I might be getting a free Raspberry Pi soon and I was looking up ideas as to how to take advantage of it. One idea was to use it with Asterisk/C*Net for VoIP phone calls,
Currently, I have a Panasonic 308 PBX and a Panasonic 824 PBX that are connected to the PSTN (they're actually chained to each other so I can call from one PBX to the other). Currently, I have 7 telephones - 4 rotary and 3 pushbutton - with 5 telephones on the 824 PBX and 2 on the 308 PBX,
I believe it would be possible to connect these PBXs to the Raspberry Pi, but I want to make sure I do it right,
There is a website out there for Asterisk on Raspberry Pi: http://www.raspberry-asterisk.org/
Someone made a YouTube video on how to install it: https://www.youtube.com/watch?v=Qgg3zkqh4js
I'm wondering if I would need http://www.asterisk.org/products/telephony-interface-cards or http://www.asterisk.org/products/voip-gateways as well,
I'm completely unfamiliar with all these products and their purposes. Can anyone let me know if this project is possible and feasible, and what hardware I would need to accomplish this?
Since I already have 2 PBXs, I'm not sure if it's necessary to have FreePBX (can I just installed Asterisk on the RP?) I'd like to just connect one of the PBXs to the Raspberry Pi somehow,
From there, I'm hoping to hook it up to C*Net, since I don't think you can just make free calls with Asterisk.
You would need he Raspberry Pi to interface to C*Net, handling the routing between C*Net and your systems. Once you have the routing, then you will need an interface (usually an ATA) to handle the translation between the VoIP SIP connection from FreePBX to output an analog signal that the Panasonics can understand.
The hardware to talk to the FreePBX switch can be an ATA (providing 1 or 2 FXS lines), a separate box providing FXO/FXS capability, a card that runs inside a PC that provides FXO/FXS capability, a physical VoIP phone, or a softphone that runs on your PC or Smartphone. (FXO: interfaces to an external system which Receives dial tone and ringing signals - aka incoming lines on the Panasonic system/FXS: generates the dail tone/ringing for a physical phone to use - aka extensions on the Panasonic).
Cost depends on the model, but the Cisco SPA-122 ATAs are rock solid, but only understand tone calling. I also like the Grandstream HT502, as it provides 2 lines and understands pulse dialing - just beware when purchasing, as they are older hardware, and are often not new, and may not work (I bought a lot of 4 on eBay and not one worked...). Others may have models to recommend...
I use a mix of ATAs and Smartphone VoIP clients to interface with my VoIP service - smartphones work great, since they can go nearly anywhere you have an Internet connection - be it cell or WiFi.
Hope this helps.
Jim
Well, I only use rotary phones, but apparently Panasonic PBXs convert pulses to tones for all external calls, so I guess it isn't super important if the ATA/translation hardware doesn't understand pulses, though that would be nice.
I'm looking for a low cost project to connect the PBX to C*Net using a Raspberry Pi.
Also, I'm not familiar with VoIP technology much. I know what it is, how it works, and how calls are handled, but not much about VoIP hardware.
Do you have any particular recommendations, based off of experiments?
I wouldn't use a smartphone, softphone on the computer, or a physical VoIP phone for sure.
My understanding with ATAs is that you only get a single C*Net number and someone has to host you, is that correct? And is that all free?
Is there any physical hardware besides an ATA I can use?
Also, I've heard some hardware requires configuring ports on the router and stuff. I won't be able to get an ports opened or anything, so I'm looking for something that I can just plug and play - no configuration on the router required.
So I'm aiming for this setup:
Phones -> Panasonic PBXs -> Translation Hardawre (ATA, etc.) -> Raspberry Pi --> Router
I'm probably going to be using an Ethernet cable for any network connections required.
Wait a minute...
If I have an ATA, I don't even need a Raspberry Pi, because someone else could host it right?
So, with a Raspberry Pi, I'd be able to host myself, right?
Now that I think about, I would like native pulse dialing support.
And what is the difference between a telephone card (https://www.digium.com/products/telephony-cards/analog/4-port)
and an analog telephone adapter (http://www.ebay.com/itm/263091705895)
And what about VoIP gateways? http://www.asterisk.org/products/voip-gateways
I'm really confused.
Can I use either/or?
IF you can find someone to host you on C*Net (and that's a big IF), then you just need either an ATA, softphone, or hardware VoIP phone.
If you setup the Pi, you may need some additional configuration (possibly router ports opened and/or Something like DynDNS so that outside calls can connect in. I haven't gotten my switch up and running yet, so I don't know the requirements right now.
Even if you don't get connected to C*Net, you can use the Pi to manage connectivity between the two PBXs (vs. daisy chaining them together).
If you do go the Pi route, you will need multiple ATAs and/or soft phones. Softphones are almost a necessity to help troubleshoot connections, dial plans, etc.
I'll answer hardware in a separate reply.
Jim
About hardware:
Native pulse dialing is extremely limited and often not officially supported, or could go away in a future firmware update of the device. Your Panasonic handles the translation between pulse phones and the outside, so you should be good.
An ATA is the simplest interface to between a phone and a VoIP system - connecting with Ethernet (as long as the ATA is configured properly).
The Digium card is something that goes inside a PC running VoIP software to provide an analog interface.
A VoIP gateway is a device which connects a circuit (T1 or T3) from a provider which provides channelized voice circuits (a PRI T1 provides 23 Voice channels + 1 control channel - 23B+D). VOice T1 Lines aren't cheap, but often used in business for office phone service. [My previous employer had a T1 in our corporate office to support our non-VoIP Intertel System of 40 extensions - with a T1, there can only be 23 external calls at one time).
Hope this helps.
Jim
Sorry, I'm still confused as to the difference between an ATA and a Digium card.
Quote from: bellsystem on July 20, 2017, 09:50:36 AM
Sorry, I'm still confused as to the difference between an ATA and a Digium card.
Digium
https://www.digium.com/products/telephony-cards/analog
ATA's
http://www.cisco.com/c/en/us/products/collateral/unified-communications/ata190-2-port-adapter/datasheet-c78-731473.html
http://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors
https://www.voipsupply.com/voip-adapters
That did not help clarify anything. It seems they are both hardware that turns/converts an analog device into a digital one. What are the differences between them, especially with regards to how they would be used with C*Net?
Quote from: bellsystem on July 20, 2017, 01:49:55 PM
That did not help clarify anything. It seems they are both hardware that turns/converts an analog device into a digital one. What are the differences between them, especially with regards to how they would be used with C*Net?
You have an ATA so you know that an ATA is a stand-alone device that connects to an Ethernet port. As was stated, a Digium card goes into a motherboard slot
inside a PC.
I don't have an ATA - not sure how I managed to communicate that...
Anyways, that explanation helps.
So, my understanding is:
If you ONLY have an ATA, someone from C*Net has to host you.
If I have a Raspberry Pi, can I host the ATA using that instead?
Also, which option will not requiring doing anything with routers and opening ports? I won't be able to do that.
Also, if pulse dialing is not supporting, there will be a delay for C*Net, right? Since the pulses must first be converted to tones, and then back to pulses in SxS systems, instead of the SxS relays moving as the dial returns, it will return afterwards and you will hear clicks again in the receiver...
Quote from: bellsystem on July 20, 2017, 01:49:55 PM
That did not help clarify anything. It seems they are both hardware that turns/converts an analog device into a digital one. What are the differences between them, especially with regards to how they would be used with C*Net?
Point was, do some research on these before asking a question. Expecting members to endlessly reply to an issue more than a stand alone unit or a card for a PC is asking members to teach you the basics of the units rather than you doing it yourself. If you run into a highly defined technical question when doing your own research, I'm sure many members will be pleased to answer that specific question instead of winding up in a bloviated revolving thread that goes no where and does not answer your question, or answer it to your ideological supposition.
This is a help form populated by experts and collectors each of which has their own field of knowledge based on real world experience and training and not a school designed to teach someone who has an idea of and/or knowledge of two or more current and/or obsolete devices how to how they would work or should work or in some unrealistic hypothetical theory, much less how to do the work of interconnecting them which seems would never happen even if given that advice on exactly how to do it, if it could be done.
Finally, there is a whole board dedicated to this topic's subject
http://www.classicrotaryphones.com/forum/index.php?board=57.0
Have you read through all of the topics in their entirety that might have already answered your question? Since you don't yet have the Raspberry Pi as of this date, first are you going to get it and if so, maybe that would be a good time to engage this topic once again, after you have it in your hand and you are actually hooking it up, trying to get it working and save members from wasting time replying to a "maybe" hypothetical situation that may never come to be.
Quote from: bellsystem on July 20, 2017, 03:23:13 PM
I don't have an ATA - not sure how I managed to communicate that...
Anyways, that explanation helps.
So, my understanding is:
If you ONLY have an ATA, someone from C*Net has to host you.
If I have a Raspberry Pi, can I host the ATA using that instead?
Also, which option will not requiring doing anything with routers and opening ports? I won't be able to do that.
Also, if pulse dialing is not supporting, there will be a delay for C*Net, right? Since the pulses must first be converted to tones, and then back to pulses in SxS systems, instead of the SxS relays moving as the dial returns, it will return afterwards and you will hear clicks again in the receiver...
In an earlier message, which perhaps you retracted, after first saying you wanted to set up * on your R-Pi you later said (paraphrasing) "oh, I have an ATA. I guess I could get someone to host me and don't need to set up *".
I'm not going to spend time looking for that statement. I'm quite certain it is not a figment of my imagination.
compubit and/or others have already explained what you need to connect to your own * server to provide voice communication terminals and the other questions you are repeating. We are not just going in circles here but in a descending spiral again.
I think BellSystem has been a bit confused by all the posts on here, so I'll try to clarify it a bit if I can.
You are quite right, you DON'T need the raspberry pi to connect to C*net. All you need is an ATA, which stands for Analogue Terminal Adapter. This does what it says on the tin and converts the analogue signal coming from your phone or PBX into a digital signal your router and the internet can understand.
Now where a traditional phone needs to be connected to a telephone exchange (or 'central office' for you guys in the US), your phone and ATA setup also needs something to connect to which will route the calls in a similar way. The thing you connect to is a C*net server, which you can think of as the VOIP version of a central office. A network of these servers are set up all over the world by C*net users. So your phones can connect (via your ATA and router) either to a remote C*net server run by someone else or to one run by yourself on your Raspberry Pi.
The problem with having your own C*net server on your pi is that you will almost certainly have to forward some ports on your router. You will also have to set up the C*net server, which is not easy, especially for someone new to this sort of thing.
The best way to go is get connected to C*net with an ATA (I recommend a Grandstream 502 which can understand pulse or tone) and point it to a server / exchange / central office run by someone else. You can have two lines on a Grandstream, each with a different C*net number and you don't have to do any port forwarding on your router. This would at least get you connected. Then if you really want your own C*net server, because you are already connected via a host, you will have the ability to call people to ask for advice. This is exactly what I did when I was getting started. I can tell you now I would never have figured out the complexities of setting up a server if it hadn't been for many calls to a couple of people here in the UK who actually talked me through the setup as I did it.
If you need someone to host you that's no problem. I could easily give you a couple of numbers plus a copy of all the settings you would need to put into your ATA.
As I said, the Grandstream 502 accepts either pulse or tone dialling. But if your PBX can do the conversion and put out tone down the line to the ATA then you might consider a Linksys PAP2t ATA which only understands tones but is more common and hence quite a bit cheaper. The thing you have to watch for is that you don't get one that's been locked to a network. Also make sure it's got the power brick with it.
If you need any more info about getting going on C*net feel free to PM me any time. Of if you get a C*net phone working then call me on the number below.
Andy.
Quote from: andy1702 on July 21, 2017, 03:41:34 PM
The problem with having your own C*net server on your pi is that you will almost certainly have to forward some ports on your router.
That's true and you mentioned earlier that you did not have access to the router to make changes. As Andy suggests, perhaps it is better to find someone (like Andy) to host you. That will provide you with a small success and perhaps encourage you to go further.
Quote
You will also have to set up the C*net server, which is not easy, especially for someone new to this sort of thing.
The type of server used significantly affects the amount of work involved and the steepness of the learning curve. This discussion can also be like discussing the merits of Ford verses GM. My opinion is that using RasPBX for the Raspberry Pi is far easier than using plain asterisk.
Quote
The best way to go is get connected to C*net with an ATA (I recommend a Grandstream 502 which can understand pulse or tone) and point it to a server / exchange / central office run by someone else. You can have two lines on a Grandstream, each with a different C*net number and you don't have to do any port forwarding on your router.
Again, I agree, but be aware that, as I mentioned earlier, the last batch of 502's I bought did NOT respond to pulse dialling. I have not followed it up.
However, instead of connecting a telephone to your ATA, you can connect one (or two) of the exchange (CO) lines of your Panasonic PABX. That will give you the benefits of the PABX with the ability to make CNet calls from any of the extensions. Remember that the PABX will convert pulse dialling at an extension to DTMF on the exchange line so the ATA does not need to understand pulse dialling.
Good luck
Jack
If I decide to hook up to C*Net, I will likely use an ATA and find someone to host me.
That being said, I'd still like to do something with the Raspberry Pi, even if it's not related to C*Net. Maybe somehow hook it up to the PBX to add some functionality, even if not entirely useful? Preferably, I'd like it work without Internet access so I don't have to run an Ethernet cable around the house.
Does anyone have any similar, cheap ideas as to what to do with the Raspberry Pi? In particular, I was hoping to make a functional pulse to tone converter or maybe a Blue Box - or maybe both in the same Raspberry Pi. I don't want to modify any existing phones - for the P2T converter, I was thinking of having two RJ11 jacks on the Pi so that pulses pass in through one jack and tones out through the other. It seems feasible I could run a blue box and other boxes through the same Pi since it would be connected to the line.
Or maybe a DTMF keypad instead of a converter would be easier.
Is Asterisk useful without Internet connectivity, if I connected it to my PBX? I do have an unused router I can hook it up to if necessary, but it would not be connected to a modem (for Internet).
Quote from: bellsystem on July 23, 2017, 12:45:13 PM
That being said, I'd still like to do something with the Raspberry Pi, even if it's not related to C*Net. Maybe somehow hook it up to the PBX to add some functionality, even if not entirely useful? Preferably, I'd like it work without Internet access so I don't have to run an Ethernet cable around the house.
Does anyone have any similar, cheap ideas as to what to do with the Raspberry Pi? . . . .
Have you checked out the ultimate site for R-Pi users to get ideas? And, in reality, cheap is really in the wallet of the spender.
https://www.raspberrypi.org/
or this link saturated site for ideas, help, products, etc.?
http://elinux.org/RPi_Hub
Yeah, there's not many telephony related items on those sites there,
I don't think I'd do a P2T converter. Too complicated after some research.
Maybe a DTMF keypad from a Pi (shouldn't be hard, right?) and/or a blue box.
Actually, what might even be cooler is one keypad with a toggle that emits DTMF and when the toggle is switched the other way emits the Blue Box tones and 2600 Hz. Could I feasibly use one touch-tone keypad, and allow it to chirp out DTMF and blue box frequencies by using a Pi?
Quote from: bellsystem on July 23, 2017, 01:58:45 PM
I don't think I'd do a P2T converter. Too complicated after some research.
Maybe a DTMF keypad from a Pi (shouldn't be hard, right?) and/or a blue box.
Actually, what might even be cooler is one keypad with a toggle that emits DTMF and when the toggle is switched the other way emits the Blue Box tones and 2600 Hz. Could I feasibly use one touch-tone keypad, and allow it to chirp out DTMF and blue box frequencies by using a Pi?
Sounds like great ideas never done before that you can research, diagram, get the parts, hook it up and report back on your success including how to do it in case someone else may want to duplicate you creation.
Quote from: Jack Ryan on July 21, 2017, 09:33:25 PM
I agree, but be aware that, as I mentioned earlier, the last batch of 502's I bought did NOT respond to pulse dialling. I have not followed it up.
Jack
That's interesting... I must have missed that post somewhere along the line. I wonder if it they are not responding because of an update? One thing I was advised when I got mine was to go into the settings and turn off the option to update the firmware.
I'll also say that when connecting a dial phone to a Grandstream 502 the ATA is very picky about the make/break ratio of the dial contacts. I think they are set up for US operation but here in the UK our make/break ratio is slightly different, which meant about 1 in 10 pold phones worked as bought. The solution (on GPO dial 21s at least) is simply to very carefully bend the contacts so they don't open quite as far. This changes the make/break ratio enough to make UK phones work on the 502 without any problems.
As Jack says, I think the best option is to connect your phones to the PBX and then connect the PBX to the ATA so the PBX can handle the pulse to tone. That's the way I do it.
Andy.
Quote from: bellsystem on July 23, 2017, 01:58:45 PM
Yeah, there's not many telephony related items on those sites there,
I don't think I'd do a P2T converter. Too complicated after some research.
Maybe a DTMF keypad from a Pi (shouldn't be hard, right?) and/or a blue box.
Actually, what might even be cooler is one keypad with a toggle that emits DTMF and when the toggle is switched the other way emits the Blue Box tones and 2600 Hz. Could I feasibly use one touch-tone keypad, and allow it to chirp out DTMF and blue box frequencies by using a Pi?
Quote from: TelePlay on July 23, 2017, 02:11:08 PM
Sounds like great ideas never done before that you can research, diagram, get the parts, hook it up and report back on your success including how to do it in case someone else may want to duplicate you creation.
You don't need an RPI for something this simple. A $2 Arduino board running without an operating system is plenty. There are several recipes on-line.
With a 2 port ATA you can have a single google voice account make or receive 2 simultaneous calls , or even 4 with a 4 port ATA.
Mark
Actually, you can ditch the Panasonics completely. In addition, if you configure your FreePBX dial plan correctly, you can also eliminate dialing 9 for a trunk, enable 7 digit dialing even when the phone company disallows it, etc. Also, you can keep the POTS lines.
What you need:
- Host computer. This can be the Raspberry Pi, there is a Raspberry Pi asterisk distro with FreePBX.
8 Port Ethernet Switch (unless you go with higher end hardware, you are going to need 4 ATAs (one for each two telephones, I suggest also considering having some POE ports if you wish to use Digital IP phones).
FXO Gateway (Converts your POTS line to a VoIP line for the Pi to access, if you port your Number to a BYOD provider such as CallCentric you can eliminate this device)).
4 Grandstream HT802 ATAs with 1.0.5.11 or newer (newest as of this post FYI) firmware.
One or More SIP softphones on computers/tablets/smartphones (for testing prior to connecting ATAs).
This system will allow you to connect Rotary Dial phones direct to VoIP. Remember the firmware version and to set "Enable Pulse Dial" and "Enable High Ring Power" settings for the ATAs. You can also use this system without dialing a trunk code (such as 9) as Asterisk/FreePBX can interpret the dialed digits and route the call, however this requires some creativity* (or dialing 9 for CNET and no Prefix for PSTN) to allow international calls to CNET.
* IE using 00 for International PSTN calls (PBX can replace 00 with 011 before sending out)