT and SIP ALG. You might have to disable SIP ALG on some routers, while on some routers it can't be disabled.

Actually it does. I enter the details in my caller id info on their site, then it shows up as per whatever name I typed.