I have a couple of (hopefully) simple questions that I can’t seem to find an easy answer to on tinternet in any of the forums.
My company is about to replace it’s existing ancient pbx system and I have recommended Asterisk as I worked with it around 18 months ago and found it to be quite exciting.
We have an ISDN30 connection with I think 12 channels enabled. The company has around 120 users and I don’t suspect there to be any more than 10 lines active at any one time. I’ve been given the go ahead to purchase some test equipment but I need to spec up an estimation for total cost of the final system.
My questions are:
Does each isdn line allow for two active voice connections… i.e. will I need just one 8BRI card or an 8BRI and a 4BRI to achieve my total of 10 simultaneous voice calls?
I am planning on using a sepearate network just for the phones and phone system and all traffic on this network should be VoIP. I plan to have a second NIC on the asterisk system for management, which will be connected to our LAN. Am I right in guessing that I only need QoS switches if the phones are to be on the same physical network or if I plan to deploy soft phones?
Possibly the easiest question… Is al the VoIP traffic really directed through the Asterisk system… i.e. 10 ISDN channels and just one ethernet port to the switches?
I’m immensely looking forward to setting up and administering this system!
Any help with regard to my questions would be most appreciated!
In answer to question number one… if anyone else is wondering this, the answer is here.
Apologies, I spent a whole day trying to find these answers then after my post, stumbled across that. I’m still searching for answers 2 and 3 though, although I suspect I know the answer to number 3.
Again, any help fro you the forum that may ease my pain would be much appreciated!
I could be wrong but as I understand it, ISDN30 is a PRI (t1/e1) not BRI. You will need a single port PRI card, not a multi port BRI card.
If you do have BRIs then each BRI has two voice channels, each can take 1 voice call or 64kbit/sec of data. So yes you can have two voice calls per BRI.
PRI is the same way except it has many more channels, so if you have 12 channels enabled you can have 12 voice calls active @ a time.
QoS- this depends on your network and what you do with it.
YOU WILL ALWAYS NEED QOS IF YOU RUN VOIP OVER A WAN LINK, so if your users connect from home w/ softphone you will need QOS on whatever your Internet connection is.
For internal use- the easiest way to do this may be to vlan flag packets from the IP phones, and configure your switches to provide higher qos for that vlan. (most phones have passthru ports so you plug computer into phone, phone into wall). If you have 2x ethernet to each desk then you can vlan those ports and prioritize them. Or if you put in a totally separate Ethernet that works too although is probably too much work.
What I would recommend deploying is PoE (802.3af power over ethernet), so your workers can just plug the phone in and not worry about a power brick. Also if the power fails the phones will keep working as long as * and the switch are on a UPS.
I also recommend against softphones whenever possible. They are a nice idea in theory but in reality they only work well for call centers and mobile users (connect from laptop). Having a physical phone on your desk that you can pick up and dial is a creature comfort we take for granted but which can be annoying if taken away. Your users will hate you for that .
- yes it is. Each voice call takes up around 80kbit/sec (64k g.711 ulaw/alaw + overhead), or less if you use a lower-bitrate codec like GSM or G.729. Note that if you plan to fax- you will need ulaw/alaw for the fax machine at least.
Hope that answers your question!
usually my thoughts too … but i had a PoE switch fail on me a few weeks ago. replaced the next day but it still meant a day of very few phones in action. it’s very tempting to buy a load of single PoE injectors and some spares for the next install.
You are an absolute diamond. Your comments have I suspect pointed me in the right direction, or rather, much less in the completely wrong direction as I was a few moments ago. My previous experience was with BRI cards and ISDN2e connections. Knowing that ISDN30 requires PRI cards also makes a bit more sense to me after having perused my comms room and not found anything that looked vaguely familiar.
Wish me luck! I will report back on my progress for good karma and endeavor to answer any questions posed by others even less experienced than myself, if that is possible.
mac- good luck! we will be here if you need any help…
bacon- single port poe’s? that’ll be expensive, as i recally they are at least $30/each, which means for 10 you pay $300 bare minimum. For $300 netgear has a decent 24port/16poe switch… also take into account power consumption and physical size- each of those injectors has its own transformer brick or wall wart (large/ugly, doesnt fit well in a rack) and also each wall wart is its own transformer, which all together probably results in a good bit more wasted power due to inefficiency than one switch alone.
IMHO a better idea is to keep either a PoE switch on hand as a spare, or keep a box full of injectors that can be rapidly deployed when needed…
Thanks guys, your comments are invaluable.
The guys in charge want to know if they can keep some or most of the company’s analogue phones with asterisk, and whether it would be cheaper to replace them all with low end ip phones or get the necessary equipment to allow analogue phones to work with asterisk, i.e. a channel bank.
What are your thoughts on this? I have no experience with this type of equipment. Do you need a channel bank between the asterisk server and the analogue phones, and an FXO / FXS port for each phone…? The specs on this equipment are not too easy for a beginner to make head or tail of…!
Je suis getting a little confused now!
Thanks again in advance!
I’m still looking into this… the Rhino channel bank sounds like it might be the best thing for the job.
Would I be correct in saying that all I need is one Rhino FXS channel bank for every 24 analogue phones?
Can it be that easy!?
1 fully populated (24 ports) channel bank maybe. don’t buy just a chassis whatever you do
you’ll need to check compatibility before buying, particularly if they’re from an existing PBX.
Thanks for the tip Bacon…
So I’d need an FXS chassis with 6 FXS cards (as long as they’re compatible with my Ascom Berkshire DS phones)? It’s weird because looking at the photos of the unit, I can’t see where the cards will fit in, i.e. there are no obvious slots for the cards and their ports. Or does it not work like that?
I think I’m going to contact these guys to get more info. Will post back here when I find out more.
It says on their website that if using with Asterisk to buy one of their T1 cards. Can I use a sangoma card instead, do you reckon?
Sangoma cards are good too. they talk the same signalling as the Channel Bank. installation is a bit longer, but easy enough (not sure it could be any easier actually).
big question is going to be the phone compat.
A channel bank is pretty much a T1 (PRI) to POTS adapter, it takes a T1 input and gives you POTS channels for it. You can use any T1 card you want, digium, sangoma, rhino, whatever.
Fully populated channel banks are around the cheapest $/port you can get… I am worried about your phones though. I am guessing that the phones are system phones from your old PBX, which means they will almost certainly not work with a standard analog channel. And if they do, the feature keys (transfer hold conference etc) probably will not work. A good way to test this is to plug a system phone into a normal analog telephone line and see what happens. Whatever happens will be what you get with a channel bank.
What you can possibly do (depending on if your PBX can deal w/ this) is setup a T1 link betweent the old PBX and *, then set the PBX to be ‘dumb’, ie just pass calls between the T1 and the phones nothing more, so the PBX becomes nothing but a T1-systemphone adapter. If or how this will work depends on your old PBX. You can then replace with IP phones one workgroup at a time or as budgets permit.
You can now get budget IP phones (grandstream budgetone 1xx) for as little as $40, which is very competitive with the $/port even of a channel bank. Plus which you get real feature keys (xfer hold conf etc). Keep in mind tho the Budgetones are really cheap phones- they don’t ‘feel’ solid and they only have a numeric callerid display. They also don’t support PoE so you need a wall wart to power them.
The slightly more expensive Grandstream GXP2000 does support PoE and looks a lot more business like…
For your own sanity, I suggest try to stick to only 1-3 models of IP phone. You want to be able to mass provision them via TFTP and each manufacturer has their own method of doing this. As always I recommend SNOM and AAstra because they make good phones and they are easy to administer.
I tallied up yesterday how much it would cost to upgrade all phones to low end ip phones compared to buying a sufficient number of channel banks to cover the existing analogue phones, and you’re quite right; there’s not a lot in it. And it seems to make much more sense to spend a little more on upgrading to new phones than paying a lot of money to support a load of old phones.
I think the plan will be to give most users a basic phone, and then give key people middle range phones and maybe a snom 360 with expansion module for reception.
My previous experience was mainly with cisco 7960’s, and I found them to be generally very good. I liked the fact that you could put your own company logo or a tux penguin on the screen (but I guess that’s not essential) and having six lines to play with etc… How would you say the snoms and aastras compare? They certainly seem a little less pricey, which is good!
Just a thought… Are all ip phones, i.e. even the cheap ones, tftp / DHCP bootable, picking up their config etc. from the server according to mac address… or do some require some manual configuration? I vaguely remember having to manually configure some cisco phones with ip configuration before they would work, but I can’t remember why…
I will rack my brains and seach the forums…
you could be me !! but don’t go too cheap on the basic phone. i usually do something like Snom 360 with extension for reception, Aastra 480i for middle range, then with the Aastra 9133i or 9112i for everyone else. makes config mgmt easier too. having said that, i actually think the 9133i is more functional than the 480i if you’re not using screen apps … the additional BLF buttons are welcome to most users.
I will try and convince the powers that be to get half decent phones. But at the moment, the biggest expense is already on the phones themselves. Doubling the cost of them may not go down too well.
Still, I reckon it’s still considerably cheaper than a similar sized BT system.
I will make my recommendations and hope for the best!
there are certainly some instances where a conventional phone system is going to beat an Asterisk system easily on price. there are plenty of small PBXs out there that have a great entry price, but are restricted on features and expandibility.
i did a quote for a new install/replacement of Panasonic PBX and Motorola radios recently, and my quote was cheaper than the competitors by about GBP1K (still GBP9K though). the majority (over GBP6K) was the cordless handsets they wanted.
so in summary, yes, handsets usually feature highly in the “bloomin’ 'eck, how much” conversation. most finance directors have no idea how much a phone system costs !
Same here Bacon,
Around ?6.5k of my planned installation is phones… and that’s using the cheapest I could find. I need to roughly double that figure if we’re to use the 9133i or x 1.5 for the 9122i.
This is my problem!
Is the main issue with the cheaper phones (apart from the “feel” or build quality) the fact that users will have to learn key shortcuts for transferring calls, putting on hold etc… in stead of having programmable buttons? I mean, are there any sound quality issues or anything more major to worry about with a cheaper phone?
depends on the phone, most SIP phones should have buttons for XFER, CONF, HOLD, etc. They will of course have to learn to dial people by extension.
As for TFTP/DHCP, most phones support this but not all. SNOM and AAstra do out of the box. Grandstream supports it but it is turned off by default (WTF). However with GS, the TFTP server is defaulted to a Grandstream address so if you set your DNS to map that address to your local TFTP, the result should be the same.
No idea about Yuxin, but there were some problems with their company spamming the Wiki a while back (posting repeated advertisements to voip-info.org despite being in violation of the Wiki posting rules and being told not to)… for that reason I personally won’t buy their stuff anytime soon, but YMMV.
Fact is the cost is usually in the phones. For small installs, analog systems can often beat *, but where the difference comes is in 2 years when you want more than 48 handsets or xx minutes of voicemail or want yy feature, and you realize that to get that you have to rip EVERYthing out… Asterisk and good phones may be more expensive up front but it will save you money over time with all the stuff you DON’T have to buy. Even a lot of the higher-end PBXs have this- every problem you run into is a feature you must pay to unlock.
Whichever way you go- especially if you have a larger deployment, set up the server and buy 1 or 2 of each model phone you are considering. See how easy they are to use, to admin, if the quality is good, etc. Make your decision from there.
Hi all, me again!
I have received quotes for the necessary number of phones etc… and am going ahead with a bit of testing. I’m going to order a couple of 9122i’s for testing and have already installed libpri, zaptel and asterisk.
I’ve hit my first obstacle which is that when I run
I get a number of error messages concerned with being unable to load certain config files and / or modules. From the asterisk install guides I’ve found it sounds as though the console should just work after running all the “make installs”. However, I’m not sure if it is assumed that some hardware will already be installed, which of course in my situation, there is none.
So my question is (before I try and troubleshoot any further, I’ve already been googling this for most of today!): how much testing can actually be done without any pri hardware? It is my intention to test this with soft phones asap and hard phones when they arrive. So can I setup and asterisk system and get two soft phones on the network to communicate with eachother? If so then do I need to install a “dummy” driver of some sort to get past these error messages when trying to get into the Asterisk console?
The error messages being generated are concerning sip.conf (which I have not created / edited yet), chan_oss.os and oss.conf, dundi.conf, extensions.ael…
Most of the errors concern configuration files that do not appear to exist yet. Are there no defaults for any of these or do they all need to be configured manually?
I fear I may have truly exposed and laid bare to all my level of noobiness.
I will of course keep googling and following the how-to’s etc… but if anyone can pick me up off the floor and set me stumbling in the right direction, it would as always be much appreciated.