atcomsystems.ca/forum Forums VOIP Phones & IP PBX VoIP & Cloud Phones, Hosted PBX, General IP Phones Why don't IP phones have universal many DSS buttons like digital system phones?

Siemens IP phones have 98% of the PBX features that their system have including DSS, BLF, Vmail keys, soft keys and just about everything else you could think of, including paging and intercom. I think that we are all going back to the point that (for the most part) a manufacturer of telephone equipment, be they Alcatel, Siemens, Toshiba, or Mitel can and do make good IP phone systems, it is when a data company decides to make a telephone system that things get poorly done, even if they have a HUGE data name, cough cough, CISCO, cough....they may sell some volume of equipment, but it is not thought of as a telephone system by its programmers, it is thought of as a data network that happens to have telephones on it, thereby making them second fiddle to other issues.

Steve

No mention of Inter-Tel here yet?
Our IP phones have full functionality of our digital keysets and at first glance look exactly the same.
If you use one our dual mode phones in SIP mode, of course you'll loose many of those features...
They are turning out to be a decent product and a good fit for some businesses.

First I'm a computer guy and an asterisk guy, I don't know much about analog PBX's although i've worked with them a few times...
Skimmed the thread, few comments...

First- Asterisk by itself is like a big tub of Legos. You can make something very cool with it, IF you know what you're doing. If a project is to from scratch set up an Asterisk install to do (features), how well that will work is very dependant on how familiar the tech is with *. Unlike a normal small biz PBX or key/hybrid where you can fill out the programming sheet and then follow a procedure to program it, Asterisk must be created uniquely for each installation, down to even the simple stuff like making one extension dial another. For someone that is new to *, this can be VERY frustrating. Additionally, simple features that most PBX's have must usually be added by hand to the dialplan. This turns a lot of people off Asterisk; it does not seem useful to have to sit and set up manually 'features' like the ability to dial out when you can buy a PBX and plug it in and be done.
However Asterisk's great weakness is also its strength, because the legos are not glued together like a PBX, you can make literally ANYthing you want. There are no limits to number of ports, number of extensions, or how much voicemail can be stored, you are only limited by the machine it is running on. Never will you have to pay more money to license more voicemail ports, for example. And the classes of service Asterisk can provide far outspec any PBX you can come up with. This is how asterisk got its name- in computer terms, the "*" character is a wildcard, which means it matches any '0 or more characters', aka * matches anything. Asterisk is much like this- it will accept a call on almost any sort of interface and protocol, do almost anything you want to it, and spit it out again on almost any other interface. It is the ultimate in flexibility.
That's not to say its perfect. Its complication turns off many users, and to address this need packages like Trixbox have sprung up. Trixbox is a full *nix distribution (operating system) that installs itself from a CD. In about 30 minutes, it will take a normal computer and turn it into a somewhat slick PBX with a web interface to configure it and all the trimmings of a normal PBX, no programming required. Many companies are selling the same thing as PBX appliances, sell a machine with asterisk+stuff on it as a pbx unit and are making tons of money doing it.

I have worked with Asterisk since before v1.0, hell even before there WERE version numbers (you just checked out the development code...). I have found one, and only one feature that Asterisk cannot usefully do- SLA (Shared Line Appearance). This is how (on a key system or hybrid), you have a LED button assigned to an analog line. If someone uses the line, it lights up. If you push that button, you barge in on the line and are now 3waying with the guy on the line and the other user who called him. Asterisk has yet to develop (that I know of) a useful way of doing this. However due to large demand (more on that below) it's planned for inclusion in their next major release (1.4).


Back on topic of the thread...
Many of the IP phones out there do not have a handful of programmable keys because in a large enterprise environment, they are of limited use. In a company with 500 people, you will only need 5 or 10 keys- your assistant, others in yoru workgroup, maybe a few other things. The other 495 people you don't care about. Until recently, real VoIP installations have been limited to either proprietary products from various companies or very large deployments as I mentioned above.

There are a few models that address the 'receptionist' need, for example there are a few Cisco phones that can take sidecar units which are nothing but a big pad with 40 or so programmable LED keys. Snom as I recall has a sidecar for their excellent 360 phone (highly recommended). However, 95% of the users in a large enterprise don't need 20 softkeys, as many of them (hold/xfer/conf/etc) are already on the phone.

In short- there was no demand for them. That is changing. VoIP has gone mainstream. 'Cheap' IP phones don't cost $300 anymore, service is more widely available and more reliable, and everybody wants in, especially SMB where pennies count.

Things like web interface panels and computer software are workarounds to attempt to address a newly formed need, and they do a decent but not very good job of doing it. In a home or SMB environment, no software or LCD soft menu will be able to replace picking up the phone and pushing the ONE button that dials who you want.

So I suspect that over the next several months, manufacturers will start to offer IP phones with handfuls of keys as the Home/SMB voip market develops. Inter-Tel is one that I know of, their phones have a handful of softkeys. (Intertel phones can also use a proprietary intertel protocol to work with an intertel PBX...) Snom 320 and 360 have 12 keys + all the other buttons so they do not take softkeys for VM/DND/etc. AAstra 9133 has 9 keys as I recall.

Well, IronHelix, I must say that you are among a very few people who are more IT-oriented to come here and have a more realistic outlook in the TDM to VOIP convergence. Most people approach this as "pro or con". I am one of them.

I enjoyed reading your post and feel that if manufacturers of TDM systems slow the pace in their rush to get into the IP side of things, they may be met with less resistance by their existing dealers.

You did a good job of offering a better view of how things are heading. Please do what you can to help us all migrate in to the new technology. I think that the overall consensus is that this technology was dropped into our laps overnight, while existing voice technology has been just fine left alone for nearly thirty years.

Nobody here can claim to know everything, so I say welcome aboard and we look forward to your input and guidance.

Now, if you would please complete your profile information, it would be appreciated so that we know who/what/where you are. It's just a common courtesy that we ask of all members.

While I speak for myself, I look forward to more information that you have to offer.

Thanks for your kind comments... profile filled out And just to warn you, I have a habit of being... verbose... at times...

I think it boils down to in some cases a fundamental difference between the existing telephony market (manufacturers, installers, users, etc) vs the IT market- IT guys are used to having large changes happen relatively quickly, which is required when the life cycle of a product may be only 4-5 years at the outside. (Would you use a computer from 1996? Probably not. Would you use a telephone from 1996? Absolutely, it's still as good as one you buy today). I recently had a VoIP client who only hired me because their 15yr old KSU cought fire when they tried to add another extension to it. Other than a few problems with old analog wiring, they loved the thing and were sorry to see it go. I've seen a few clients with 10 year old computer systems, but not many, and they are usually very eager to get up to date.
So as a result, since VoIP technology was being developed on the computer side of things, it's rolled out rather quickly as soon as the products are mature enough to be used and there is demand. Demand especially, VoIP has become a buzzword- a lot of people (customers) don't quite understand how it works, but they know it will probably save them money so they want it. A lot of other customers understand the flexibility it will give them, and they want it even more. Demand springs up overnight, everybody starts making VoIP gear, and this leaves an entire industry going 'WTF just happened?' as their product lines are turned upside down and their industry in general jumps ship into the IT Department side of things...

For those that want to get into VoIP more, a few suggestions-
First, understand that hardware is not software. A standard analog PBX may have modules which either work or don't; VoIP is often based on a computer and there are a thousand more levels of complexity (and thus problems) that go with it. Hardware failures happen, but software related issues (misconfiguration, bug, conflict with something else, network problem, etc) are going to cause most of the issues you have.

To that end- learn computers. Understand basic computer troubleshooting, networking, etc. Especially networking. I can't speak for big manufacturers, but I could guesstimate that a lot of the VoIP server systems are going to be software that will share a server with other things. Asterisk is such a system. If you can come to a clien't site and competently install it on an existing server, you will be a much more valuable worker.
When you learn this- CONCEPTUALIZE, DON'T PROCEDURALIZE. What i mean by that is 'To do X, perform in order steps A B C D'. If you just remember that procedure, and don't understand the concept of what steps A-D do, you will be far less effective. On the other hand, if you understand what the steps do, then it is easy to figure out that to do Y, do steps A D C E.

Anyway i'll stick around and I have email notify turned on, so if any of you have questions about VoIP or Asterisk, ask away

Ironhelix if you get a chance please post the link to your paper on * being installed on an AT@T Partner system as a voice mail replacement. You did a very thorough analysis of the installation and programming. Saw it on another board. Great info.

Welcome to the board IronHelix its been interesting reading your posts I am glad to see Astrisk being discussed and hope you can teach us more about it. I would like to see that article RBF mentioned also. I hope you hang out here for awhile but I'll warn you some people here don't like Voip or IT guys. I think * would be a great addition for several of my customers and want to learn more about it. If you want into the installers section you can post your request here.
https://www.sundance-communications.com/forum/ultimatebb.php?/ubb/get_topic/f/1/t/001607.html
:rofl:

Hi, here is the link for anyone interested- my writeup on replacing the PartnerMail Short version- client didnt like his PartnerMailVS and wanted voicemail-to-email so I set up a VoIP server connected to 4 free extension ports that pretends to be a PartnerMail. Works great.

I've already noticed some resistance to VoIP but it's cool, everybody has an opinion. I'll stick around and answer what I can, and you all are welcome to PM me if you have any VoIP or Asterisk questions...

LOL I stumbled in here to read everyone's post and no one has posted since August 2006. I take it everyone is out trying to teach customers how the hell to click on a web browser and answer a phone call. These are the same people that need instructions on how to answer a phone that's ringing audibly with a flashing light. I've seen a lot of changes in telecommunications in the last 30 years and all I can say is god help us...


CISCO = Can I Still Call Out?

Hmmm....

Well, this is what I get for playing with my blackberry pearl (the boss got them for us, I think I'm getting used to typing on this ruddy thing!) and going to this site!

Well as for us, we are working on a service platform that uses a T1, SIP trunks, and the asteriskNow appliance. We're still working out some bugs, but will be selling soon. Bo' it can't do all the things the InterTel, Avaya, DSX, etc. Can do, but then, it doesn't have to!

We see it as a price point VOIP system (sorry, VOIP isn't going to go away). Eventaully, we all will have to adapt.

Also, remember, TDM systems won't just disappear for a loong time either!