SUGGESTIONS PBX Questions (Noob) - New to VOIP/PBX

[Nallchan]

Good Afternoon,

Ive got a few questions if someone would be so kind as to guide me a bit... Im no stranger to networking / Computers.. but i've honestly never done a PBX before (UNCHARTED TERRITORY for me!)

Ive been reading a few of the Nerd Vittles Specifically ( http://nerdvittles.com/?p=14787 )
Watched a few youtube videos... but im missing some basics and some specifics..

Alot of what I am Missing is Terminology...( I understand what SIP Trunking is... and im rather network savy (CCNA/Network+ etc)

Here is what I've Done:
- Purchased a Raspberry Pi 2, (32gb MicroSD)
- Loaded said Raspberry Pi 2 with Raspbian Wheezy and Incredible PBX
- Logged into the Web GUI (browser) / SSH via putty no problem.
- I have a Google Voice Account with a Telephone number
- I have a USB to RJ-11 Adapter (says Its Linux Compatible)
- I have a NETTALK Voip device (Wife uses it to talk to her family overseas)
- About to purchase a Cheapo IP phone: (Stop me if this is the wrong decision)
http://www.amazon.com/Cisco-SPA-303-3-Line-Phone/dp/B0041ORNJ2/
- Business class internet (50 down / 10 up)

Here is what I am TRYING to do:

1) Utilize the Nettalk Number to TAKE the calls...the PBX auto answer.
(Because ive already given out this number NUMEROUS times) - Im flexiable in this approach as long as I dont have to give out a new number.

2) Id like my friends/Users/Customers to call in on my Voip line and be presented with a simple Menu
- PRESS ONE FOR ME - (then it calls my Cell Phone)
- PRESS TWO for my Wife - ( then it calls HER cell phone)
- Press THREE for my Dog - (then it rings the phone in my house) - presumably an IP phone
- Maybe goto voicemail if no one answers??

(I figured out the Music on Hold thing already)
(how does the PBX understand a FAX vs a person calling? - I assume it listens for the fax noise and Turns on HylaFax/AvantFax??


Also:

Im willing to toss a few bucks someones Way if they want to take the time to hold my hand / possibly remote in (Teamviewer) or Skype me and Guide me thru the process to a functional point.. (Im not rich)

 

[Nallchan]

If someone has a youtube video that covers a majority of this, im perfectly happy learning that way..

 

[Addisonb]

Okay,
Here goes.
To start I don't quite understand how the NETTalk works just giving their website a once over. Do you have one of their DUO devices, does it create a pots line that you then have a generic home phone plugged into? Is that what your usb to RJ-11 adapter is for?

From here I am going to presume that it is creating a pots line and you want to use that as your "telephone line" just like you would get if you called the local home phone provider and said "give me a telephone line."

I could be wrong but I doubt that your usb to RJ-11 adapter will work. What you need is an Analog gateway aka POTS gateway. These are typically one or 2 port devices that connect to the phone line and convert it to SIP. (SIP is Session Initiation Protocol, it is a VOIP protocol but you could almost use them interchangeably because SIP is what everyone non proprietary uses.)
This is an example of one: http://www.grandstream.com/products...alog-telephone-adaptors/product/handytone-503
Grandstream is the cheap Chinese brand everyone loves to hate but if you ask me they work fine, I don't expect Cisco quality and I'm not disappointed when they show up. They also make cheap IP phones which would be a cheaper alternative to the Cisco you have noted. There are other better alternatives then the sticker shock of the Cisco phones. You're not installing a big enterprise system. There are other better analog gateways than grand stream. Search any of the big sites like voipsupply or telephonydepot.

Back to the Analog Gateway. The one I linked has 2 ports, one FXO and one FXS. You would need an FXO, an FXO "receives" a pots line from someone generating it (your phone provider) and a FXS generates a phone line so you could use a cheap phone from goodwill as an extension in your system.

Once you have the Analog Gateway setup it generate a SIP account to register your freepbx to. You would add this as a Trunk (SIP) in your PBX. A trunk is basically a pipe or tunnel or connection into or out of you pbx.

To be honest, I have never setup an analog FXO so the following info is how you would do a IP trunk or PRI circuit, I am going to make some guesses as to how an FXS will work.

I'm guessing that in the analog gateway you configure that when the line is ringing it is going to tell you PBX that a call is being made to your phone number, in my example 4805554444. Your system then needs an inbound route, by default you should have a default route, if you had more than one phone number that needed to different things you would need at least one inbound route per phone number.
There are three most important fields in an inbound route. CID, DID and Destination.
CID is the caller ID of the person calling your system, DID is the number that the person dialed and destination is where you want that call to go in your PBX. You will most likely only put a DID and no CID in your inbound routes and that says anyone (no specific CID because we left it blank) calling this DID (that we entered) gets sent to this destination.

What you are describing "press 1, press 2, press 3" is an IVR in freepbx. So the destination of your inbound route would be an IVR, where you would then configure 1, 2 and 3 which then gives you 3 more destination fields just like you had in the inbound route. In your case they would all be extension. A virtual extension (created on the extensions page) for your cell, a virtual extension for her cell and a generic SIP extension for your shiny new IP phone.

But wait!!! ready!! In the extension pages there are 3 more destination fields, they say something like "unavailable message if enabled," "busy message if enabled" and "unavailable message if enabled." You said they would go to voicemail if not answered, by default when you create an extension voicemail is disabled, you could enable voicemail on all 3 extensions and calls intended for your cell would email the voicemails to you, calls intended for her cell voicemails would get emailed to her and then a home voicemail.


But what is going to happen when the system tries to call your or her cell is nothing, because the system doesn't know how to get calls out. So, we need an outbound route, which tells the system "if I dial these numbers, send it out of the system on trunk X," I believe you want this to be your google voice, which should be setup as a trunk. So you will create an outbound route and in the match fields you will configure XXXXXXXXXX. That tells the system that if I dial any 10 digit number, send it out on the google voice trunk.

The only thing I didn't cover was how to add your cell numbers to the virtual extensions. You will go to the follow me tab and browse to your virtual extension and enter your cell I'm the box which is like the second option and change the "initial ring time" to 0. If you want to know more about the options in the follow me then look at the wiki.

I'm too lazy to proof read this, so apologies. Please ask questions.

 

[Nallchan]

Appreciate the Reply AddisonB... so THIS is what I understood for physical Topology. (more below picture)

1) Yes, the Nettalk Takes an RJ45 connection in and puts out a standard RJ11 "Pots" telephone Connection that you can plug any cheapo telephone into

2) Now that I understand that the PBX software needs a SIP Account (IP / Registration) instead of just an Interface to point at.... yes.. Ill Buy an FXO Analog gateway device - Saw this one on Amazon?

LINKSYS SPA-3000 VOIP FXS FXO PSTN UNLOCKED Phone Analog Adapter
http://www.amazon.com/gp/product/B00HJIJ59I?keywords=FXO&qid=1453469977&ref_=sr_1_4&sr=8-4

(Diagram on Amazon)

3) Thanks for the CID / DID Definitions

4) Ok IVR is what I was thinking (for some reason, Call group was the term stuck in my head)

5) " That tells the system that if I dial any 10 digit number, send it out on the google voice trunk."
I would assume 7 Digit numbers also...?

6) I Figure if the line rings to the local Voip Phone.. goto voicemail locally (New IP phone).. else im OK with it going to the Cell phone's voicemail.

Appreciate all the info! let me know if i misunderstood anything... THANKS!
Ill get back as soon as the new IP phone and FXO device arrive.

 

[Nallchan]

Another Question: ..
If i didnt have the NETTALK... Would i just use TWO Googlevoice Lines? or have to buy a SIP service?
(Im assume one line in.... One Line to forward calls OUT... to the cell phones etc)

 

[Addisonb]

2) that Cisco would be fine. They may even make usb Dahdi adapters but I've never seen any.
4) a ring group will ring multiple people all at the same time and the first person to answer wins.

5) no, you would need to add a second field for 7 digit, 911 wouldn't work and neither would 14805554444 (1 on the front) because now that's 11 digits. Your outbound routes will usually vary depending on what your phone line provider is expecting to receive. Google voice should be pretty forgiving.

6) then just leave voicemail disabled on the 2 virtual extensions

Net talk) so this is where VOIP/Digital telephony gets interesting. Phone number does not equal phone line.
A phone number (DID) is an endpoint address, think of an IP address. A channel is a "line" that can carry one call of audio in and out.
For every call you want to be able to make at a time you need a channel. (Someone calling your system, then your system calling your cell is 2 channels) I believe but don't know because I don't use it that google voice gives you two channels on a free account.
For some providers, inbound and outbound channels aren't the same. I use Vitelity and on their unlimited plan you get 4 inbound channels, but they charge you outbound minutes so there's no practical limit on the outbound channels. I can keep making concurrent calls until the cows come home but the 5th person to call me gets a busy tone. It all varys.

One fun thing that you can do with providers like Vitelity is for a couple bucks a month, you can have a phone number in a foreign country (where your wife's family lives) and then for them they dial a local number and it calls your system.

Another thing I meant to mention in my first post was you can start to do tricks in things like your inbound routes that say "any call to my number (DID) with a CID that starts with the area code of your wife's family, will automatically go to her cell" cause we know you definitely don't want to talk to your inlaws.

 

[Addisonb]

I will say that this isn't how I would do it. I know you're married to the nettalk number, but if you think this is a system you will actually use long term then VoIP to POTS back to SIP and out of your system via Google is a bit of a mess.

You could likely port the number to a native SIP provider and eliminate a lot of components. Now it's just provider, Internet, pbx and phone.

Any extra cost you would spend porting the number would be less than the cost of an FXS.

 

[Nallchan]

I will say that this isn't how I would do it. I know you're married to the nettalk number, but if you think this is a system you will actually use long term then VoIP to POTS back to SIP and out of your system via Google is a bit of a mess.

You could likely port the number to a native SIP provider and eliminate a lot of components. Now it's just provider, Internet, pbx and phone.

Any extra cost you would spend porting the number would be less than the cost of an FXS.

I haven't Purchased the LINKSYS SPA-3000 yet, and I definitely understand the more components in the chain, the more likelihood of failure.

Just picked up 2 of these: ( CISCO CP-7970G)
http://www.ebay.com/itm/271657296943?_trksid=p2057872.m2749.l2649&ssPageName=STRK:MEBIDX:IT

So your suggestion is Pick a SIP Provider and port the NetTalk Number...Im really honestly not against that...
Just gota find one that allows calls to Mexico (for the wife) AND will transfer the number.... hmmmmm

 

[jerrm]

Agree with dumping Nettalk if you can port the number away from them. Place it with Google and do away with Nettalk completely. Alternatively just forward the Nettalk number to the GV number.

Haven't ported a number to Google in quite a while, use to be they only ported in mobile numbers, not sure if that's still true. If so, steps to port to a cheap pre-paid SIM and then to Google can be found online.

One GV number should be all you need, as it can handle multiple calls in and out. Be aware calls transferred to the cell phone will show the GV number as the Caller ID. If you use a SIP provider that accepts any CallerID you may be able to preserve the incoming CID when transferred to the cell.

An alternative to IP phones for the house would be an FXS adapter set up as an extension. Disconnect the internal wiring from the outside, plug the FXS adapter into the wall, and all analog outlets in the house should be live allowing you to use any existing analog handsets.

 

[jerrm]

So your suggestion is Pick a SIP Provider and port the NetTalk Number...Im really honestly not against that...
Just gota find one that allows calls to Mexico (for the wife) AND will transfer the number.... hmmmmm

The beauty of VOIP is you can mix and match. Use GV as primary for free US calls in and out, use the best domestic rate SIP provider for calls forwarded to your cell if you want to preserve CID. Use another for the best Mexico rates. Apply the proper call routing, and it is all transparent in daily use.

Depending on call volume, getting a Mexican DID may be worthwhile. For instance, Voip.ms offers Mexican DIDs with unlimited inbound calling at $8.50 per month. It takes a lot of minutes to add up to $8.50 per month, just need to do the math. Looks like voip.ms termination rates for outbound calls to Mexico are $0.0064/minute for most non-mobile numbers.

 

[Nallchan]

Ok - According to Google Voice, you CAN NOT port Nettalk Numbers to GV.
- https://productforums.google.com/forum/#!topic/voice/SChgC8msG3A
Also typed my number to verify
- https://www.google.com/voice/b/0/porting


Reading some online Forums about porting the number to a cheap Cell Carrier THEN to GV...
http://forums.anandtech.com/showthread.php?t=2222963
https://productforums.google.com/forum/#!topic/voice/6pgR4LXRRGc

still researching on who to port to....

 

[Nallchan]

The beauty of VOIP is you can mix and match. Use GV as primary for free US calls in and out, use the best domestic rate SIP provider for calls forwarded to your cell if you want to preserve CID. Use another for the best Mexico rates. Apply the proper call routing, and it is all transparent in daily use.

Depending on call volume, getting a Mexican DID may be worthwhile. For instance, Voip.ms offers Mexican DIDs with unlimited inbound calling at $8.50 per month. It takes a lot of minutes to add up to $8.50 per month, just need to do the math. Looks like voip.ms termination rates for outbound calls to Mexico are $0.0064/minute for most non-mobile numbers.

yeah ... gonna cost me $2 a month for the amount she calls mexico at .0064 a minute.. sounds perfect...
so yes... I need to figure out how to route calling rules as you suggested... any tutorials I can read/point me towards?


also - ""One GV number should be all you need, as it can handle multiple calls in and out.""

so wait... someone can CALL me on my GV line... and IT can call out to Forward the call to me cell on the SAME account / line???!

 

[jerrm]

yeah ... gonna cost me $2 a month for the amount she calls mexico at .0064 a minute.. sounds perfect...
so yes... I need to figure out how to route calling rules as you suggested... any tutorials I can read/point me towards?

Just google "FreePBX outbound routes"

also - ""One GV number should be all you need, as it can handle multiple calls in and out.""
so wait... someone can CALL me on my GV line... and IT can call out to Forward the call to me cell on the SAME account / line???!

Yes.

 

[Nallchan]

Just google "FreePBX outbound routes"


Yes.

Thats fantastic

 

[jerrm]

still researching on who to port to....

Last few I did were TMobile. Think it was less than $10 in TMobile fees for SIM/activation/porting, but again, it's been a while.

 

[Nallchan]

I was able to transfer my number to Vitelity for $5 .. and $1.50 a month ..gonna try that out

 

[Nallchan]

Ok... Just an Update.. I started out with "IncrediblePBX" as outlined in the guide - had no problems building it...

After a day of learning where everything is and how to do things.. I kept running into my Sip trunks (Vitelity) being dropped after about an hour...the system locking up and requiring a reboot... just FIGHTING to get things to work the way i wanted it to...

I researched everything i could think of... power settings.. any log errors.. nothing said SCREW IT and rebuilt it using Freepbx from http://www.raspberry-asterisk.org/ (Zero Affiliation btw) and i seriously could not be happier.. it was pre-built (used the http://www.raspberry-asterisk.org/downloads/beta-image/ )... worked better / faster... and Zero issues with it locking up / dropping trunks etc... after being frustrated with the previous install ..

just to be clear, im not bashing incrediblepbx, but my results / happiness with the new install were amazing in my opinion

Ive got the number transferred to Vitelity, it comes in the pbx.. made my own announcements, forwards to my cell phone / voicemail as needed..
Now i just need to optimize which trunks are used for certain dial rules...super excited..

 

[Nallchan]

I think im going to make a noob guide of everything I learned ... pictures / arrows etc...for the TL : DR people like me heh