How do I make a single ringer ring from 2 different lines?

[markosjal]

I figure there is probably a more simple answer for this as it seems to be done in !ultimate line phone systems. Of course since I am using asterisk I can set up an extension for the bell alone and make it ring whenever a line to have but that ties up another ATA.

I know long ago I had 2 pots lines and bought a radio shack auto switch device. With that device, a single line phone (or bell) connected to it and when either line rang it rang the phone ( or bell) . You would always get line 1 for dialing out unless you sent a series of dtmf tones which would then switch it to line 2. I used this for a metered kids line so they would always dial out on flat rate line . It did not pass caller id but worked otherwise.

[Alex G. Bell]

The simplest way is to use an AC operated relay (with capacitor in series) to transfer the ringer to the 2nd line when it rings, leaving it on the 1st line to ring for that line when the 2nd line is not ringing.  A 48VDC high resistance relay with DPDT contacts, a diode bridge and a 0.5uF 200V polyester film capacitor will accomplish this. 

This is similar to the circuit a number of people have published for allowing the ringer in a magneto phone to operate both when the magneto is operated and for incoming calls. 

Of course if calls come in simultaneously on both lines and they don't ring in synchronism an odd ringing cadence will be produced by the ringer.

Someone made up a small ckt board containing a DIP-16 relay and these other parts which also implemented this circuit.  It was for transferring the T&R of an ordinary single line 2500DM set to the ringing pair on 1st Generation Horizon hybrid PBX systems, which assumed the use of 2500MM sets with ringing on a 2nd pair of the mounting cord.  These board occasionally show up on eBay.  I don't know the mfr's name and IIRC, there is no part # marked.

[markosjal]

I need to worry about loading down the ring voltage on the line though ? I have fried ATAs by loading down the ringing too much. My precious rotary compatible ATAs I need to take into account.

Would it be better to use say a bridge rectifier in series with .47uf cap > electrolytic filter cap> resistive voltage divider > 5V Buck converter > 5V relay

Buck converters are great and efficient and there are even solid state relays at 5V . They are also available most anywhere as most of the cell phone car charging USB port adapters now use buck converters. Get em at autozone or walmart.

a 48V relay would probably be a large beast anyway

[Alex G. Bell]

I need to worry about loading down the ring voltage on the line though ? I have fried ATAs by loading down the ringing too much. My precious rotary compatible ATAs I need to take into account.

Would it be better to use say a bridge rectifier in series with .47uf cap > electrolytic filter cap> resistive voltage divider > 5V Buck converter > 5V relay

Buck converters are great and efficient and there are even solid state relays at 5V . They are also available most anywhere as most of the cell phone car charging USB port adapters now use buck converters. Get em at autozone or walmart.

a 48V relay would probably be a large beast anyway

I'm going to take your first sentence as a statement even though it ends in a question mark given what you said afterwards.  I'm surprised that any ATA would fail for that reason.  If that's really the reason the ATA's are very poorly designed.  It's not difficult to design a self-protected ring generator circuit and someone connecting an excess number of ringers is, frankly, predictable.  What ATA were they?

Using a voltage converter inevitably wastes power, increasing the input power requirement to operate a relay of given power requirement.  No matter how efficient they are, they can't possibly be 100% efficient.  Moreover, the lower the relay voltage the more power is wasted in the 1.2-1.4V drop of the rectifier bridge (whether built into the converter or external), which for a 5V relay is ~25% of the output power (1.2-1.4V/5V).  So a 48V relay is much more efficient and sensitive.  A 110V relay may not operate if ringing voltage is not especially high but is still within specification.

There is also a requirement that the voice frequency impedance be suitably high when ringing is not present so it does not cause voice signal loss.  A FWB with capacitor in series with the input, feeding a filter capacitor at it's output, barely meets that requirement, with threshold voltage about 1.2-1.4V for the two diodes.  This may result in clipping the peaks of DTMF dial signals, preventing proper dial operation.  It's prudent to add a 2.4-3.9V zener in series with the output of the bridge to raise the threshold to 3-6-6.1V.

DIP-16 form factor relays are readily available at 48V.  The Clare LM14F00 is an example.  Everyone cloned this relay: Omron, American Zettler, lots of others.  I don't know of any DPDT solid state relays, which is what you need.
http://www.ebay.com/itm/151929288279

Meanwhile, I discovered one of the relay boards I mentioned hiding under a pile on my workbench, but Google knows nothing of them and the phone # appears to have been reassigned.  The label on the bag reads:

D&C Electronic Engineering
(303) 296-5550
Part # BC-001
Com Code 405-148-743

So from the fact that there is a ComCode, apparently they were selling them to Bell operating companies.

That said, this circuit uses a single HW rectifier and 1uF 63V polarized electrolytic cap.  But it's not intended to operate across T&R, rather across the dry ringing pair of a Horizon system so this is sufficient for the purpose.

Thanks for bringing this topic up.  I have a number of 2511, 2515 and similar 2-line sets.  This is a good solution for getting them ring for both lines.  Had not thought before to do this.

[markosjal]

I am actually thinking about buing some old AE 3 line sets , so depending on how I go with this it could be useful info

[Dominic_ContempraPhones]

That's the problem -- you need a proprietary set where the ring signal is a supervisory command to the phone, and not a voltage.  With voip, use sip phones.

[Alex G. Bell]

That's the problem -- you need a proprietary set where the ring signal is a supervisory command to the phone, and not a voltage.  With voip, use sip phones.

Do you have a solution for using am existing telephone set?  That's what he asked for, not an alternative technology. 

CRPF is mostly inhabited by people who want to make existing phones work rather than replacing them with something entirely different.

[Alex G. Bell]

I am actually thinking about buing some old AE 3 line sets , so depending on how I go with this it could be useful info

There's no limit to the number of lines for which you can extend this idea by using N-1 relays where N is the number of lines.  So 2 relays for 3 lines.  DIP relays make this a practical solution.

[Dominic_ContempraPhones]

The solution is to install an auxiliary external ringer, and then enable it for both phones.  We could do that.  Those phones had a lamp indication though so you knew which one was ringing.  It's done all the time.  I'm sure Asterisk can do it.  We did it when we needed a LOUD ringer, or an overhead light to go on and off.

[Dominic_ContempraPhones]

Do you have a solution for using am existing telephone set?  That's what he asked for, not an alternative technology. 

CRPF is mostly inhabited by people who want to make existing phones work rather than replacing them with something entirely different.

He is using Asterisk, a VOIP switch, not legacy technology.  When in Rome, you do as the Romans do.  I'm a purist.  SIP switches were meant to be used with SIP phones.  Do you see business converting to VoIP and using 302s?

That's like me going to  Cisco and saying how do I make a Centrex p-phone work on their platform.  We have an aux. port for a ringer, so we're hybrid, but he's soft.  You're telling him to use relays?  There's  probably a Digium card for that.

I'm sure Asterisk can do what Nortel BCM can do in this case.  It's such hypocrisy to use VoIP with analog endpoints, and then criticize TDM and whatever.

[Alex G. Bell]

He is using Asterisk, a VOIP switch, not legacy technology.  When in Rome, you do as the Romans do. 

He closed his question with:
"I know long ago I had 2 pots lines and bought a radio shack auto switch device. With that device, a single line phone (or bell) connected to it and when either line rang it rang the phone ( or bell) . You would always get line 1 for dialing out unless you sent a series of dtmf tones which would then switch it to line 2. I used this for a metered kids line so they would always dial out on flat rate line . It did not pass caller id but worked otherwise."

It's clear that he established a context around an adjunct device switching T&R which would work in any analog environment including lines not switched by his *.  Accordingly that is the suggestion I made.

In reply he said: "I am actually thinking about buying some old AE 3 line sets , so depending on how I go with this it could be useful info:

So it appears that he is not primarily looking for an * based solution.

[Dominic_ContempraPhones]

Then why is he using Asterisk?  For VoIP.  Why are you guys using VoIP.  It sucks.

He's talking about a fax switch with distinctive ringing.  Asterisk can do distinctive, at least I don't think so.

My friggin Meridian 9516 CW can do that, but DMS provides the double ring.  What's the reasoning for the one bell?

[Alex G. Bell]

Then why is he using Asterisk?  For VoIP.  Why are you guys using VoIP.  It sucks.

He's talking about a fax switch with distinctive ringing.  Asterisk can do distinctive, at least I don't think so.

My friggin Meridian 9516 CW can do that, but DMS provides the double ring.  What's the reasoning for the one bell?

When I search this thread (all replies of which appear on my screen on a single page due to my profile settings), the only instances of "FAX" or "distinctive" are YOUR mention of them.  So if you think he's talking about a FAX switch you're misunderstanding what he's talking about and reading something into it he did not say (or mean).  You're reading in something which is not there even by implication and not correct in terms of what he meant.

Radio Shack made an adjunct device for sharing an answering machine between 2 lines.  It had "piano keys" for selecting which line was used for OG calls.  It automatically connected an incoming call to the TAD IF the TAD was not busy on the other line.  It had high input resistance voltage detectors monitoring both lines and logic to make a decision whether to switch the TAD to the other line.

Later they made an ultra simple and very clever device with a single magnetically latching DIP relay which was caused to operate if one line rang and release if the other rang, leaving the output jack connected to whichever line rang last.  This was for the same purpose but had no controls but was quite an ingenious embodiment of a very simple concept. 

Then they also made a device which allowed either line to be selected by DTMF digits sent from the output port.  This is the device Marko described.

A FAX switch shares a single line among multiple devices, not multiple lines to a single FAX.

As a matter of fact "in the day" I shared a 14.4kbps data/FAX modem in my PC between a direct CO line and a PBX EXT using the first "piano key" R-S device so that incoming FAX calls on the PSTN line could reach the modem to deliver FAXs and OG data or FAX calls could use the PBX to select one of a number of OG PSTN lines to take advantage of different calling plans on the trunks available on the PBX.

The device Marko described would serve his purposes, in fact any of the three would, though the more complex ones are overly featured for just sharing a ringer, and all are limited to 2 lines whereas at last word he was talking about 3 lines, and it is probably near impossible to find any of them with any predictability even on eBay.

[unbeldi]

Then why is he using Asterisk?  For VoIP.  Why are you guys using VoIP.  It sucks.

That is your opinion.  And it probably is not based on much experience, since you decry it at every opportunity.

The fact is that a good portion of telecommunications traffic is already transported via VoIP, unbeknownst to many, and undetectable to all probably, and you yourself stated that the core of the US network today is 100% packet (frame) based.

I have designed and built communication systems using Verizon Wholesale Business's SIP product (Nortel CS2000) with call initiation rates exceeding 100,000 calls per hour without dedicated circuits.  Not only was the quality superb, they are reliable, and cheap (~ $0.005/min).  This could not have been achieved using TDM circuits without astronomical costs in both circuits and hardware, and floor space.  Instead it was accomplished with a total of four to six off-the-shelf datacenter Unix servers.

[Payphone installer]

Why use VOIP,because it saves you a fortune. I deliver telephone service for  inmates in county jails in 14 states I used to have to buy on site custom built PBX's and tons of POTS lines.  In the larger ones we used PRI back in 2005 I was convinced to try VOIP which was against everything I believed in at the time. Being a telephone man of 27 years at that time,
I was just resistant to change. I hired a top notch internet and network guy and set down the path of VOIP. It took a long time to get the voice quality to a level I was happy with.
Even to this day I use a hybrid system, it consists of Quintum gateways at the sites talking to Quintum CMS giant gateways at my office. the reason for Quintum is because of a product called packet that compresses the voice packets on the inbound traffic,this traffic is H323.
The traffic is converted back into analog traffic to feed custom racks of custom switches and fed in as T1,  the processed traffic is then outbound on what I call fake PRI's and sent to a digital call router which picks the cheapest path based on area code and exchange. I have piles of DIDs and deliver most of the traffic as local as it should be according to the area. All this is sip traffic. The entire front end traffic is wrapped in tunnels inside firewalls or a WIDE AREA NETWORK (WAN). My calls are perfect and at times cranked up to a level defined in calls per second. This system cost millions to build but saves millions in terminating fees. Every day the cost goes down. Just think what will happen when the calls are  delivered to a cell phone app or soft phone app. At that point there will be no use of the PSTN only data. In the industry everyday there is a race, staying up with it is exhausting,everything is changing at the speed of light if you don't keep up you get run over.