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trim
join:2003-08-12
| FreePBX -how to (newbie)
Hi ALL,
I am very new to the PBX systems, but understand some simple networking tcp/ip.
We just opened a business.. running 2 POTS line (1 voice/ 1 DSL/fax)
-We want to add an additional Viatalk VOIP line, and make the primary SBC/AT&T POTS line ring (basically, when the customer calls our main line, 1 out of the 3 (2 AT&T lines, and 1 Viatalk like) will ring.
Can someone give me some pointers on what hardware I need for freepbx.com? Is that my EASIEST OPTION?
Also, perhaps there is a wizard pre-setup out there? All the PBX stuff sounds so confusing!
thanks in advance. | |
druber
join:2000-04-11
Marlborough, MA | I'm pretty well versed in using freepbx, but i'm not entirely sure what you're asking here. can you clarify? are you saying you want your sbc line to hunt to the two voip lines? something else? | |
trim
join:2003-08-12
| apologize for being unclear...
Basically, we want the phone lines to never be "busy" for the outside world.
We will advertise a main Voice line (primary) for our business, and will have about 3 receptionists at the firm.
--Also, when dialing out, is there a way to show ONLY the primary voiceline?? (and not the Viatalk number?)
How should we proceed? (and what is the EASIEST way to implement this?)
Thanks again | |
tommy13v
Premium
join:2002-02-15
Glenville NY
 ·ViaTalk
| You will have to enable or add the feature to your pots line for call forward on busy to get this to work or call the phone company to setup a hunt for the lines so that if the SBC line is busy it can rollover to the other lines that you have. This is not something you can setup in the PBX. | |
druber
join:2000-04-11
Marlborough, MA | note also that call forward on busy may not forward more than one call by default (it doesn't for verizon). if so, the telco can set more than one for the line (i think their terminology is "more than one path") | |
tommy13v
Premium
join:2002-02-15
Glenville NY | That is correct. | |
druber
join:2000-04-11
Marlborough, MA
·VOIPo
1 edit | reply to trim
I would also warn you against trying to use your SBC line with asterisk. The problem is: how do you interface to it? There are two choices:
1. get a (relatively) expensive card from digium (or sangoma) to talk to the PSTN.
2. get a cheap X100P card (basically a winmodem). The problem with the latter is that you have a VERY high probability of having horrible echo problems - it's the nature of the beast. if you get 3 VOIP lines, you can (i believe) just leave nothing plugged into the SBC jack, and any incoming call should be treated as a busy condition.
oh yeah, do NOT even think of moving the fax to VOIP. it may work or it may not... | |
VoIPdevotee
| reply to trim
Okay, here is the way to do this (or rather, one way to do this).
On the two AT&T lines, you want call hunting from the phone company, so if a call comes in on one and it is busy, it will route to the second. Then you probably want to add call forwarding-busy line. There MAY be an interaction between those features, so be sure that the hunting takes precedence (the CF-BL should only take effect if both lines are busy). I am assuming you plan to take incoming voice calls on your FAX line. Personally I'd keep the fax line totally separate and hook it only to the FAX machine, but that's up to you.
Note that if you are in a state where there is a charge for EVERY outgoing local call on a business line, YOU WILL BE CHARGED for forwarded calls! So if there are a significant number of forwarded calls, you may want to just get a third AT&T business line. There may be a way to mitigate this somewhat - since the per-call charge applies only to LOCAL calls, you may be able to do the CF-BL to a toll-free number, if you can find someone that will give you a toll-free number with a very low per-minute rate AND most of your incoming calls tend to be very short.
In any case, you would call-forward on busy to your Viatalk number.
Now, one other word about faxing: If you used Trixbox to set up FreePBX and Asterisk, and if you want your Asterisk box to handle incoming faxes, then I'd suggest you think about switching to Elastix (in preference to Trixbox), because it is supposed to have better FAX support. You still get CentOS, Asterisk, and FreePBX in Elastix, but I have been told they include some add-ons that give better fax support. Note that you can still use all the FreePBX configuration options that you have now.
As for connecting your AT&T lines to your system, there are a few options. Do you want excellent quality connections but very costly, or good quality and reasonable, or horrible quality (noisy if you can get it working at all) and very cheap? If you want the latter, then buy one of those cheap $10 cards on eBay and enjoy pulling your hair out trying to make it work. If you want best quality but expensive, then get one of the real cards that runs around $400 and connects through a card slot inside the computer (Digium makes them but there is another company that also makes them that has better built-in echo cancellation, but I can't recall the name of that one offhand).
To me it sounds like you probably want option 2. That would be a Sipura SPA-3000 or SPA-3102 (make sure you get an unlocked one). The best setup instructions for that are here: »aussievoip.com/wiki/Setup+Sipura···+FreePBX
You can find used SPA-3000 and SPA-3102 units on eBay:
»search.ebay.com/SPA-3000-SPA-310···Zunknown
Make sure the description says the unit is unlocked. Units that have the -NA suffix on the part number are USUALLY unlocked but you should still ask to be sure.
Now in my experience the SPA-3000 works pretty well, BUT, that doesn't mean it will work for you. If you are a long distance from the telco central office (or the remote unit that acts as a central office for your area) then you might experience echo issues that the Sipura unit can't eliminate. However, to me it's worth spending the $50-$75 to try it before going out and springing for the expensive cards, especially if money is an issue - if they work well then you've just saved a significant sum of money, if not you can resell them on eBay. I know there are the "purists" who will totally disagree with me on this and you can feel free to listen to them if you like, but unless you're under some time constraint and don't mind buying the expen$ive cards, I'd at least try the Sipura route. Note that if it works, you'll need one for each incoming line (the unit has two phone jacks, but one is to plug a phone into and the other is to connect to a PSTN line, which is what you want to do. You CAN make use of the phone port independently, for example if you need to plug in phone or telephone device of some kind. With the correct configuration settings the phone port and the line port can be totally separated from each other and used independently).
If you do decide to go the card route, do your research. There are better choices than what are perhaps the most obvious ones. As I say, there is one card (not the one from Digium, but I just cannot think of the name of it - for some reason I'm thinking that it's either Sangoma or Rhino, you might check those out) that has better hardware echo cancellation. Since echo is the biggest problem you typically encounter when connecting Asterisk to a PSTN line, if you are spending the big bucks on a card you want the one with the best echo cancellation. I'm not the person to ask about this because I've only used the Sipura 3000, which works well enough for me - the only real issue I've noticed is when calling out it sometimes distorts the ringing signal, but once the other party answers the distortion seems to clear right up. I did have to play with the incoming and outgoing volume settings to get them right, but it wasn't difficult, just a little time-consuming (change the level, make a test call, repeat until you get correct levels).
Hope this helps. | |
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