Simple PBX to SIP provider without a trunk? 1

[GriffMG]

Is there a way to do this without a SIP Trunk?

Still thinking about my two IP phone home... wanting to be able to transfer
calls from one to the other... but trying to avoid paying for a trunk...

Regards

Griff
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[Westi]

You could check if your provider allows parking of calls
If yes then you could park a call and pick it up from the other phone again

Joe
FHandw, ACSS, ACIS

 

[GriffMG]

Oh, thank you

Regards

Griff
Keep [Smile]ing

There are 10 kinds of people in the world, those who understand binary and those who don't.

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[gbaughma]

Stupid question here... but if you don't have a trunk, how are you getting calls?

That being said, look at Asterisk. You can run it even on a Raspberry Pi.

Phones would connect to Asterisk... asterisk would connect to... wait.. how are you getting phone calls again?


Just my $.02

"What the captain doesn't realize is that we've secretly replaced his Dilithium Crystals with new Folger's Crystals."

--Greg

 

[GriffMG]

At the moment we have two free accounts with Sipgate, had them for a very long time and they work well.

We used to use a Cisco ATA, but in May/June this year it started to cause our ISP to drop our connection,
never found out why, tried new ATA etc. Stopped doing that and tried Voiper and headsets for a few months
but that was never quite what we wanted.

Bought two aastra 6730i sets and they work very well indeed, but would like to pick up on one
and transfer to the other... hence looking for a way to do that.

Have Raspberry Pis, one running Pi-Hole and NAS...

Regards

Griff
Keep [Smile]ing

There are 10 kinds of people in the world, those who understand binary and those who don't.

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[curlycord]

"Simple PBX to SIP provider without a trunk?"
NO!

SIP Trunk's are what connects everything together.

Sounds like you need to explain your business/setup/flow/tree a little more.

"but if you don't have a trunk, how are you getting calls?" & "wait.. how are you getting phone calls again?"
You did not respond to this, since you don't seem to want to pay for anything it seems confusing as to what type of call your trying to transfer.

You have SIP phones/devices and SIP trunks

SIP Trunks can be these:
-DID (Phone #) or accounts/ext's that is provided by the carrier so that you can receive your business calls and to call each other (charges apply for DID's/ext's)
The carrier acts like a PBX.
-A way to link on premise PBX's together (no charge as the PBX's are programmed to connect to each other over internet without the use of any carrier)

SIP Phones/Devices:
-They register to a PBX/Server as an ext.
-The PBX or Server(Sigate, Voip.MS etc) can be programmed to point the DID's to your phone to receive calls as well call other

So...
-Do you have DID's? or just an account where a phone can register to?
-Can you call the other Sipgate account as I mentioned in your other post?

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[GriffMG]

Hi Curly

I thought I was pretty clear about our set up, two phones, one each side of the kitchen table.

Mine has one line, linked as a sip client to Sipgate on one number, my wife has a similar phone
with two lines configured, one matching mine and the other on a different number altogether.

There are no sip trunks, just connect to the sip server and use an id and password.

Sipgate does do trunks, but not for our profile - free!

Yes line 1 can call line 2...



Regards

Griff
Keep [Smile]ing

There are 10 kinds of people in the world, those who understand binary and those who don't.

I'm trying to cut down on the use of shrieks (exclamation marks), I'm told they are !good for you.

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[Westi]

I have to say free stuff that doesn't do what you want it to do, maybe you should complain to the people that provide you the free service and that you want more free features free of charge or you stop paying them nothing any more.

I know sarcasm comes over badly when written but you get my point. You want stuff for free that has capabilities like other options that cost money due to the fact that they are developed by programmers and techs that feed their families with the money they make.

Joe
FHandw, ACSS, ACIS

 

[GriffMG]

Hmm Westi

I appreciate your comments, but I was just asking 'is it possible?' not asking the supplier to change anything or add features to the free version
or take the food from their children's mouths...

It could well be that one of you might have said 'just use your sip creds for the trunk interface on your FreePBX on your raspberry Pi...'



Regards

Griff
Keep [Smile]ing

There are 10 kinds of people in the world, those who understand binary and those who don't.

I'm trying to cut down on the use of shrieks (exclamation marks), I'm told they are !good for you.

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[gbaughma]

It is possible.
Even with your free extensions you're using.

I could do it with Asterisk running on a Raspberry Pi or a cheap PC.

Here's how it would work:

You program your two phones to connect to Asterisk as SIP extensions. They would have extension numbers, and could call each other without any external connectivity.

You would program your two free accounts also in Asterisk (I did this with Google Voice back in the day). The Asterisk box would "log in" to the two free accounts.

You would program DIDs so that when the external account got a call, it would DIT into an extension. Or, both extensions could ring, whatever you want to do.

When you're on a call, Asterisk is acting as the "Go between", connecting the SIP extension to the external SIP extension (it's not really a trunk, but you're using it as such).

All the features of a PBX would be there... voicemail, transfer, forward, whatever.

Easy peasey.


Just my $.02

"What the captain doesn't realize is that we've secretly replaced his Dilithium Crystals with new Folger's Crystals."

--Greg

 

[GriffMG]

Thank you Greg,

That's the kind of thing I was thinking of!

I have a Pi right beside me, I've even heard of Asterisk... so you are saying it can connect to the SIP account

Any advice on where to look for help setting it up like this?



Regards

Griff
Keep [Smile]ing

There are 10 kinds of people in the world, those who understand binary and those who don't.

I'm trying to cut down on the use of shrieks (exclamation marks), I'm told they are !good for you.

There is no place like G28 X0 Y0 Z0
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[gbaughma]

Yeah... when I first started playing with Asterisk (and before they blocked it) I was able to use my Google Voice account for testing.

There are a lot of "Flavors" of Asterisk, and I'm not sure all of them run on a Pi.
You will probably want to look for something like FreePBX for Raspberry Pi.

Once you get it installed, it's pretty easy to go into the trunk configuration, put in your account and secret and server (just like you would do with your phones) and Asterisk should connect.

Set up your extensions first though... makes it a lot easier for testing.


Just my $.02

"What the captain doesn't realize is that we've secretly replaced his Dilithium Crystals with new Folger's Crystals."

--Greg

 

[gbaughma]

This seems like a good place to start.

http://www.raspbx.org/documentation/

Just my $.02

"What the captain doesn't realize is that we've secretly replaced his Dilithium Crystals with new Folger's Crystals."

--Greg

 

[GriffMG]

Thank you again

Regards

Griff
Keep [Smile]ing

There are 10 kinds of people in the world, those who understand binary and those who don't.

I'm trying to cut down on the use of shrieks (exclamation marks), I'm told they are !good for you.

There is no place like G28 X0 Y0 Z0
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[GriffMG]

** Update **

Followed InnovationAsterisk youtube instructions to get started... not bad, good enough to get the Asterisk pbx up and running
on my old Pi.

Took a bit of study to get the extensions dialing in and out... but for our system, no trunks needed.

One phone working, the other should be a piece of cake.

Thank you gbaughma



Regards

Griff
Keep [Smile]ing

There are 10 kinds of people in the world, those who understand binary and those who don't.

I'm trying to cut down on the use of shrieks (exclamation marks), I'm told they are !good for you.

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[curlycord]

Glad you have it sorted, thanks for updating.


It was not clear to me I think mostly because of terminology.

I am replying/posting this as I could use another cheat sheet.

For example you "wanting to be able to transfer calls from one to the other."

I still do not know how you get calls when you have not mentioned that you have any DID's.
Without a DID's then there is no call to transfer.

How I see it...

Accounts - The main account has the ability to have DID's, sub accounts and features.

SIP Trunk - A tunnel that is automatically created the second a device registers to the account on a server.
SIP is the protocol, Trunk is the path created.
This includes:
Device to PBX or Device to SIP provider
PBX to the SIP provider
PBX to PBX.

DID(DDI in UK) are phone numbers, they just point to accounts(ext's), groups etc.

Lines - are just appearances(buttons) of your registration to the server, they are used for internal and external calls and to receive calls via a DID.
The more appearances on your phone the more calls you can handle.

Channels - is how many simultaneous calls are allowed on the account.


Maybe somebody can confirm, correct or advise me otherwise but that is how I see it.

Cheers

=----(((((((((()----=

http://www.curlycord.com

Toronto, Canada

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[GriffMG]

Hi Curly

In our case external parties cannot dial a specific extension using DDI, the calls go to Sipgate and then ring
both the phones on the table. If my wife picks up and it's for me, she can now forward the call using the button
on the aastra gizmo and the caller is put on hold, with music, until I take the call.

The free Sipgate account allows one call at a time, on one telephone number.

We have two accounts on different numbers, I have no reason to answer the second account - that is for my wife's
other business, so she has two lines on her phone and can answer either.

We do not have a 'proper' sip trunk, with this set up it is not needed.

Really lovin' this arrangement, the Pi is being used for more than Pi-Hole and a NAS, the phones are tidy on the table
and I don't have to use a headset... what's not to love.

Regards

Griff
Keep [Smile]ing

There are 10 kinds of people in the world, those who understand binary and those who don't.

I'm trying to cut down on the use of shrieks (exclamation marks), I'm told they are !good for you.

There is no place like G28 X0 Y0 Z0
​

 

[gbaughma]

Curly:

A lot of that same terminology (or at least the concepts) still apply in Asterisk, which is just an IP based phone PBX.

So, he's using his SIP provider (which he would normally just have a phone connect to on a one-to-one basis) as a Trunk. Think of it as like having a POTS line plugged into an old AT&T Merlin. He then has SIP stations (phones).

So, a SIP TRUNK would be able to have multiple phone numbers on it, pass DNIS digits, all that kind of stuff. So, in this situation, he doesn't exactly have a trunk; it won't do multiple calls, you can't have multiple numbers, stuff like that, although he's using his sip phone provider as a pseudo-trunk to make outbound calls.

He would have one channel for external calls.

Now, at work, I have an asterisk system, about 130 phones (SIP stations), two PRI's (46 trunks) plus a few SIP trunks. Then I have a IAX (Inter-Asterisk Exchange) over our internet provider to another asterisk system, which allows for dynamic connections between the two systems (I only have to do one IAX connection, but it will create as many channels as needed to handle the traffic). This allows us to call station to station between two different offices without using external trunks, or long distance charges.

Additionally, on my system here, I have 8 virtual fax modems, in a fax modem pool. This is really slick... since I have DID numbers pointed to the fax pool, I could even receive multiple faxes on the same number. Asterisk looks at the inbound number, receives the fax, then based on the inbound number routes the fax (converted to a PDF) to the appropriate department. We have about 15 different fax numbers, with 8 modems in a pool. Currently, we receive roughly 2000 pages a week in faxes.

Additionally additionally, everyone has a fax client on their PC. Using a virtual printer... so they just choose the fax as their print destination, a little popup comes up asking the number that they wish to fax to, the fax is queued up and sent out. As simple as printing, the users can fax.

We have call queues, pickup groups, group ringing, status (not only DND but Lunch, In A Meeting, etc.) all visible to the two touch-screen operator stations at the front desk.

Lots of stuff you can do with Asterisk... and if you couldn't tell, I'm pretty proud of the versatility as well as the stability of the system here.


Just my $.02

"What the captain doesn't realize is that we've secretly replaced his Dilithium Crystals with new Folger's Crystals."

--Greg

 

[curlycord]

Thanks to both of you for trying to explain it.

I have setup plenty of SIP trunks and point to point with PBX's.

But I had the same question as "Stupid question here... but if you don't have a trunk, how are you getting calls?"

For me it's not so much about the trunk, it's about GriffMG not having a DID/DDI for somebody to even begin to call them.

I concede on this thread and will move along, and stand in the corner with the dunce cap, it's sorted and that's all that matters.

HNY to all.

=----(((((((((()----=

http://www.curlycord.com

Toronto, Canada

Add me to LinkedIN

 

[GriffMG]

SipGate provide a service with a single, geographically appropriate number, (in the UK at least).

Someone calls that number, any SIP client registered with that number rings. It is DDI to a point I suppose.
Just one number many clients.

Regards

Griff
Keep [Smile]ing

There are 10 kinds of people in the world, those who understand binary and those who don't.

I'm trying to cut down on the use of shrieks (exclamation marks), I'm told they are !good for you.

There is no place like G28 X0 Y0 Z0
​