Our existing telephone system is Centrex Mate (AT&T), where nearly all extensions can be dialed directly (accessed) from outside. We also have a main number that is answered by an IVR, so one can dial the main number and then transfer to an extension (which can also be accessed/dialed directly).
If we are to adopt a complete VOIP solution and use Asterisk/FreePBX as the PBX an call processing switch/server, how would we need to set it up? I know we will have to sign up with a VOIP provider for externally access/dialing (to the rest of the world), but my question is, can I setup the FreePBX extensions that can be dialed directly from the outside world? Any examples/illustrations would be most helpful. Thanks.
Ultimately SIP providers have to route calls on to the telephone company networks which they usually do by putting the calls on to PRIs. The real issue of using PRI vs SIP is really about economics. A VOIP carrier is charging a [perfectly legitimate] markup over the PRI prices they pay for the services they provide to you of taking your call and putting it on to the PRI.
At sufficient volume, it is simply more economic to do this yourself but the trade off point will vary around the country depending on the pricing of PRI to your location. Bill has suggested that the trade off point is around 15, but in my location [Silicon Valley] and in the case of competitive markets around the country, that tipping point can be as low as 10 channels.
This is a matter of working the numbers and calculating an ROI. For a PRI you pay a fixed monthly fee [I have seen as low as just under $200/mo and as high as over $400/mo] but usually the per minute charges are minuscule [and of course, you do have to recover the cost of your PRI card for your server]. Run the numbers and make your choice, but you will have to call companies that provide PRI services where you are located, that cannot be done via a forum like this.
Inbound Routes is what you need to do this. If you want to keep the same numbers, you’ll have to do an LNP port. The big question is how many inbound trunks you’ll need. If it’s over about 15, or so, you may want to use a PRI rather than SIP.
Assuming you have a VoIP provider that can provide all your DID’s over a single trunk (remember, in the VoIP world a single “trunk” can handle multiple simultaneous calls, up to whatever limit your provider may set), then in your inbound routes you simply have to create an inbound route for each DID, and you can then point it to the extension number that corresponds to the last few digit of the DID (or any other extension, or a ring group, or however you wish to route it). If you only have a few DID’s, then it is easy enough to set them up by hand. If you have several to several hundred DID’s, then you’ll probably want to obtain and use the third-party Bulk DIDs module so that you can use a text editor (or other program) to create and maintain your DID list.
The only real problem that may occur is if the provider doesn’t send the DID information in the expected manner in the SIP headers. In that case, FreePBX won’t be able to determine which DID is associated with the incoming calls. Sometimes, if the DID information is available in the SIP header but not in the expected format, it’s possible to write a bit of custom dial plan to extract the DID number (often only a couple of lines of code are needed). But if the provider isn’t sending the DID info in the first place then there is no way to extract it, so if this is a major project you might want to make sure that there are other FreePBX or Asterisk users already using that provider’s services, and not having any issues receiving the DID’s.
Edit: Bill and I were apparently posting replies at the same time. His idea of using a PRI might be a good one depending on your circumstances, but I have only ever used SIP or IAX and never with a huge number of DID’s (I think the most I’ve ever had is five on a single trunk).
What is LNP port? We have over 1000 extensions (numbers). Why do you suggest PRI over SIP? Is there any specific advantage with PRI over SIP? Actually, we began to consider a VOIP solution owing to the high cost of PRI (monthly rates) that we would need in order to convert our existing legacy voice mail system (Centigram) to Asterisk/FreePBX and interface the SMDI with Centrex Mate. The existing Centigram VM has a bank of analog POTS lines feed to it with a modem b/w Centrex central office and our VM server for SMDI/MWI. The outside consultant we contracted to configure/program the SMDI/MWI, basically a conversion from Centigram to Asterisk/FreePBX, tells us it would be very difficult to configure the SMDI since those analog lines do not pass on the enough information, like which station is being called, whether the call is being made by a user to the VM to retrieve messages, or whether it is an outside call, etc. According to him, if we went with a PRI line that would include all those relevant information for SMDI. Our budget however, will not permit us to allocate the funds required for the monthly recurring charges for the PRI rental. Thus, we are back on the drawing board and now take a look at a total VOIP solution, ditching Centrex PBX.
Thanks for your advice.
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Assuming you have a VoIP provider that can provide all your DID’s over a single trunk (remember, in the VoIP world a single “trunk” can handle multiple simultaneous calls, up to whatever limit your provider may set), then in your inbound routes you simply have to create an inbound route for each DID, and you can then point it to the extension number that corresponds to the last few digit of the DID (or any other extension, or a ring group, or however you wish to route it). If you only have a few DID’s, then it is easy enough to set them up by hand. If you have several to several hundred DID’s, then you’ll probably want to obtain and use the third-party Bulk DIDs module so that you can use a text editor (or other program) to create and maintain your DID list.
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Since I have just begun looking at a VOIP solution, I yet to research any DID/VOIP provider. Could you recommend one?
We have over 1000 extensions (telephone numbers), so we would have to use the Bulk DID module to configure. That shouldn’t be a problem.
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The only real problem that may occur is if the provider doesn’t send the DID information in the expected manner in the SIP headers. In that case, FreePBX won’t be able to determine which DID is associated with the incoming calls. Sometimes, if the DID information is available in the SIP header but not in the expected format, it’s possible to write a bit of custom dial plan to extract the DID number (often only a couple of lines of code are needed). But if the provider isn’t sending the DID info in the first place then there is no way to extract it, so if this is a major project you might want to make sure that there are other FreePBX or Asterisk users already using that provider’s services, and not having any issues receiving the DID’s.
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If this is a common issue/problem then we would have to make sure the VOIP provider is capable to deliver exactly what we need. Is it a common problem?
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Edit: Bill and I were apparently posting replies at the same time. His idea of using a PRI might be a good one depending on your circumstances, but I have only ever used SIP or IAX and never with a huge number of DID’s (I think the most I’ve ever had is five on a single trunk).
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Since both you and Bill suggest PRI over SIP, I would like to know what is the disadvantage with SIP and what is the advantage with PRI?
What is the industry standard? Doesn’t a complete VOIP end-to-end service exist where you would not need something like a PRI when large number of DID extensions are needed? Aren’t there any reputable VOIP providers in existence today to cater to medium to large companies with many many direct dial telephone numbers? Is that why most companies with legacy Centrex PBX are maintain the status quo and do not migrate to VOIP?