Business Phone System Upgrade - Question about Analog and VOIP Extensions

Hi,

We’re currently using a Norstar system from back in the day and I would like to upgrade our system to a new PBX with a combined analog and VOIP solution.

Our system right now has 20 Analog lines.

I would like to create a system that has 8 Analog lines and unlimited VOIP Trunks (looking at flowroute.com). I plan on purchasing an FXO gateway for this and new IP Phones. I will also install FreePBX. The reason for the 8 analog lines is certain people cannot afford any call quality issues due to potential internet outages.

The goal is to have everyone call in to our company number and then for the receptionist to be able to transfer calls to varying extensions. The number people call in to will be an Analog number.

I was wondering if calls from this single number will be able to transfer to VOIP extensions and Analog extensions, or would I have to use some form of call-forwarding to separate DID’s. I want the receptionist to simply be able to dial the extension and it will transfer from the analog line to the person to a new voip trunk, or one of the 8 analog lines quickly and rapidly for the phones I have laid out as voip phones.

The ideal is for everyones phone to have access to both analog lines and the voip trunks, but for certain phones to have priority for the analog lines.

I was also wondering if it’s possible to say all outgoing long-distance calls will use VOIP lines, and all incoming long distance and local calls will use the Analog lines in the PBX, however only in the case if a certain minimum number of lines are still free for the priority people, or is this getting too complex

I can’t stress enough how much I appreciate any help you’re able to provide and help lift my confusion on this topic.

Thank you again and cheers

You will be able to transfer calls to VOIP extensions or analog extensions.

What I think you are asking is if you will be able to have a call that came in on an analog trunk transferred to a VOIP trunk. That will not be possible.

I understand what you are asking but I don’t think you’re going to get the results you want with the system you’ve thought out. As mentioned, transferring from analog to VOIP isn’t really possible because the analog line is still in use. The only way to do that would be for the phone company to hand off the call for you and every company I know will call that line “in-use” as long as their circuits still have the call to any end point. In other words… it won’t work.

The outgoing part of your suggestion will work fine with outbound calls on VOIP but a person CAN use the analog lines by dialing a code first, say 6-xxx-xxx-xxxx for an anolog line or 9-xxx-xxx-xxxx for voip. It can default to voip if no code is dialed etc.

You COULD make it so people call the analog lines and they only roll to VOIP once all the analog lines are full (this is done at the phone company, not on FreePBX).

Bottom line, you want a system that needs to be thought out better than you’ve thought it out so far.

Some analog lines (at least here in the states can be provisioned for “call-transfer” which is basically a co handoff of the call, thus freeing the pair (oops, I mean analog line, old thinking dies hard.) , to do that that you will need to send a “flash” of the right length on the anolog line, send the transfer-code, dial the number to be transfered to and hangup. Although common on Centrex systems, I noted the flash length as Centrex hookflash is shorter than normal flash, 200 ms v 750 ms. or something ( I suffer from CRS).

I note that many providers still offer it.

As to integrating legacy PBX’s then

http://www.voip-info.org/wiki/view/Asterisk+legacy+integration

is a good starting point.

Problems will be:-

  1. Where do you want your voicemail/forwarding/presence handled
  2. who will turn on who’s lights
  3. etc.

most are discussed there. It works well but you will need to intimate details of your current PBX’s signaling and also Intimate details of how contexts work in Asterisk.

Good Luck, when you get it working you will be impressed with yourself.

Call transfer is a Centex option that is part of Feature Group D.

I don’t think it is available unbundled on a 1FB busy line.

PRI lines of course can be sent a redirect.

The Asterisk dialplan can be hacked to send a hookflash, however I think that it will provide a less than satisfactory integration. Ports are cheap. I would put the Asterisk in between the Nortel and the CO. In other workds 8 FXO and 8 FXS ports. This would allow the greatest flexibility.

Without a digital line the Asterisk box won’t be able to send ANI to the Nortel for station level access. I do realize that you can program analog DID lines.

If the Nortel has any analog ports in it one of those could go to a group of FXO’s on the Asterisk box and provide direct access via a route.

Since in this scenario the Asterisk is providing dial tone to the Nortel station level dialing back to the Asterisk can be programmed in the Nortel outbound dial plan.

Perhaps they where nice to me, I had it on my 1MB’s (at a small cost), no longer use it as I only use 1MB’s for alarm lines anymore.

MB or “metered” lines are great for 911 access, alarms and such.

Thanks Scott, a good discourse, Whether you use Nortel, Mitel Avaya, Panasonic etc the question remains,

Who does Vmail, if Asterisk then full integration also will require

externnotify = somethingyouwrite.sh

to make the lights go on on the pbx

If the PBX does vmail then the converse, i.e. to send that vmail to asterisk in the right format, and so signal the external stations.

As you can imagine ,the first option is easier.

As to call flow, that is generally trivial, best way is in my opinion to build a trunk between asterisk and PBX with context from-internal , if this is not possible then perhaps custom extensions on asterisk that dial the pbx trunk as appropriately dial/ZAP/g1/ext or such (G1 not G0 to reduce glare)

Although it is simple to integrate a PBX, it is not not to be done without knowledge.

JM2CW

dicko

I always thought is was an acronym for Measured Business ?

I always have bought them as that, perhaps I was calling them wrong. I will check.(Pac-Bell-> AT&T, GTE->Verizon, stoopid_vendors_in_the_desert, etc. at least)

Sorry but I like rigor when talking to the telcos, the are incapable of doing otherwise.

The phone company here considered the line “in-use” even after handing off the call via the Centrex transfer. It took me several calls before I was able to confirm this is intentional on their part. Anyone calling me will get a busy signal after the transfer even though no lines here are being used any longer as the call was handed off to a VOIP line.

We have a couple of analog lines that are permanently forwarded to our PRI. (*72xxxxxxxxxx). This is done so that in the event of a failure of the VOIP box, a backup phone can be plugged in, call forwarding taken off and the line answered. There were a couple of problems with this, the first being that only one call could be made at a time. After working with TELCO, they added 100 “paths” to the line which allowed up to 100 calls to be forwarded at the same time (Hope we never have that many). The second problem is that the analog line is centrex measured service…1 cent a minute, so every call that comes in is charged that way.

As was said earlier, transferring a call on a pots line at the trunk level is a centrex (or similar service) feature.

Anyway…All of the mumbo jumbo aside. With 8 POTS lines and an undetermined number of SIP trunks I wonder if the cost for all of this isn’t approaching or even exceeding the cost of a PRI, which would totally solve your problem of reliability. You didn’t mention if you had looked at that option, perhaps I missed something in your post.

Bill