How to I configure my "Rollover" Line for Sangoma

=== Old phone system ===
I have two phone lines (only 1 jack).
Main Number - 905.639.XXX7
Rollover Line - 905.639.YYY4 (not published)
My old phone system was a "2 line" Panasonic Phone where the one jack went into the "Line1/Line2" jack. When you were on one line, and someone else called it just "rolled over" to the other POTS.
=====================

Are there any good sources to figure out how to configure things ?

Inbound Rollover as you describe, is actually a hunt group with your carrier... this is defined and controlled by your carrier. As such, whatever was in place is still in place, and you do not have the ability to change it (without calling them).

Now... if you have a dual/two-line station jack that you're trying to plug into a Sangoma A200 or whatever, then you will need an adapter to split it for those two ports. As it is now, you only plugged in one line, and the other will be ring-no-answer.

You should be able to get one of those line splitter gizmos at your local Radio Shack. It will have one plug and two jacks. One jack will be line one and the other will be line 2. Plug a Sangoma cord into each one and you are set.

Uh oh !
I was assuming I could get this going !

Re: dual/two-line station jack

I have one of these

... but would that work ?

It doesn't seem too. []

I read that POTS line management has to be done by the Telco.

I am going to call them and ask them to forward calls to a VOIP DID if the Main Line is busy. I would like it to be all POTS, but I dont think that can be arranged easily as I only have 1 phone jack and 1 fax jack. I dont want to mess with the fax right now.

I am moving my office in 10 days as well... so I also need to figure out how I want Bell to do the phone lines for my new office. This is getting stressful !

With this jack, and two cords going to the Sangoma card, the second caller never shows itself in Asterisk. It just rings and rings and goes to my Telco Voice Mailbox.

Hmmmmm.

Help ?

I got this from the Asterisk mailing lists.

=====================
Hi All,

I have a simple question about incoming phone line rollovers. How are
these usually done? Is this done at the phone company usually, or is
this something that Asterisk or channel bank is capable of? I just need
someone to give me a brief explanation how it usually works, and if
someone was implementing an Asterisk system, how they would go about
providing a call rollover (single advertised phone number, but allow
multiple incoming calls)

Thanks in advance,
Leif Madsen.
============

Check the zapata.conf.sample for the keyword 'group'
Jeremy McNamara

============
Q: how does a PBX control the call setup of inbound calls from the PSTN??

unless you are doing something like ATM an your switch is going to
handle processing a call setup request, I don't see how * can
deal with hunting from a PSTN side.

Certainly from the station or SIP or IAX or H323 side it can
deal with it, but I'd be surprised if from the PSTN side.

=====================

I am planning on getting 4 analog "trunk" lines from my carrier (SBC).
~US$14/month/each
And a block of 20 DID numbers for these trunk lines.
(~US$15/month/block of 20) (a block of 20 is the smallest)

Inbound calls come in, and the lines (on the * side) are set to the
same context. (which contain the DID extensions)

If you are not going to use DID, the lines still come into the same
context, just you do not handle the DID extensions. You would just
answer and provide a menu of some sort.

The rollover feature is really just "busy call forwarding".

You can buy 4 residential lines. (do not get call waiting)
Setup "busy call forwarding":
1 -> 2
2 -> 3
3 -> 4

Then advertise the number for line 1.

Outbound calls would be handled by the "group" feature of Zapata.
You put the 4 lines in the same group (in zapata.conf) and the
extension.conf would have Dial(Zap/g<groupnumber>/${EXTEN})

Bell Bill

My Bell Bill.


In the package it says I can only forward the First Number in the Ring Group.

What exactly should I tell them ?

1) stop the ring group
2) setup Call Forward on Busy ... Main Line Busy ... Forward to my Rollover number ?

Q: would this allow the 2 prong thing to work ? Seems like it shouldn't since it's only one phone jack ?

Hmmmm.

This help

Add Zap Channel

Zap Channel DIDs allow you to assign a DID to specific Zap Channels. You can supply the same DID to multiple channels. This would be a common scenario if you have multiple POTS lines that are on a hunt group from your provider. You MUST assign the channel's context to from-zaptel for these settings to have effect. It will be a line that looks like:

context = from-zaptel

in your zapata.conf configuration effecting the specified channel(s). Once you have assigned DIDs you can use standard Inbound Routes with the specified DIDs to route your calls.

(from FreePBX).

http://www.voip-info.org/wiki/index.php?page=Asterisk Zap channels

Detecting Distinctive Ring on Incoming Calls

Your telco may let you use the one analog telephone line for several different phone numbers. Your telco will make your incoming line ring in different ways depending on which number the caller dialed.

My zapata.conf file.

Code:

;autogenerated by /usr/local/sbin/config-zaptel  do not hand edit
;Zaptel Channels Configurations (zapata.conf)
;
;For detailed zapata options, view /etc/asterisk/zapata.conf.orig

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1

immediate=no

;Sangoma A200 [slot:9 bus:0 span:1]  <wanpipe1>
context=from-zaptel
group=0
signalling = fxs_ks
channel => 1

context=from-zaptel
group=0
signalling = fxs_ks
channel => 2

context=from-zaptel
group=0
signalling = fxs_ks
channel => 3

context=from-zaptel
group=0
signalling = fxs_ks
channel => 4  

Hmmmm ....

http://www.trixbox.org/forums/vendor-forums-certified/sangoma/a200-and-rollover

Look for something like this. http://www.radioshack.com/product/i...n=Type/Adapters&allCount=21&parentPage=family

If you shop around, you can probably find it cheaper. It breaks out your line 1 and line 2 single jack in to two separate jacks. Then just plug one line each from line 1 and line 2 in to your Sangoma cards.

I got the requested item, but it didnt seem to help.
The asterisk forums say it won't work unless you work it out with your Telco to make some changes. I am leaning that way.

needed to add telephone numbers as inbound routes and Zap Channel DIDs.

It appears I needed to do two simple things

1) needed to add telephone numbers as inbound routes
2) and Zap Channel DIDs.
____________________________________

1) needed to add telephone numbers as inbound routes



2) and Zap Channel DIDs.



After that ... everything started working properly.


I THINK !

Adding a FXO Trunk using FreePBX

Adding a FXO Trunk using FreePBX


http://www.cadvision.com/blanchas/Asterisk/AddingaFXOTrunk.html

Cool configuration !

On some models of phones the line1/line2 jack was designed two take to phone lines. Line one would be on red and green, line 2 on black and yellow.

You have to go to the wall jack and get an old phone with some alligator clips on the red and green wires. Put the clips on the red and green terminal and see if you get dial tone. Then do the same on the yellow and black.

If you get dial tone on both then your next step is to purchase a two line phone jack box from Home Depot or Lowes. Then unhook the wires from the old box and put red and green to one jack and black and yellow to the other. Then just plug your sangoma cables into each jack.

With Bell's local link lines you have two options. One is to have the standard roll over that is controled by them, the other is to have call forwarding. If I were you I would go with call forwarding. Then you set up call forward on busy from line one to line two. This way if you get set up with a sip line you can call forward line two to your new sip line and no one will know the difference.

We are in the midst of doing this now. We had 6 pots lines at 55 bucks a pop. We are reducing it to 4 all with call forward on busy down to the last line. We then forward line 4 to our sip trunk which will add another 8 lines.

Rob

The device you have pictured appears to be just a splitter. What you need to do is get a "breakout box" or make your own concoction. Someone else posted a picture of a 3way splitter and is typically a breakout device. The one you need will specifically say L1, L2 L1+L2, etc.

What you need to do is breakout L1 & L2. Otherwise, grab yourself a phone cable with four wires (color-coded if possible) in it (some cheap ones have 2) and cut the end off. What you need is red/green going to one plug, and yellow/black going into the other. If the cable doesn't have colors, grab the middle two, and the outer two. You may need to play with it a bit...

The other thing I've done in the past is to make your own breakout box. A little pricier, but more elegant.. Home Depot has the Leviton telecom stuff. Grab an appropriate QuickPort wallplate, as you may be able to stick it in the existing wall, 3(at least) Cat3 (Voice) jacks and cross-connect the incoming jacks to two individual "Line" jacks. You'd go from the incoming blue/white-blue (green/red) to the blue/white-blue Line 1, and from the incoming orage/white-orange to the blue/white-blue of the second jack. You've then created your own breakout box. Plug the incoming line into the in and then you should have two individual lines ready for the Sangoma card.

How did this work out ?
I cancelled my Bell Rollover ... (not sure if I should have)(is going to cost me $50 to try it and $50 to reinstall it if needed).

I will attempt to setup the Call Forwarding as you suggest.

I think Bell finally got around to taking out my Rollover (aka Hunt Group, aka Call forward on Busy), as the second caller doesn't show up on the phones, we hear "the beep" and know someone is calling but we can't answer it. Just started today.

I am going to:
(1) cancel Call Answer
(2) cancel Call Waiting
(3) order Call Forward on Busy for Main Line -> Rollover.

Any other thoughts ? []