Potential Asterisk convertee - Questions

Hello, I administer a ~50-user 3COM NBX system at my place of employment. Since NBX phones communicate with the “PBX” via ethernet (at layer 2) it’s almost VOIP, at least on the in-house side of things. For external communications we have 2 POTS lines and a T-1 connected straight to the NBX via a digital line card. I’m in the process of meeting with telco/internet vendors (starting Monday) in hopes of negotiating new contracts for both internet and voice connectivity. We’ve got a lot of phones showing wear and tear combined with the feeling that we’ve reached the crest of what this system can do for us. So, I’m very interested in beginning research to potentially replace our current system.

Our company already develops and markets software solutions to clients based around Linux servers for which my department in particular provides installation and support services. We’re a very much do-it-yourself and open-source heavy company, whatever this project might require I’m confident we can make it happen…

Having said that, and having read about (fairly heavily) Asterisk and similar products and related technologies, I’ve got a few questions and concerns some of you might be willing to help set me straight on.

In it’s simplest form I could probably replace the NBX and all desktop phones with a Linux server running Asterisk and SIP phones. If this happened and the voice T-1 stayed in place there wouldn’t be much to differentiate it from our curent NBX system. Potentially though I could extend that beyond our networks perimeter such that intsead of a “voice” T-1 delivering calls in and out, data packets delivered over an IP network could be implemented. An advantage is unused “voice” channels being available for data applications. One network (IP) in and out of the building dynamically sharing bandwidth.

Now to do this though my thoughts are that whomever supplies “dialtone” in this fashion needs to be close, few hops, low latency, higher chance of stability. So the actual companies I’m meeting with to negotiate internet T-1 or higher connectivity are the ones I’d rather have delivering “dialtone”. Does that make sense and is it the norm? I’m leary of my voice calls traveling halfway across the country as internet data before hitting a company to put them on the PSTN. As a general rule the internet is “reliable” but it’s still not quite like the PSTN I think… So, Sprint/AT&T/Qwest and some smaller regional companies are all planning on making their pitches to us in the coming weeks. What questions do I ask? What track records do any of them have at delivering VOIP services? Will they interact with Asterisk at all or do any/all their VOIP products require proprietary solutions?

If it helps as far as any suggestions you might make here’s a breakdown of our expenses and some call metrics:

2 voice T-1s, one to the NBX the other to a channel bank which I’d love to do away with. Mainly modems and faxes on it, what’s the likelihood of me being happy with modem/data calls over VOIP via an ATA? I digress though, local loop on these are $450 each, ouch.
Inbound and outbound intra/interstate any time of day though is <.04/min.
I’ve been quoted roughly $650 for an “internet” T-1 from most of the tier1 companies mentioned above so far, hence my desire to combine things and not pay $450 for a voice-only circuit!
Last month we had about 40,000 minutes of LD, split about 50/50 inbound/outbound.

Things I’d love to be able to do:
Have bluetooth on the desktop phone detect my cell phone when I arrive to work and set me up accordingly…
Have cordless “desktop” phones I can carry around the building and use via our WIFI network with calls either going to both or automatically directed to the cordless simply by the act of removing it from the cradle.
More easily support telecommuters working from home or other sites.

Things the NBX is doing for us now we really can’t do without:
Receptionist console with lights to indicate who is available, who’s on the phone, etc…
Hunt and Calling Groups with user login/out ability. Round-robin, linear, etc…
Auto-attendant menu trees, etc…

Smaller things we take for granted that I’m at least slightly concerned about. Not the ability to do these things but the ease and integration of how they work and interact. With 3COM and the NBX at least all devices are made by one company and are therefore pretty much guaranteed to interact properly.

Configurable hold and call-park timeouts. Employee directory via LCD along with missed/answered calls history. Seeing an incoming calls CLID on the LCD and a second line presence light blinking while already on a call. Putting the first on hold, grabbing the second, pressing conference/first on-hold caller/conference and being on a 3-way with both. 4 rings at my desk, no answer, automatic forward to my cell phone. Paging zones. Web-based administration of the whole system with auto-phone discovery, etc…

That list could go on forever and I apologize for not knowing the answers via my reading already. A lot of them I’ve read tidbits about but just not enough definitive information yet.

Thanks in advance for any and all advice you can provide!

I’m not going to touch the other parts of your question, but your discussion of th e receptionist seeing who is available and what lines are in use makes me a bit leary. This is phone “presence” and SIP doesn’t do it very well. Do your research on this if you absolutely can’t live without it. There are some web-based apps that assist with this.

I’d do a search for “presence” to get a little more information on this.

The following is a mind boggling good reception / monitoring software for asterisk. It has so many features and also based on JBOX so you can interface with it easily.


thats my 2cents regarding a switchboard.

Thanks to both of you… The icecom stuff does look nice. I had also read about Flash Operator Panel, Quadra Receptionist and others. I think we’d prefer a piece of hardware to do this but realizing doing it in software is more flexible, that will probably work just fine. A second monitor and video card will also probably be cheaper than a dedicated console.

I’m curious about the comment of SIP not doing this well. Is this not a function entirely of Asterisk? Do calls get “handed off” in some configurations where the phone is “talking” directly with the end-user and the Asterisk server is somewhat out of the picture? Is that the reason why presence is more difficult with this type of solution? I ask because that’s kind of how the NBX works, phones and line cards have DSPs onboard which handle all the actual voice movement. The main system is there only to setup and tear down calls, it’s only a 16mhz 386 I think.

exactly, except that * with a voice T1 may offer more flexibility. You can definately extend beyond your network. You can also (with the right scripts) allow users to log into their phone lines from cell phones, so calls dialed by extension can ring a cell (this may or may not appeal to you). Using an efficient codec like GSM, G.729 or iLBC, you can get almost a hundred channels on a full T1. However you will want quality of service controls on your router to prioritize voice data.

First, see if you can get something faster than T1, depending on where your bulding is you may be able to get something better. If you’re on top of or near fiber, you’re in luck. You’re right that a TDM circuit like a T1 will probably overall be more reliable than an internet telephony service provider (ITSP). It also depends on what your calling patterns are like, your T1 from the local telco may offer cheap/free local calls, whereas with an ITSP you’ll be paying a cent or two per minute usually. OTOH, if you do alot of long distance, ITSP will almost always be cheaper. You can easily setup both- most ITSPs will let you register an account with like $20, and it stays attached to your account until you use it up with phone calls.
As for Sprint/ATT/Qwest, to the best of my knowledge they dont really even offer business voip service unless its locked to an ATA (which is NOT what you want). I could be wrong, maybe they do, but they’re probably gonna push TDM solutions.

on to your usage patterns…
the likelihood of you being happy with fax/modem over an ATA is very low. Keep the channel bank around for now. The one exception is faxing, if you have a TDM circuit going straight into Asterisk, it can with some tweaking recieve the fax pretty well and send it to email or a printer, so you can use * as a fax-email gateway. However doing this over a voip line isnt a great idea. If your modems are for dial in data, consider getting a contract with a national dialup provider and some good VPN software. Might cost less too.

Asterisk has a chan_bluetooth, but i don’t know much about it.

Cordless phones can be done, depending on how your building wifi works. Wisip type phones aren’t great with WPA encryption (most don’t support it), however they are readily available from a handful of companies. No need to redirect- you can configure asterisk to ring both the desk phone and the wisip phone simultaneously. Whoever picks up first gets the call.
Telecommuting can be greatly helped with Asterisk. Your users can have SIP phones or softphones (software phone that uses sound card+headset) to use, or you can configure * to allow the user to log in using their cell phone, and while they are logged in any call to them will also ring the cell. They can then log out and the cell will be left alone.

Hunt and calling groups with login and calling patterns is easy, use asterisk queues.

Auto attendant is extremely easy to setup using extensions.conf

Receptionist lights (this is called BLF or busy lamp field) is a bit harder but is possible. You can do this with a computer based login (Flash Operator Panel) maybe, or asterisk now supports BLF if your sip phone does. Polycom 601s with the sidecars might be useful.
The one thing asterisk DOES NOT DO: shared call appearances. That is, if your receptionist pushes one of the lit buttons, she is conferenced into the active call. Asterisk doesn’t support this currently and as I understand it support isn’t a high priority.

park timeouts are easy. Hold timeouts are hard, it’s more of a phone thing (i could be wrong on this).
Employee directory via LCD is possible, but this depends on the phones you use. Some phones can access an LDAP directory and display that. Other phones, such as AAstra 480i can use XML applications, which could be written to read LDAP.
3way conference is easy. Caller ID w/ name is easy. Blinking light can be done, see BLF. Also depends on what phones you use. Auto forward can be easily done- you can even have it simulring the phone so they both ring at once. Paging is easy, but you need phones that support it, just make extensions that page different groups of phones.

Web based admin is possible but you get more flexibility if you go straight to the conf files. There are however a few good web based config packages, that might open some flexibility up to your users. Alternatively, with Asterisk Realtime, all the configs are stored in SQL which can be manipulated from any web app.

Lastly- i suggest that before you dive in headfirst, setup * on a small scale, (ie your house or just your extension). Buy one or two of the phones you plan to use and learn their ins and outs. Look into Cisco, Polycom, SNOM, AAstra, Grandstream (gxp2000 only, the bt1xx series looks and feels like a toy), Linksys/Sipura (SPA-841/941). Digium interface cards are good. Try www.voipsupply.com if you need to buy stuff (i dont work for them but i’ve had good lukc with them).

Hope that all helped.

As others have suggested setup a test Asterisk box with SIP phones. That’s where I’m at now, and I don’t care for it much compared to a real phone system.

Also the Cisco 7960s I’m using suck they have no back light for the display, and the SIP software load I’m running is very basic. I plan to switch to the Cisco call manager native SCP and try the 3rd party SCCP driver for Asterisk to see if that works any better.

Asterisk lacks shared call appearances - this kills me.

Reception needs to run a third party application to see who is on the lines? A real phone system does this via a hard phone with an extension brick of some kind - kills me!

I’d love to hear how people have done remote users with Asterisk, with the issues of NAT being a pain with SIP phones.

If you switch from a voice T1 to a data T1 your quality and reliability can only go one direction with that trade DOWN - are your users screaming for lower quality and reliability?

For ciscos’s i’ve heard the new SCCP driver is pretty good, give it a shot.

SCA- yeah, it sucks.

As far as the receptionist- its possible now with a sidecar just like you describe. The only problem, is when she pushes the button for somebody that is on the phone, she won’t be dumped into the call like you would expect. Also I’m not sure if you can push a blinking light to pick up a ringing phone or not.

Thanks for all the suggestions everyone… For a “first post” from you IronHelix you really come out swinging! Thanks…

I’ve met with a few internet/telco providers and one in the area does have fiber and they’re checking to see how much it’ll cost to build out to us. Fingers crossed! If that works out they can provide “T-1” delivered over the fiber connection back to their 5ESS so hopefully that will decrease the “local loop” charge tremendously and avoid shuffling voice calls across the “internet” and the associated problems with that. They provide a VOIP product but it’s like you suspected, a managed something where we have Cisco or soft-phones and they control all the “telco” equipment on their end. Not what I’d want to do…

As far as faxes and modems, let me ask it this way then. Are there ATAs and/or CODECs that do work better with 33kbps modem connections if the “IP” portion of the VOIP call is totally on the local LAN with good switches and great bandwidth and latency going straight to a T-1?


hehe thanks, i try to be useful :smile:

I believe that’s called a virtual T1. T1 routed against packet based backbone. You should get the details on their voip product anyway- if they give you SIP logins, its possible to just hook * up to it, and buy as many lines as you want. Just configure the ‘virtual pbx’ to be as dumb as you can, ie when a call comes in call a phone (hooked up to a * sip channel), dont voicemail ever. Really depends on how possible it is, prices, and if they’re willing to work with you at all.

ah yes faxes…
the problem is that a fax or modem expects to get a relatively reliable voice path, that stays the same for the duration of the call. With VoIP, you have to use the ulaw codec, which most faxes can at least deal with (almost no sound data is tossed with ulaw), but the packet-based nature of the system makes quality much more likely to change over time. Besides, with such things you are encoding the sound of the modem tones and sending them as audio data, which is certainly not the best way to do it. Think of it like reading a book into a tape recorder and then mailing him the tape instead of just handing him the book.

There is a protocol called T.38 that will give you great faxes. A T.38 compatible endpoint can decode the modem tones, and then send the resulting data frames directly across the system where they are remodulated back into modem tones at the other end. T.38 is designed for faxing only (as I understand it at least), not modems. However, very few ATAs support T.38, and asterisk’s latest development on T.38 is that it will correctly reject any T.38 attempt and say it’s not supported instead of becoming frightened and ignoring it entirely. Very few providers support T.38, maybe one out of a hundred.

That said, you can try faxing over VoIP, its worth a shot at least. You will need the G.711 ulaw codec, and the ATA set up with no error cancellation whatsoever. Turn off all the nifty features. Some ATAs (sipura i think) can have a port locked into fax mode, so it uses a set of presets always to try to make faxes work.
The main problem is the codec- ulaw is good, but not great. To compress audio to 64kbit/sec/channel, you have to lose SOMEthing.
Some people report success by plugging in their fax machine through a DSL filter (small frequency filter that you plug into DSL-equipped phone lines so that normal phone calls don’t interfere with DSL signals). This filters out inaudible frequencies supposedly to make the ulaw codec have less data to deal with.
Some people report luck with reducing their fax transmit speed to 9600baud and disabling error correction (ECM). Other people report more luck without doing so.
If these sounds like black magic-ish ugly hacks that may not work, thats because they are and they often don’t. It really depends on your fax machine and the machine you are communicating with.

You could try a Zap channel, ie a zap fxs card, since it stays as zaptel it never gets encoded to anything.

While faxing over VoIP in its current state may work for you (and it’s worth a shot since you’re only going over a LAN), just about everybody should agree that you will overall get a higher chance of being happy with a standard TDM-based or analog fax line.

//edit- since you are terminating to TDM, you might have better luck with an asterisk-based soft fax solution. Based on SpanDSP, there are a few apps (rxfax/txfax/etc) that can send and recieve faxes from * itself without any hardware. It’s also possible to use asterisk as a fax-to-email gateway in this respect. However, to transmit a fax, users must save their documents as TIFF files and then email them in a certain way. Not quite as easy as load+dial+press send.