Setting up an IVR and voicemail for an existing PBX (non VOIP)

FreePBX Installation / Upgrade
1

I’m quite disappointed that nobody offered any answers. I have made progress through trial and error, but still need help on how to integrate my AsteriskNow/FreePBX with our existing (legacy) centrex pbx system. Thus far, I have been able to make the IVR work such that whenever a call comes through the PSTN interface, the IVR answers.

My question now is: how can I make a call that is being forwarded from the legacy pbx hit the correct extension on FreePBX so that it goes to that person’s voicemail? For example: suppose there is an extension on the legacy pbx 2111. When someone calls the legacy 2111, if no answer, the call is (rolled over) transferred to FreePBX. Now on FreePBX, there is a SIP extension with the same number 2111. How can the transferred call (2111) automatically go to the the SIP extension 2111, so that a voicemail can be left?

I am sure there are experts out there who have the answers. Would please take a moment and provide some guidance? THANKS.
…

Hello Everybody.

I have a very basic need and would like to know whether what I wish to accomplish is at all possible. Obviously I have not found the answer thus far.

There is an existing PBX system that is connected to a very old proprietary voicemail system with auto-attendant (IVR). While the PBX has over 100 extensions, the voicemail system has only 32 POTS lines connected to it, which means that these 32 lines can be dialed to reach the vm system to access each of the PBX extension’s voicemail mailbox.

I would like to replace it with AsteriskNow/FreePBX.

In FreePBX, the choices for EXTENIONS are SIP, IAX2 and ZAP. Since I have analog POTS lines coming into the system, say using Digium TDM400P card, I assume my choice for creating extensions, and thereby mailboxes would be to create ZAP extensions. However, if I choose ZAP, it has to be configured for a specific channel, which I believe is connected to one of these POTS lines. If that is the case, how can I create over 100 extensions for the PBX (extension) mailboxes? Is this doable, meaning can I have say only five physical POTS lines connected to AsteriskNow, that will serve all of my 100+ (PBX extension) mailboxes?

Also, where do I designate one central number/extension, say 2300, that will be the main number that people can dial to access the voicemail system?

Thanks in advance for all your replies/suggestions.

2

Our FreePBX system was intended to be used in homes. When that sale fell through they had me install this to connect our metal shop across the road into our legacy PBX. So I can tell you about the issues, and sympathize.

For each line that connects to your Centrex PBX you can connect to either the IVR or to a specific extension. If you have 100 PSTN lines or lines from your existing PBX to feed into that many ports on your FreePBX system, well this might work:

Use Tools / Config Edit to change zapata.conf and zapata-channels from context=from-pstn to context=from-zaptel

Set up the Zap Channel DIDs to the extension used by outside the PBX.

Then set up your Inbound Routes to that DID Number with the check in CID Priority Route.

You can set the CID name prefix.

Set the Destination - Extension.

The system must be restarted for this to work.

Disconnect supervison must be enabled on the providers switch.

Now in my case, I only had two incoming lines from the old PBX and as soon as I had 3 IP Phones, I needed to change and use the IVR.

I wrote a manual on my experience and posted it here:
http://www.scribd.com/doc/23981850/PBX-Administrators-Guide

3

on your disappointment regarding no answers…

if you read the top of the forum, it is recommended that you provide details about your system…

 A short paragraph about the problem you are having.
 A copy of the output of the Linux status command. This will help those in the know a lot and it will go a long way to narrowing down your particular version. It provides:
PBX in a Flash Version 1.3
Operating system (CentOS release 5.2)
Asterisk Version 1.4.21.2
 FreePBX Core Version (from Module Administration)
 What processor/motherboard/amount of ram:
CPU & Companion Chips VIA C7 1.5GHz + CN700
Memory 1GB DDRII (SO-DIMM)
 Have you run update-scripts, update-fixes, and update-source?
 Relevant logs only! Don’t post full logs into the forum! Post only snippets of long log files. If there is not enough info people will make some suggestions.

but hey, at least you didn’t hijack another post.

4

The BIG question is how you “roll over” a call from the legacy to asterisk? When you roll it over, where does it go?

Bill

5

With Centrex and analog lines into the Asterisk, you generally need to install an SMDI interface (SMDI=Simplified Message Desk Interface) provided by the phone company into your Asterisk. This is used to link Centrex systems into standalone voice mail or other systems. See: http://www.dialogic.com/support/helpweb/mg/tn943.aspx and here is some detail about how to use SMDI with Asterisk: http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.0+smdi.txt

The Centrex will need a hunt group set up. Your no-answer Centrex calls will forward to the Centrex hunt group whose members terminate onto the Asterisk box. The SMDI link from the Centrex is a serial link which must connect to a serial port on the Asterisk. The data stream is usually at 9600 baud or less. The data will tell the Asterisk that on port X, there is a call for extension XXXX and the Asterisk must know how to deal with it. This will likely require some custom programming.

Now if you have the Centrex hunt grup terminate into the Asterisk with a PRI connection, the Centrex (especially one based on Lucent 5ESS) can indicate that the call is forwarded and who it was for. Again, some custom programming would be needed to handle this.

Asterisk has the ability to do all of this but you will likely need to hire someone to program and coordinate it.

If your existing legacy system is not Centrex but a local box of some sort, there are many more variables to address but the general principles still apply.

6

When the two lines from the old PBX were connected to my TDM400P, he told me what numbers they are. It is not critical that the FreePBX assignment is the same.

When a call is answered at the main PBX, we transfer that call to one of those two numbers.

After that, where the call goes is up to your FreePBX.

Does that help?

7

On February 17th, 2010 w5waf (Leap Frog) said:

The BIG question is how you “roll over” a call from the legacy to asterisk? When you roll it over, where does it go?
…

On the centrex, the legacy pbx (300+ extensions), all calls are “rolled over” to 2200, which is the main # in a hunt group. The hunt group has about 32 POTS lines that goes into an antiquated locked-in voicemail system.

Since I need to be able to prove that AsteriskNow is up to the job, I set up a test system using a Sangoma USBfxo, AsteriskNow/FreePBX. They gave me two POTS line that I connected to the USBfxo, 4398 on channel 1,and 2251 on channel 2. On AsteriskNow/FreePBX, I have configured ZAP Channel DIDs: ch1 DID 4398, ch2 DID 2251. For INBOUND ROUTES I have two: 1) a catch all any DID/any CID where the destination is IVR, 2) for SIP extension 6743, with DID/CID as 6743, and that destination is the voicemail for 6743.

On centrex I had them 6743 (the extension on centrex pbx)rolled over to 4398, the POTS line that is connected to USBfxo ch1. Since, I have no 4398 configured on FreePBX as a ZAP extension or a SIP extension, I believe, it goes to the catch all (any/any) and therefore the IVR responds.

My question is, how can I have 6743 answered and connected to its voicemail (mailbox)?

I hope I have been able to explain my set up, adequately. Thanks so much.

8

6743/4398 comes into the asterisk and is able to activate an Asterisk/Freepbx function, (ivr)right?

I think one way to do this would be to set up a custom extension and VM with no dial commands.

You would then hit an IVR. This IVR would say:

“If you Know your Party’s extension Dial it Now”
" To Reach our company directory press 2"

You would end up dialing the extension you have set up, and since there is no dial command it will go directly to the associated Voicemail.

For the employee to get to their voicemail, they would dial a number to hit the above IVR. During the dialing of the IVR they would dial *nnnn where nnnn is their extension number.

I hope I understand what you wanted, if so, this would do it, even though you’d have to set up the 300+ mailboxes on the asterisk system.

Bill

9

On February 18th, 2010 w5waf (Leap Frog) said:

6743/4398 comes into the asterisk and is able to activate an Asterisk/Freepbx function, (ivr)right?

I think one way to do this would be to set up a custom extension and VM with no dial commands.

You would then hit an IVR. This IVR would say:

“If you Know your Party’s extension Dial it Now”
" To Reach our company directory press 2"

You would end up dialing the extension you have set up, and since there is no dial command it will go directly to the associated Voicemail.

For the employee to get to their voicemail, they would dial a number to hit the above IVR. During the dialing of the IVR they would dial *nnnn where nnnn is their extension number.

I hope I understand what you wanted, if so, this would do it, even though you’d have to set up the 300+ mailboxes on the asterisk system.

Bill

Thank you for your idea. I would like to give it a try, however, I am not sure what you meant by “custom extension and VM with no dial commands”? I know custom extension but what would be VM with no dial commands, and how do I do that?

Also, does it mean that when a call from the legacy PBX rolls over it will hit the IVR first and at that time the caller who called the (legacy pbx) extension would have to dial that extension again? Normally, when a call rolls over to voicemail, it hits the correct mailbox (associated with that dialed extension) automatically, and the caller is prompted to leave a message. So, is this a round about way without using SMDI?

Thank you very much, once more.

10

tislam -

As I indicated in your previous post on this subject you are going to need custom extensions and SMDI integration to achieve your goals.

It seems that you consider not receiving the answer you are looking for as not receiving a valid answer. Asking the same question again is not going to produce a different response from the forum.

You have two options. You can embark upon a study of analog telephony and Asterisk configuration and learn how to set this up. With a properly configured custom extension the calls will enter FreePBX properly and you can use the inbound routs and extensions modules to configure your users.

Your system utilizes analog DID,'s they are not FXO’s. They share the same electrical interface however they are not the same type of circuit. Depending on the start of signaling method (wink or loop) dialplan manipulation is required to receive and collect the fed digits that identify the voice mail box.

The SMDI interface is required to light the MWI lamps.

With 300 users in service your ROI justifies the use of a consultant to configure the system for you properly.

There are many talented individuals that code FreePBX, cosulting hours support them so this terrific software can continue to be free. If you don’t have the time or desire to learn how to do this yourself I suggest you avail yourself of these excellent resources.

11

On February 18th, 2010 kenn10 (tadpole) said:

With Centrex and analog lines into the Asterisk, you generally need to install an SMDI interface (SMDI=Simplified Message Desk Interface) provided by the phone company into your Asterisk. …

Our existing VM system does have an SMDI interfaced with our Centrex serially. Apparently we rent (or lease) the device from our Centrex provider.

Now the programming that you speak of, would that be required from the Centrex provider’s side or on Asterisk? I took a look at the link you provided for SMDI configuration and it doesn’t look too bad. I could probably struggle through it however, if there is any programming that needs to be done on the SMDI, then that will most likely incur some cost. Am I correct?

At the risk of sounding like a total idiot, could I not just use the existing SMDI with my Asterisk/FreePBX?

Thanks again.

12

On February 18th, 2010 SkykingOH (Leap Frog) said:

tislam -

As I indicated in your previous post on this subject you are going to need custom extensions and SMDI integration to achieve your goals.

It seems that you consider not receiving the answer you are looking for as not receiving a valid answer. Asking the same question again is not going to produce a different response from the forum.

I’m sorry, I did see your answer but did not really know what SMDI is all about. I needed to read up on it.

While it is absolutely valid that 300+ users could easily justify the ROI for a paid consultant, please believe me, we are literally running on thin air as far as IT is concerned. I cannot remember when was the last time they paid money to hire an outside consultant to do anything. In the 11+ years that I’ve been here, I was never sent for any formal training on anything. Everything I have implemented thus far have been free (open source), save for the hardware.

It may boil down to setting up SMDI, but I’m not sure whether the people with the purse strings will go for it. We may just end up nursing the old VM system.

Thanks for your answers.

13

While I can’t speak directly to your organization I do work with IT managers in similar situations.

What are you paying now to keep the old system running?

The nice thing about the bounty system is that you can name your price. I am sure you can get someone to do this in the $500.00 range.

Your other choice is as you said, read up on Asterisk SMDI configuration.

My concern is not the SMDI as it is actually straight forward. Configuring the trunks and the dial plans is the difficult part of this project.

Do you have a butt set? They indicate polarity. Come off hook on one of the VM tie trunks from the PBX and watch the lights. Do they briefly switch polarity or are the DNIS (Dialed Number Identification) digits sent immediately as soon as you are off hook?

The DAHDI trunk configuration has an option called immediate. If you set it to yes the extensions (not to be confused with phone extensions) in the dialplan will execute immediately when the station comes off hook. This will allow you to create analog DID trunks to your legacy PBX.

Take a look at the www.asteriskdocs.org

This is the definitive reference for Asterisk programming. Make all your changes to extensions_custom.conf so you don’t interfere with FreePBX operation.

14

Should be…Custom Extension with no dial command and VM.

When you set up an extension you can select “custom” as the type. About midway down the page you’ll see something that says " This is a custom extension" or something like that. There is a box to enter the dial command, leave it blank. Then set up VM normally.

Yes, they would have to dial the extension again, I agree that to do this perfectly, you would need SMDI, but that costs money!!!

I assumed you wanted to try something that didn’t cost anything!!!

Bill

15

Hi

I’m trying to deploy an IVR with FreePBX and AsteriskNow for an Analog PBX but I don’t know if it’s possible or not. The idea is the Asterisk server connected to the analog PBX, could be used like an IVR with its extension.

I’ve been looking for answer but I didn’t find yet. If you know any tutorial or if you’ve any idea, please post it :slight_smile:

Regards