Cisco 7975 Phones and New user to FreePbx Business

I run a small company. We have 3 phones. I would like to use FreePBX for my company full time and save some money. I would need the following features:

1.) Software based phone on laptop 2 lines
2 lines per 7975 Cisco phone in house.

I need to be able to monitor if someone is on the phone and transfer calls, along with voicemail.

Is this something freepbx can do? How many lines do I need to purchase through sip station and what i the best guy to setup? Our internet is also not a business line but we do use dyndns.

Since you’re just starting, some info that might help:

  1. FreePBX Is a “management suite” for Asterisk, which is the actual telephony engine pretty much everyone (the cool kinds, anyway) uses.

  2. Monitoring calls is an Asterisk feature, so you should be good to go there.

  3. Soft phones are usually a relative piece-of-cake.

  4. Cisco 7975 phones are “interesting”. There are two routes you can go to get these working. I have a preference, so you might want to get a second opinion (more on that below).

  5. SIPStation connects via something called a “trunk”. It’s different than a traditional phone system, in that it can carry multiple calls at one time, and can have multiple numbers assigned for inbound dialing. It’s s single connection that your system handles separately from the incoming call and outgoing call processing (which are both handled separately as well). SIPStation is the sponsor for these forums, but there are other low-cost providers out there. You can have pretty much as many as you want.

  6. The fact that you don’t have a static IP address isn’t a huge problem for some ITSPs (SIPStation, et al.) and is a problem for others. My primary ITSP requires static addrtesses. Check with SIPStation to see if they have a problem with DynDNS addresses. I don’t think so, but you NEED to check with them - anything else is a rumor.

  7. SIP (and SCCP) phones do not rely on “lines”, so your requirement that each phone in the system have “two lines” might not make any sense. Each device “home runs” to the server and you assign one or more extensions to the phones. You can implement Call Waiting if you want, or you can set up each phone with multiple extensions so that they can have the phone ring when another call comes in.

  8. It’s important to understand that, once you put Asterisk in, you now have a server that answers your phone and routes it to an extension (or extensions). You can have as many calls come in on your primary number as your trunk supports, so there’s no longer a requirement for hunt-groups or fail-over numbers. If you have two numbers now (a local number and an 800 number, for example), your ITSP will assign both of them to your trunk and Asterisk will receive them. It will then route them to wherever you want to send the calls.

In other words - almost everything you want to do (except maybe the 7975) is an hour’s work.

Now - on your Cisco phones. There are two ways to do Cisco phones, the native Skinny Protocol and the SIP protocol. They require different firmware images to be loaded to the phone (so the phone can be SIP or SCCP (Skinny) but not both.

If you decide to go with the SIP firmware, the phone are relatively easy to set up and use, but you lose a lot of the features normally associated with the phones (programmable buttons, VoiceMail light can be a problem, etc.) They can also require some very specific changes in your servers configuration. When Cisco released the SIP firmware for these phones, they couldn’t include a lot of extra features because the hardware just can’t handle a lot of the “cool stuff” that SIP does in the phone. Also, the phones are limited to 8-character SIP passwords, so it is very hard to keep the connections secure.

If you decide on the SCCP (Skinny Call Control Protocol) firmware, you get the full functionality of the phone, but it’s harder to set up. You have to download the “Chan-SCCP-B” driver, compile it (which means you need the Asterisk sources for your Asterisk software), and install it as an optional module. After that, you need to set up the “sccp.conf” config file to point the phones to your Asterisk extensions. The SCCP interface relies on the phone’s MAC address and internal device name to connect to the server, so is inherently more secure.

Once done, either configuration can handle using the phone as a phone. I find the Skinny firmware to work better, the programmable buttons work better, and I can control the lighted buttons and VoiceMail lights much easier. I also don’t have to worry about someone figuring out the password on my phone and calling Dubai on my dime again.

First I want to thank you for replying. I don’t have to use 7975’s but I had one around for testing. What guide should I follow to get this setup? What phones would you recommend? Like I said, I’m pretty basic. Also is there a free softphone I might be able to use to get started? Are we allowed to sell these systems? Seems like a great solution.

As far as the FreePBX of the setup goes, just go to the website and download the latest distribution.

That will get a lot of it set up while you sip on an iced tea.

The Chan-SCCP-B stuff is all one sourceforge - search for it. You can also look for “chan-sccp-b freepbx” and the guide I wrote to set up the system should pop up.

The 7975 is a fine phone - I prefer it in SCCP mode, but the SIP load will get you on.

Sangoma makes some passable phones (they are new at this).

Zoiper makes a good cross-platform softphone that I find works pretty well most of the time.

Are we allowed to sell these? I hope so - it’s the cornerstone of my business. On the other hand, if you decide to sell it, you’re going to have to be a real expert or you screw it up for the rest of us that have invested literally over a decade on supporting these systems.

most my client have maybe 5 phones, was just thinking out loud but wanted to test a while on my end. I’ve been a linux admin for over 10 years but this appears to be all gui. I have downloaded that but was hoping there was a step by step guide in at least getting the basics so I can make calls, etc.

You mean more than this guide?

I am a little bit confused on the guide. Isn’t this all gui or do I need to go to the command line to start the process?

Most of the actions in the guide require access through the console.

Chan-SCCP-B is separate from FreePBX, so until I get the kinks on the FreePBX module ironed out, working from the console is the fastest way to get everything done.

I take it this is just for the cisco phone setup? Where is the guide to get auto attendant setup and route to an extension? The idea is for each phone to be able to access 2-3 calls at one and transfer calls. Also be able to see if another user is online and mailbox function. The only other thing I can think of is a software that installs on the workstation where it pops up when calls some in, and has logs, etc.

Got up and running last night with x-lite for testing. I am going to pass on Cisco for now. Does polycom out of the box? Which models would you recommend? I need to purchase a few. Also how are you presenting this system to customers? Meaning do you just tell them it’s an asterisks system or what do you call it?

Just my two cents…Sangoma and Yealink phones are probably the easiest to setup out of the box especially if you purchase the endpoint manager module along with the Restapps. Maybe worth taking a look at :slight_smile:

whats so great about end point manager? Are the polycom phones hard to get setup and going? They are very popular in my area. What modules are most people purchasing? I do need auto attendant, and a software app for the pc that shows call logs, etc

Yes, it is to connect a Skinny Cisco phone to the system.

That would be FreePBX - once the phones are connected through Chan-SCCP-B, you assign them extensions which are then connected through the “CUSTOM” extension type to the “SCCP/ext” dial string. After that, they are just phones like a Polycom or a Yealink.

This is very simple on the Cisco phones in Skinny mode - you add the hint to the end of the button definition (the third field is the hint) for each phone definition. Also, IIRC, the SIP image phones can only have one line definition. The Skinny image phones can have all of their buttons defined as lines. In fact, the Skinny phone can have more lines than buttons (up to 64, as I recall).

Your requirement to be able to “access 2-3 calls at once” has me a little confused. I’m not sure what you’re talking about there.

Transferring calls is handled through Asterisk, so that’s actually outside the scope of this entire discussion. The soft-button configuration is Fixed in the SIP image, but is manageable in the SCCP configuration, and actually follows the phone’s current call-state.

This is outside the scope of Chan-SCCP, FreePBX, and Asterisk. Note that once you install the CRM interface that talks to your phones, that functionality is easily added to your extensions, regardless of connection technology.

I think I am going to pass on cisco for now. I did purchase Endpoint and sysadmin pro. I am looking at polycom phones for testing and have purchased 1 trunk line from sipstation. Do I need multiple trunks if I want multiple lines? Is sipstation really the best option? I’m in a trial right now. I want to use polycom because that is what my customers are used to.

Possibly a good call! We use the Cisco 8845 phones and personally I love 'em. They run SIP and there’s no loss of functionality at all, but they are somewhat ‘involved’ to get running first time. I’ve no experience with the 79xx series and SCCP yet but if a phone supports SCCP that would be the way to go over SIP for sure.

I’ve used Polycom and I can’t stand the configuration process, it is greatly helped with EPM though. The simplest phone I ever found for configuration was the old Aastra (now Mitel) phones.

With reference the the lines question, be careful with the term ‘trunks’ and be aware of ‘channels’ even though ITSP’s will mix them up too. A trunk would be like a physical phone line, a channel would be the number of simultaneous calls that trunk can carry. As a SIP trunk has no inherent limit other than bandwidth, you could have one trunk configured to 1000 channels. So no, you don’t need multiple trunks, but you will need that trunk configuring to multiple channels.