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More lines into Norstar system

trebble

Technical User
Jan 12, 2025
17
Hello everyone


I have a norstar modular ics system with a call pilot 100

Is there a tried and true method for getting more lines into these things without bringing more physical lines onto my site.
I believe they do not support VOiP so I'm wondering if there is a different approach.


I already added something called smarthubs which work of a cell signal but they don't work very well for me.
If someone thinks they make the hubs work better for me please look at my other thread called Norstar help.

Any help is appreciated.
 
What city & state are you in. I might be able to help if you are in socal ?
Located in Canada, best country there is.

Is it possible to do this without physical wires?
Internet or cell signal or some other means?
 
VoIP line with ata to transfer to analog lines can be an option
 
VoIP line with ata to transfer to analog lines can be an option
I did google ata and I see now what it is.
Does this work with a norstar system?
Has anyone used it before?
 
I use a grandstream ht812 with my Norstar. I get voip sip trunks into my old norstar.
 
I use a grandstream ht812 with my Norstar. I get voip sip trunks into my old norstar.
Thank's for the reply

I already use something that work of a cell signal, convert's it to analog than I come of this device with a patch cord and punch onto my system incoming line port's.
It's called a smarthub. ZTE MF279T.
But I'm having issues when someone phones in on these line's and dial an extension, it say's(not always) you have dialed an invalid extension, or that command is not recognized.
One of the guy's on the forum think's my DTMF of the smarthub isn't good enough.
Are you having any issues as far as that is concerned
 
Why is the smart hub on "Cellular"?
What type/service/equipment do you have for Internet? is that cell too?
Are you in some remote location?

Norstar/BCM have there own version of ATA for analog devices to work on the digital ports.
Carrier ATA's (Gateways') work with SIP such as jsaad posted.
 
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Why is the smart hub on "Cellular"?
What type/service/equipment do you have for Internet? is that cell too?
Are you in some remote location?

Norstar/BCM have there own version of ATA for analog devices to work on the digital ports.
Carrier ATA's (Gateways') work with SIP such as jsaad posted.

What NT numbers are on the line cards?
Maybe swapping for older or newer might help, or changing the CO settings in programming.
The smart hub is to provide internet and phone service of a cellular signal but I'm only using it for phone service to provide more lines into my phone system.
Internet on site is provided by starlink.
Yes, located in semi remote area in Sask.
Where would I find the NT number's and which CO setting's would I be changing.
The only progamming I've done is via the terminal's on a phone not even sure you can hook a laptop up these old system's.
Thanks
 
I will respond to that in your other thread:
 
Last edited:
I sort of have the same setup as jsaad...

I keep a Grandstream HT818 in my truck in case a client loses all or most of there lines, or to test the KSU side.
I also have it registered to voip.ms SIP Trunks/DID

One example:
Backhoe nearby cuts major telephone feed cable but still has Internet via other means.
I would install the HT818 in place of the current POT's lines going into the KSU, then forward their main line (be it me or the carrier) to one of my DID's tied to the HT818.
Once the carrier has fixed issues then I reverse all.

One note on this though, it will not apply to you but Rogers was installing these in place of their standalone modems (even on Bell's internet service, LOL), however Callpilot's Outbound Transfer from a mailbox does not conference the call, but works fine with SIP trunks such as voip.ms.
Rogers (Business) has since given up on supporting analog in any way.

You see how much more powerful the programming is on this admin guide.
 
I sort of have the same setup as jsaad...

I keep a Grandstream HT818 in my truck in case a client loses all or most of there lines, or to test the KSU side.
I also have it registered to voip.ms SIP Trunks/DID

One example:
Backhoe nearby cuts major telephone feed cable but still has Internet via other means.
I would install the HT818 in place of the current POT's lines going into the KSU, then forward their main line (be it me or the carrier) to one of my DID's tied to the HT818.
Once the carrier has fixed issues then I reverse all.

One note on this though, it will not apply to you but Rogers was installing these in place of their standalone modems (even on Bell's internet service, LOL), however Callpilot's Outbound Transfer from a mailbox does not conference the call, but works fine with SIP trunks such as voip.ms.
Rogers (Business) has since given up on supporting analog in any way.

You see how much more powerful the programming is on this admin guide.


In your personal experience does it have any issues connecting to the Nortel system like I am experiencing with the smart hub's.

I have never dealt with VoIP so a couple question's

Would Sasktel still be my service provider?

Does each VoIP line has a monthly fee or is it covered with my internet subscription?

It seem's I could pick one up on amazon for 200 buck's so I'm wondering would I need any more equipment or do I just bring internet to them and
connect the phone lines to my line card's ,do a little programming and I'm good to go?
 
I have never had one running at a site for too long so I cannot speak for issues.

"Would Sasktel still be my service provider?
Does each VoIP line has a monthly fee or is it covered with my internet subscription?"


Voip.MS is a SIP Trunk carrier, they do not provide Internet.
You would normally use just one carrier but sometimes another is used depending on the situation.

If after testing with the SIP Trunk Carrier & Gateway method via SaskTels internet, then you could transfer all your lines over from Sakstel to the SIP carrier.
However I would keep the analogs you have and just use the new ones for dialing out as a house rule, prime the sets to Pool B.

The ZTE MF279T smarthub seems to be a unit for home and not much for programming, not sure why they would give you that for a business.

I think to sign up for Voip.MS is free but you need to pay for a DID, and can have several free sub accounts.
Once you have a DID then you program it at Voip.MS admin page, and on the Grandstream gateway.
BTW the HT818 has a v2 which has a nicer GUI than v1, maybe more features?

This situation is not easily rectified because you have new and old technology so you need to try and see if SaskTel can go in the backend and change anything or contact ZTE yourself.

Have you tested with a Butt Set between the Hub and the MICS to see if if you can here tones when buttons pressed in or out?

About the cards, another user fixed a similar DTMF issue by going with the much older LSDS (DS) card which is life before Caller ID.




If you have exhausted trying all other ways and want to be a Guinea pig and try the Grandstream/SIP Trunk method:

-Programming at the SIP carrier
-Programming at the Gateway
-Connect the telephone line ports to the KSU and program the lines to set/s
-Connect the Grandstream ethernet(WAN?) port to the network switch, or available port on the back of the Smarthub.


From your other thread:
"I have 8 lines coming into this system now.
3 are wired
5 are with smart hubs"

You could forward your line 3 (wired) to the Voip.MS DID.
Now make some test calls and if it works.

Note that I think you get 2 channels (calls) per DID but hopefully others can confirm, this means program/wire up 2 ports on the Grandstream to the KSU.

Although it may take a few days for somebody to respond.

I have never had one running at a site for too long so I cannot speak for issues.

"Would Sasktel still be my service provider?
Does each VoIP line has a monthly fee or is it covered with my internet subscription?"


Voip.MS is a SIP Trunk carrier, they do not provide Internet.
You would normally use just one carrier for lines but sometimes another is used.

I think to sign up for Voip.MS it's free but you need to pay for a DID, and can have several free sub accounts.
Once you have a DID then you program at Voip.MS and on the Grandstream gateway.

This situation is not easily rectified because you have new and old technology so you need to try and see if SaskTel can go in the backend and change anything or contact ZTE yourself.


If you want to be a Guinea pig and try the Grandstream/SIP Trunk method:

-Programming at the SIP carrier
-Programming at the Gateway
-Connect the telephone line ports to the KSU
-Connect the Grandstream ethernet(WAN?) port to the network switch, or available port on the back of the Smarthub.


From your other thread:
"I have 8 lines coming into this system now.
3 are wired
5 are with smart hubs"

You could forward your line 3 (wired) to the Voip.MS DID.
Now make some test calls and if it works.

Note that I think you get 2 channels (calls) per DID but hopefully others can confirm, this means program/wire up 2 ports on the Grandstream to the KSU.

A small business with basic needs does not even need a phone system with these SIP trunk carriers.
Once you see the programming and all its features you will want to play.

Outta Ammo!
 
I have never had one running at a site for too long so I cannot speak for issues.

"Would Sasktel still be my service provider?
Does each VoIP line has a monthly fee or is it covered with my internet subscription?"


Voip.MS is a SIP Trunk carrier, they do not provide Internet.
You would normally use just one carrier but sometimes another is used depending on the situation.

If after testing with the SIP Trunk Carrier & Gateway method via SaskTels internet, then you could transfer all your lines over from Sakstel to the SIP carrier.
However I would keep the analogs you have and just use the new ones for dialing out as a house rule, prime the sets to Pool B.

The ZTE MF279T smarthub seems to be a unit for home and not much for programming, not sure why they would give you that for a business.

I think to sign up for Voip.MS is free but you need to pay for a DID, and can have several free sub accounts.
Once you have a DID then you program it at Voip.MS admin page, and on the Grandstream gateway.
BTW the HT818 has a v2 which has a nicer GUI than v1, maybe more features?

This situation is not easily rectified because you have new and old technology so you need to try and see if SaskTel can go in the backend and change anything or contact ZTE yourself.

Have you tested with a Butt Set between the Hub and the MICS to see if if you can here tones when buttons pressed in or out?

About the cards, another user fixed a similar DTMF issue by going with the much older LSDS (DS) card which is life before Caller ID.




If you have exhausted trying all other ways and want to be a Guinea pig and try the Grandstream/SIP Trunk method:

-Programming at the SIP carrier
-Programming at the Gateway
-Connect the telephone line ports to the KSU and program the lines to set/s
-Connect the Grandstream ethernet(WAN?) port to the network switch, or available port on the back of the Smarthub.


From your other thread:
"I have 8 lines coming into this system now.
3 are wired
5 are with smart hubs"

You could forward your line 3 (wired) to the Voip.MS DID.
Now make some test calls and if it works.

Note that I think you get 2 channels (calls) per DID but hopefully others can confirm, this means program/wire up 2 ports on the Grandstream to the KSU.

Although it may take a few days for somebody to respond.

I have never had one running at a site for too long so I cannot speak for issues.

"Would Sasktel still be my service provider?
Does each VoIP line has a monthly fee or is it covered with my internet subscription?"


Voip.MS is a SIP Trunk carrier, they do not provide Internet.
You would normally use just one carrier for lines but sometimes another is used.

I think to sign up for Voip.MS it's free but you need to pay for a DID, and can have several free sub accounts.
Once you have a DID then you program at Voip.MS and on the Grandstream gateway.

This situation is not easily rectified because you have new and old technology so you need to try and see if SaskTel can go in the backend and change anything or contact ZTE yourself.


If you want to be a Guinea pig and try the Grandstream/SIP Trunk method:

-Programming at the SIP carrier
-Programming at the Gateway
-Connect the telephone line ports to the KSU
-Connect the Grandstream ethernet(WAN?) port to the network switch, or available port on the back of the Smarthub.


From your other thread:
"I have 8 lines coming into this system now.
3 are wired
5 are with smart hubs"

You could forward your line 3 (wired) to the Voip.MS DID.
Now make some test calls and if it works.

Note that I think you get 2 channels (calls) per DID but hopefully others can confirm, this means program/wire up 2 ports on the Grandstream to the KSU.

A small business with basic needs does not even need a phone system with these SIP trunk carriers.
Once you see the programming and all its features you will want to play.

Outta Ammo!
I'm not quite sure yet what I'll do about it but I will try to update what worked for me.

Your help was appreciated.
 

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