SIP only, 10 phone numbers, new install 3 questions

Hello, we are moving from analog phones to SIP running on trixbox or freepbx.
We have 1 SIP trunk for testing and we can call internall, outgoing, and incomming just fine.

My question is what do we do on the day we go live?
I know how to add SIP trunks, do I create a new trunk for each phone number that is being switched from our current analog system to the SIP system? Telephone company#1 handles our current analog phone system and manages the phone computer and voicemail computer in this building. Company#2 provides our Fiber Optic and will also provide a seperate fiber connection dedicated to the SIP phone system.

  1. I have a list of our current phone numbers - what exactly will I do with them. Also they are all the same unlimited plans, no special long distance or anything. Do I just make a new trunk and change the name and context to the phone number I am entering?

  2. One of the telephone SIP techs said the dedicated SIP line was NOT on the Internet, just went to their office. Does this mean it is secure and no software or hardware firewall is needed?

  3. How do I tell freePBX that all incomming lines go to the IVR? I have it set now so that “any DID any CID” goes to the IVR, but again we only have 1 trunk running.

thanks!
John

1 trunk can handle as many concurrent calls as your system can support. If your SIP provider uses the same server for inbound as outbound you only need 1 trunk.

As far as the security is concerned, dedicated fiber for a system of this size is unusual. I would do a trace route from your server to the address of the SIP server and see if all of the hops are on net. It still would be good practice to use a firewall. In this instance I would use the APF software firewall and simply allow traffic from your providers net block.

John,

I would do a lot more reading. Your questions suggest that you lack a basic understanding of telecom and SIP telecom, and that’s going to make it difficult for anyone to give you a coherent answer to your questions.

Start here:

http://www.freepbx.org/support/documentation/administration-guide

Eventually, I plan to write a complete guide to setting up FreePBX. But, until I get it done, the links under “other documentation sources” will give you a lot of useful information to get you started. If you have more specific questions once you’ve finished this reading assignment, we’ll be happy to help.

In a general response to your questions, a single SIP trunk CAN handle multiple phone numbers (DIDs), but that is entirely dependent upon your SIP provider. If your SIP provider will accept all your phone numbers, it is best to set up a separate inbound route for each DID, rather than having one inbound route that is a catchall. Again, however, your SIP provider may not deliver DID information and so you may have to set-up a DID based upon your account number. This is all provider dependent. You never enter phone #s in the trunk settings. They are always set in the inbound routes module.

Another concern you need to address is the # of channels. Your provider might handle ten phone numbers, but only give you 2 channels. If you only have two channels, then you can only make/receive 2 calls at one time. A third call will get a busy signal or a reorder.

Right, we are new to this - been reading for a month now and think it’ll take 6 months to get used to all of this. They want the phone system switched over ASAP…

I am documenting everything step-by-step so that anyone who works here in IT can get the same system working with the same settings on their own, if they ever had to. It is also what I am going to use to set the real system up when the new servers come in next week.

*CHANNELS: I’ve read some about this, but it is not clear to me at all. Yestruday I installed a linksys PAP2T-NA for the intercom and it can use port 5060 for line#1 and poprt 5061 for line2. The phone system uses port 5060. Would port 5061 be considered a channel?

We have a phone number in the test trunk. It is the same phone number we call to test the system from the outside. We also use 7777 for testing. The 10 digit phone number on the trunk page is set as the User Context, username=, and the start AND end of the registration string.

This is why we were thinking 1 phone number per trunk, but I asked because I was reading that trunks have channels.

The providor said they would move the 10 numbers to SIP so that we can have incomming and outgoing on the same phone numbers that we currently use.

The registration string appears to be username(phone#):[email protected]/phone#

I have not seen anything at all about how to add multiple phone numbers to any Asterisk based system. I guess from what your saying is that it doesn’t matter because if they give us the same 10 phone numbers, they are just redirecting those calls to our server and if they include a DID then we will know what “line” or “phone number” the incomming call was made on. If they do not include this information then we’d have no idea.

Some of those links are 404’s, but I will check out the ones that work.
Also, I will ask the SIP providor these questions:

  1. Do you provide DID information or will it be 1 inbound route using a catch-all?
  2. How many channels will we be getting? We should have 1 per phone number.

Now that you mention it, they did say something about some will be incomming and some will be used for outgoing… I will have to ask them what they meant by this too.

thanks!
John

No, a port # and a channel are not the same. I haven’t used a PAP-2 in a while, but I think it only supports one channel, since you can only plug one phone into it.

Configuration of the PAP2 is fairly complicated. I like the Grandstream HT-503 better.

If your SIP Trunk provider will move 10 numbers over, then you will probably just need one SIP Trunk configuration and then you need to configure the ten phone numbers as inbound routes (or you could just do a single catchall inbound route with the DID field blank).

It’s really up to them though- they can configure one trunk to handle all ten #s, or they can configure a separate trunk for each #. Even if you are using a phone # in the registration string for the trunk, they may just be using one # as your account number. You’ll have to ask them whether they are delivering all #s on one trunk, or going to have one trunk for each #.

Some providers pass DID information on calls, and others just pass account numbers on calls. If they pass account numbers, then you have to set-up a inbound route with the account #. Again, this is all up to them.

John, seems likes the guys have taken you under their wings and are walking you through this.

I just wanted to drop you a quick note on two things.

1 - all SIP devices on your network use port 5060, the PAP is just an extension off of your system, don’t overcomplicate it.

2 - If you want to get a bunch of items cleared up quick you may consider purchasing a couple of hours of support from the the FreePBX support team.

We currently have a receptionist phone and the Office Manager has the same thing minus the expansion keys. Each phone has 3 rows of buttons that are labeled with the current telephone numbers.

When we move to IP, how will these 10 phone numbers show up on the operator phone? See, something is missing… The phones are 6-lines phones and 14 with the expansion model. Don’t these 2 phones need the 10 numbers programmed into them or something? I know 1 phone is a 6-line phone, but the Polycome 650 manual seems to say they can have unlimited lines, just no buttons to manually access them.

Let pretend that I understand the previous questions and answers. We have no idea how to set up the receptionist phone or the backup phone to answer each line as it “rings” and such.

So this is another large piece of the puzzle that we are missing and have not been able to find documentation for… well we did not read all of the Polycom manuals yet either…

thanks!
John

OH, forgot to mention it hooked right up and only took about 5 minutes to setup and get working. The loud speaker control box just gets the telephone wires from line #1 hook

Also, the support sounds good, but it says they require remote access and the boxes will not be on the Internet. If they just gave telephone support then that would be great. Also, would they answer questions like this?

thanks
John

Support would answer questions like this however they would also help you develop a plan.

You are too hung up on this lines and keys thing, this is not a key telephone system, you don’t have a key for each line.

The line keys on the phones are extension appearances, you need as many line keys as you intend to have concurrent calls at that extension.

The ten phone numbers are irrelevant. These are digital trunks, the call comes in with a CLID (Calling Line ID) message that indicates the number the user dialed to reach you. You can use this CLID information to route calls.

The only relevant point for you is are you going to answer all of these numbers the same way or are the for different businesses? If they are for different businesses you can simply have the phone display the name of the business based on the CLID information so the operator knows how to answer the phone (done with inbound route CID prepend feature).

The side car for the receptionist is used to monitor extensions (the light comes on if the person is on the phone). You press the button to transfer the call from the operator to the user.

Experienced VoIP professionals know how to ask questions about your business and how you work so you can get the most out of a VoIP system without changing your business processes. Sure we can play 20 questions and you may be able to get your system working. Is that really a good use of your time or should you be leading and growing your business. While the software is free it is a large, feature rich system that requires experience and proficiency to implement.

I think your right, but the SIP provider, who has been our long time Internet provider, told us it was a piece of cake, download some free software and let them know when we want the phone numbers transfered over. We could not figure out why they were telling us to do everything using free software and them not making any money from the deal.

That’s how this all got started and from what I’ve been reading it takes at least 6 months of running these things to be any good at maintaining them.

If you know phone systems it is that easy. It’s also great software, with lot’s of features. It is also free!

I think my original point is relevant, a little support and you would be up and running, still huge savings.

Where are you located? I also might know someone in your area that can help you.

The only point I think I’d add to the King of the Ohio Sky is that the line buttons on each phone are line buttons for that phone ONLY. They don’t correlate to the incoming phone lines.

So, the first incoming call to a particular phone will ring on line 1, no matter what phone number the call comes in on. If the person answers “line 1” on their phone, when the next phone call comes in, it will come in on “Line 2” on their phone, regardless of what phone # is dialed, until they hang-up on the call on “Line 1.”

Because each phone has its own lines, you can’t simply put a call on hold on one phone and pick it up on another phone. Because each phone has its own lines, putting a call on hold simply holds it on your phone. If you want to put a call on hold and take it on another phone, you “park” the call in a parking lot, by transferring it to the parking lot extension (usually extension 70).

If you need to be able to know which telephone line a person dialed, then you’d want to program the inbound route for each phone number to use the Caller ID prepend feature, and add some text to the Caller ID, i.e. “L1-” or “L2-” etc. That way, when Line 1 starts ringing on a particular phone, you’ll know which telephone number they dialed by looking at the Caller ID field.

There is a complicated way of making Aastra phones do Shared Line Appearance (so that Line 1 really is like the old way), but I’ve never figured it out, and I see no reason to, as the VOIP way is, IMHO, better.

Well, the spot we are in right now is the operator phone shows extension 100 on the display (it’s own extension). I guess the inbound route (any) plays the intro Annonucement that simply welcomes the caller and tells them to press 0,1,2,3 or stay on the line for the operator.

From what I gather it will work like this:
First caller presses 0 for the operator, her phone will show a call on her extension #100.

Second,3rd,4th,etc. callers press 0 for the operator while she is still on the phone with the 1st caller. They will automatically show up on her console?

If this is the way it works the great, but it seems too simple. I also read about adding “lines” to the phone’s config file to tell it to use more lines. Not sure what this is. The standard Polycom 331 and 335 phones we have are all 2 line phones and we can place calls to internal extension as if we had 2 real lines.

*OK, Polycom IP 650 - I just used 4 phones to call extension #100 at the same time. I am only able to answer 2 calls (the phone is setup as default showing 2 “100” extensions on the screen). The other calls comming in flashed the lines already in use, and pushed the previously on hold call into a queue showing them as held on the console screen.

Shouldn’t there be a way to tell the phone that it can accept 14 calls - 1 per each line key without pushing previous callers into a hold position? I think this is what I was trying to ask originally, but also thought it would work the same way for the 10 incomming phone numbers.

Also, the IP-650 phone says it can handle 6 lines, 14 with the expansion module. It says the calls will “flow-over” to the expansion module automatically once the main phone screen is filled up.

The way it is now, the screen cannot get filled up, only the top 2 lines.

What do you think?
thanks
John

The Polycom phones have a setting for the number of appearances of the registration. You can set it to whatever you need.

I don’t recall how you provisioned your phones, if you used the endpoint manager you need to modify your templates.