[SEEKING ADVISE] Getting started

I just heard about asterisk from a friend and is pretty interested in it, so I tried to do some research on. However, maybe I’m relatively new to this field, after hours of google/wiki/glossary queries… I’m still not sure if I really understand what’s asterisk about, so I’ve decided to ask my questions here.

  1. Traditional Telephony
    I noticed that most articles for asterisk is about VoIP. However, I’m currently using a traditional phone line (actually it’s a “digital phone” but I guess it won’t make any difference: http://www.shaw.ca/en-ca/ProductsServices/DigitalPhone/HowItWorks.htm) as my primary phone line. So, I’m wondering what asterisk can/cannot do with that. Can I implement features like caller log/speed dial/conference/answering machine with a traditional phone line? What else? And what are the limitations? Also, (in addition to my old fashion, traditional phones) can I answer/dial calls from my networked computers without additional hardware? (i.e. making ‘software’ call) How about fax?

  2. VoIP
    In addition to my traditional phone line, I also subscribed to a HK based VoIP service (which is, I can dial/pick up calls from HK as if I’m in HK here in canada). It currently works by connecting a VoIP box provided by the service provider to my router (and to a traditional phone on the other end). Could I replace that box with an asterisk server? If yes, what’s the benefit/what ‘extra features’ could I get from that? If not, can I connect that line to the server as if it was a traditional line (via the ‘box’)?

  3. Multiple lines
    As mentioned above, I am currently subscribing to 2 land lines (Local number: ‘digital’ line; HK number: VoIP line). Can I connect them both to the asterisk server? Could I forward calls from both lines to one set of traditional phones? (This would allow me to dial both CA/HK numbers from the same phone.) Is it possible to ‘bridge’ the two lines? (For example, incoming call from line 1 -> asterisk ‘relay’ -> dial another number by line 2 then create a ‘conference’) Also, I be make separated, independent calls from both lines at the same time? (perhaps one from phone on from computer?) Are there any advantages/disadvantages for connecting more than one line to the asterisk server?

  4. Hardware
    I have a spare pc currently running ubuntu. It’s a P4 1.8 (i guess) with ~512MB DDR. Is that generally enough for running a phone server? If not, how powerful will that have to be? Also, what extra hardware do I need for connecting traditional phone lines? (I’ve heard that modems can’t do the trick.) How much would that cost? Any extra requirements if I need to connect more than one outgoing line? How about my phones? Any special requirements? Will I lost the ‘caller ID’ feature on my phone if there’s an extra asterisk layer between the line and my phone? I’ve also heard that you can actually run asterisk w/o any extra hardware. What can asterisk do w/o connecting to traditional phone lines? (I’m assuming it’s like VoIP stuff?) Do you need to subscribe for any service for that to work?

  5. Management
    What’s the difference between installing asterisknow and installing that on top of an already installed linux? How is asterisknow ‘optimized’ for running asterisk? Can I run extra service like apache on asterisknow? (and would that be considered ‘too much’ for my machine?) Is it possible to manage the box via lan? (ssh/console: fine; gui/web based: great) Can I do the ‘cool stuffs’ such as filtering/redirecting/monitoring (i.e. routing) calls with asterisk?

  6. Any other comments/suggestions/advise?

Thanks for reading this post (sorry I know it’s long :stuck_out_tongue:)
Thanks in advance for any advice/comments :smile:

i just noticed that most users here are for the business field… so just to make it clear, i’m just a home user and is deploying * mainly for experimenting… (and hoping to make free long distance calls outside home by bridging the two lines)

  1. You can use POTS (“regular” phone lines) on asterisk however you will need to purchase some hardware for this. Asterisk supports a wide variety of ways to interconnect to the PSTN (POTS, T1/E1, ISDN etc).
    Asterisk is limited to the users imagination and creativity. It has call logs, conferencing, you can set up multiple voicemail box’s, etc. On the system you can use soft phones (a software based voip phone on you computer), IP hard phones, and POTS phones (again you will need hardware for it). IMHO the amount that it costs per port for an analog phone you may was well go with a lower end IP hard phone that will give you much better functionality. Asterisk at the moment supports incoming faxing (with the SpanDSP patch). There are ways of doing outbound faxing but I have never implemented it. To clarify the faxing support that I have mentioned is to have asterisk pick up the phone and convert the call to a pdf. If you have the hardware cards to support POTS then you should have no issues. Also asterisk 1.4.X supports T.38 pass through.

  2. It depends on your provider. A lot of providers want you using the “box” (aka ATA) so they can limit you to one call at a time. If your server registerd directly with your provider there are many advantages. For 1 you get rid of the extra box and you get all the functionality of asterisk such as IVR’s, CallerID manipulation, Call Recording, Conferencing and the list goes on.

  3. You can run as many lines as you want to asterisk. If it’s a voip line see if you can get the information for it (username, password etc.) and register directly. You can do call bridging (“patching two lines together”), you set rules as to what happens when (i.e. if your mother in law calls to just hang up on her, if its your wife to have the calls go to your cell phone etc).

  4. The specs that you have listed should do well for a few calls. Extra hardware will depend on what kind of lines you use. You can pick up these cards on google. You can have as many phones as you want connected to asterisk. The beauty of asterisk is that you are not limited to x amount of extension, lines etc. (provided that you have the right hardware etc.) Asterisk fully supports CID. Again if you are strictly using VOIP you do not need any special hardware. You do need any licecne to run asterisk. There is a version for windows but I would stay very far away from it. There is a lot more support out there for Linux.

  5. AsteriskNow is asterisk loaded on CentOS with a GUI. I have never used it but there is a lot of positive feedback on it. IMHO the only real way to run asterisk is to code it yourself. When using a GUI you are limited to the imagination of the one that created it. One you get the hang of asterisk it won’t be hard. I have written many scripts which would be hard to incorporate with a GUI. IMHO GUI’s are not the way to go for a “real asterisk user”. The best advice that I got when starting to use asterisk was to stay away from GUI’s. I came from a windows environment. It took a while to learn linux and how to code asterisk but it was well worth it.

  6. Have a look at my post here:
    forums.digium.com/viewtopic.php?p=59418#59418

Don’t worry about all the questions. We were all new to asterisk at some point. It’s open source. I believe what comes around goes around.

Good Luck.

Actually, AsteriskNow runs on rPath Linux. (see www.rPath.com) AsteriskNow is more of an “out-of-the-box” PBX and is very easy to install. All you have to do is download the ISO, burn the CD and follow the prompts. Personally, I don’t think there’s anything wrong with using AsteriskNow to get started. You can get an idea of Asterisk’s capabilities and it’s easy to browse through the .conf files and gain an understanding of their purpose. When you’re ready to move to Asterisk, I suggest you get the book “Asterisk: The Future of Telephony”, 2nd Edition. It will get you up to speed quick. Make sure to get the 2nd Edition. If you just search for the title on Amazon, the 1st Edition comes up.

Wow thanks for your reply Dovid and Dwag! I have to say that you guys have been REALLY helpful:)

But then, there’s still a few thing that I don’t fully understand.

I’m assuming that you don’t really need an ‘carrier’ for making voip calls… since your call is encoded to some UDP packages and they’ll make their way to the other end through my internet connection… so if i’m subscribing for a so called ‘voip package’, is it like subscribing for a IP <-> PSTN service + a phone number that’d translate into my ip address? where can i subscribe for such kind of service? can i make >1 concurrent call with 1 subscription? how does it work? and are there any such service provider for home users in canada? (not those that provide you an ATA box) if i subscribed to such service, is it true that i’ll only need an asterisk server (w/ no extra h/w) + a few (possibly software) ip phones?

and for ip phones… any suggestions? any ‘starter’ model? i notice that they have ‘single line’ and ‘multiple lines’ models… what does that means? shouldn’t i be able to make unlimited calls if my network could support that? (at least internally) and most phones said they support 3-way conferencing… is it possible to do 4(+)-way conferencing with those phones (by doing sth on the server side?)

and… are they any (gui) tools that allow me to do ‘real time’ (internal) call routings? like… if i picked up a phone and it’s looking for someone else… i can press a few buttons and route it to another extension, or perhaps join the conversation with someone on another extension?

Thanks again :smile: (excuse me for my baddd english)

If you just want to call from one voip phone to another using your asterisk box then you do not need an ITSP (aka carrier). If you want to have incoming and or outgoing calls to the PSTN then you will need to sign up with some ITSP. I am unsure who provides numbers for Canada but you can try:
flatplanetphone.com


telix.com
With the following providers above you do NOT need an ATA. The amount of concurrent calls you can have at once depends on your ITSP.

If you want to start messing around I sugest you get a soft phone. X-Lite ( counterpath.com/xlitedownload.html) is my favorite free softphone. For a physical phone I would recomend you get the SPA-941 (it is a bit more expensive than the SPA-921 but you can have multiple SIP accounts on it). You can get it from:
telephonydepot.com/product_p/105-054-941.htm
or
voipsupply.com/product_info. … ts_id=1057

FOP (voip-info.org/wiki/view/Aste … ator+Panel) Should be able to help you with real time calls.

Good Luck

Correct… Actually, you aren’t required to get a phone number if all you want to do is make outgoing calls on the PSTN. If you want to receive calls from the PSTN, then you will need a phone number.

I currently use Junction Networks and Teliax. Teliax allows you to make and receive an unlimitted amount of concurrent calls with one subscription. Last time I checked, Junction Networks limitted your concurrent calls to 25. That may have changed… I don’t know if those providers can provide Canadian numbers. You will have to check it out.

The providers use the SIP and/or IAX protocol. Do some Googling… There’s plenty of good explanations of the SIP protocol on the web. You don’t actually need an Asterisk server to connect to the providers. You can use a soft or hard phone to register to their service and make and receive calls. If you want an auto attendant, voicemail, conference bridge, call routing, etc., then you need Asterisk. You do not need any addtional hardware for Asterisk if you are using a VoIP provider.

My suggestion would be to check eBay for Snom 320s or 360s. I’ve seen them go for dirt cheap lately. I bought two 320s for $58 each. Although I perfer Polycom phones, the Snom phones have some advantages for someone just starting out. First, The Snom phones boot up quick where the Polycoms take forever. Second, the Snom phones have a great HTTP interface so you don’t have to do central provisioning. And finally, as I mentioned, the price. The Snom 320 supports 6 lines. That means you can receive 6 calls before the phone will be busy. The Snom 320 supports 3 way conferencing in the phone and you can do >3 conferencing using the conference bridge in Asterisk.

Yes…