Newbie to Asterisk need a little nudge

New to the forums, been reading for days, Learned quite a bit, I have experience with some PBX’s and Key Switches, so the terminology isn’t throwing me too much, and I have read the installation instructions so I think I am set there. This is my scenario and plan, and my confusion is with the VOIP providors.
First, I have a small technology consulting company.
I have 4 PSTN Lines 3 for voice 1 fax. So that tells me I need a 4fxo and 1fxs (correct if i am mistaken any of my assumptions)
My plan is to use * with all these great features to replace a Panasonic Digital DBS. So out of the box * with SIP phones replaces my antiquated switch. Now the confusion,

  1. I want inbound answered calls (receptionist) to the office to be routed/transferred to me or techs when we are working away from the office. Assume Broadband connection is always available.
  2. I need a busy no answer mode to forward to a cell phone (blackberry) I have searched but not aware of any VOIP client for blackbery except there Enterprise Server and pluggins and I dont want to go there.

Do I need a VOIP providor for this? My initial thought is no, but to add LCR and other features like DID and trunk outbound overflow pay as you go trunks yes. If I got VOIP providor with 5 DID’s, over two lines (Does DID over VOIP still work like tradional DID), and routed those on * to the appropiate extensions, with busy/no answer forwardind to IP phones or external numbers would that work.

Your insight and a manifesto or guidline is most appreciated, also any suggestions or experience with a VOIP vendor that fits this scenario would be appreciated.

First, I’d leave the fax out of the system… If you have 4 pstn lines, and 1 is going to the fax, just leave it that way. There’s no reason to run it into an Asterisk box unless you intend to use it for an alternate outgoing line. (Which would block incoming faxes.)

3 FXO ports would be just fine for the remaining lines.

It is NOT necessary to use VOIP for outgoing busy/no-answer calls. If you’re trying to get calls from your office to reach your Blackberry phone, you can just use PSTN lines the way you do now. You can receive a call on PSTN line 1 and either:

a: Flash the line, and transfer the call to your Blackberry.
b: Place a second call out on PSTN line 2 or 3, and connect the two calls together.

You can also re-direct the call to a VOIP provider. Most VOIP providers can complete calls to the public network as well, so a VOIP client on your Blackberry isn’t necessary. Even if you decided to go with a VOIP provider, your calls could easily reach your cell.

Generally for DID, you would get 5 individual VOIP accounts, and just have the Asterisk system register to all 5. It’s not quite the same as TDM DID.

Depending on the provider, you may (or may not) have the ability to receive multiple simultaneous calls on any of your individual lines. Some providers allow only 1 call per account. If you want/need three simultaneous calls to the same number come through, you should shop around for such a provider.

What is the difference? is one manual and the other automatic?

a: Flash the line, and transfer the call to your Blackberry.
b: Place a second call out on PSTN line 2 or 3, and connect the two calls together.

And with only three lines, I would only have one line vacant, hence using VOIP for a low cost solutiuon. Can I make multiple outbound calls with one line?

Does anybody have a sugestion a provider, that provides ability to receive or send multiple simultaneous calls on any of your individual lines.

The difference isn’t “automatic vs manual” or anything.

I guess we should step back a bit.

Asterisk is best described as a “development platform”. Once you load it up it won’t do much at ALL. You have to put it all together yourself, and you do that with your dialplan.

You could have three different phone numbers, and each one would be answered and handled differently if they were not answered. You just have to sit down with the functions, and put them together the way you want.

Here’s an example:

exten => 4321,1,Dial(SIP/4321,15)
exten => 4321,2,Voicemail(u4321)
exten => 4321,102,Voicemail(b4321)

This will call extension 4321, and if it goes unanswered will transfer the call to voicemail. If the station is busy, it still goes to voicemail, but plays a different greeting.

Now, you COULD have the system send calls to a different number instead.

exten => 1234,1,Answer
exten => 1234,2,Dial(SIP/1234,15,t)
exten => 1234,3,GotoIf($"${DIALSTATUS}" = “NOANSWER”?:4:5)
exten => 1234,4,Followme(1234)
exten => 1234,5,VoiceMail(u1234)
exten => 1234,6,Hangup

In this example, the calls will follow the plan described for extension 1234 in the followme.conf file if the call is not answered. That destination could be another station in the system, your cell phone, or anywhere you want.

The route that the call takes out of the system is pretty much up to you. You can choose to send calls out your PSTN lines, your VOIP service, or some combination. Again, it’s just a development decision. You might want to send the calls out a VOIP line to keep your PSTN lines clear. You might choose the VOIP route because it’s cheaper. You might choose to flash the PSTN line, and transfer the call to the cell so that it can take another incoming call…

exten => 1234,1,Flash()
exten => 1234,2,SendDTMF(16175551234)
exten => 1234,3,Hangup()

It’s all up to you.

I would reccomend you begin by downloading a copy of Asterisk TFOT here:

The book will give you a better idea of what functions you can use to do the things you want.

I would also build a test platform first. Just get an old PC, load it up, and put it through it’s paces. You don’t need to buy anything if you use one of the many free softphones out there. Your current desktop PC can be a phone in the system for you. Once you get a better idea of what the system is capable of, you can better plan what you’ll do with the system.