Replacing old phone system
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@dashrender I don't think you understand. If you have a computer at a location, it has a line. Plug that line into the VoIP phone. The phone as a port that goes from the phone to the computer. One "backbone" line you could say for two devices. No need for PoE or injectors because phones are powered via AC plugs. As far as licensing, yea, spend the money on new phones. Elastix has no licensing limitations or extensions cap, etc. What's the issue here? All you need is the extra 3 ft Cat 5e/6 cables.
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@ajstringham I do understand that I only need one ethernet port at each desk to run both a phone and the computer (I'm doing that now in the second building that has VOIP phones already.
I want PoE because I don't want to use "phones are powered via AC plugs" as you put it - I said power bricks in my previous posts. More plugs under the desk just invite more problems (and add to the cost because most phones don't come with them) Since I have to have the cost anyway power brick or PoE switch, I'll take the switch if available.
Most phones come with a Cat5e cable so I wouldn't need anything there.
I did understand that Elastix does not have a licensing cost, it's just the phone cost itself.
Earlier when I mentioned that Asterisk was made by Digium ( I didn't say Elastix was made by Digium - should have said Mark Spencer of Digium) is that wrong? And a search I just did showed that Elastix is based on Asterisk and several other projects.
Let's pause on this conversation until I get a full phone count.
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@Dashrender Ok, sorry about that. Misread a couple things.
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At the time of the original post, Elastix was still an awesome project. Since that time it has really gone down hill. I would skip Elastix and look at FreePBX today.
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yeah we want to know how our friend @Dashrender ended up in his voip setup project ??
i'm curious to know
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I was all set to make the conversion from our Mitel system to a FreePBX system in July 2015. Unfortunately many of the features which only exist in their precise form on Mitel. When other department leads were brought into the conversation, they immediately indicated their complete disapproval of the loss of those features.
Unfortunately I can't recall the exact line item I couldn't achieve that lead to me immediate reverse course on the deployment of FreePBX, but the decision to change gears and go with a Mitel solution was made and acted upon.
Features that were different:
Console - FOP2 was a significant step backwards compared to the Mitel console. FOP2 was difficult if not impossible for the operator to create their own groups that quickly allowed them to see status of given phones. Also calling someone from the console was a challenge, while not insurmountable, it was just one more strike against it.DND - a SIP based solution works completely different than a proprietary system. SIP phones traditionally just provide a busy back to the PBX, This gives the PBX itself no information about the real state the end user desires. I.E The default DND can't tell the main system (and by extension other users on the system) that a user is at lunch, in a meeting, etc. Only through the use of dialing codes was this possible. This could be made a bit easier for the end user if the phone had many buttons that could be programmed to enter the codes automatically, but often times the display wasn't forthcoming on the status of DND on the endpoint. This could lead the end user to easily not realizing their phone was on DND.
Call parking: This would have been a complete process change from how Mitel works. Mitel provided an option to transfer a call to a given extension and place that call on hold on that extension with no action required by the person sitting at the extension in question. From a customer service perspective I consider a bonus. Example: A caller calls in, the Operator answers the call, places the call on hold, then calls around to locate someone to take the call. Once that person is located, the operator returns to the caller and transfers the call on hold to the found assistant. If the hold option wasn't available, the assistant would often have to take the call, then tell the caller they will be putting them back on hold while they (the assistant) finishes the task in front of them). The hold ability allows one fewer interaction, and the probable better 'feeling' of the caller, i.e. they feel less like they don't matter because, hold please, hold please, hold please, etc.
Switching to the calling parking situation that is available in FreePBX required the Operator to first send the call to the parking lot, noting the lot number.. then finding a person to give the call to and giving the lot number to the assistant. The assistant then has to remember that number while finishing their current task before retrieving the call. All the while hoping someone else doesn't grab the wrong parking lot number and moving the call inadvertently.
Of course neither of these situations are my (or others) ultimate desire. Instead we'd rather just see a ACD that these callers could be dumped into, and the available assistants would get calls from the ACD as they were free to work on them. Sadly the assistants have made this a more or less unworkable solution up to this point, but changes are being made in management, we might get this change soon (as in Jan/Feb 2016).
These were the larger challenges I faced. If I can recall the item that ultimate pushed be away from this, I'll come back and update the thread.
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Things of note:
I know that Scott really trusts getting your phone calls over SIP via normal internet lines, and he has provided countless stories of telephony companies that have PRIs down for months at a time.
The volatile nature of the internet leads me to concerns of throughput and voice quality.
When providing options to management I offered both traditional SIP trunks through one of the many providers mentioned on ML as well as a much less flexible but separated solution provided by Cox. Cox offered a dedicated line and a SIP gateway device. I suggested that we could back this service up by having a secondary SIP provider that we could transfer the calls to in case the lines between us and Cox went down.This solution was the one chosen.
Phone:
Yealink phones are definitely inexpensive. There are a great starter phone, but compared to the Mitel phones I had before, they felt cheap. The handsets have a low arch that make them difficult to hold the phone to your ear with your shoulder. If this is something you do regularly, you better expect to either deploy headsets or the foam addons for the handset.Also, determine the button layout you'll need before you decide on the phones themselves, you may find that you need/want more buttons than the less expensive phones will allow.
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@Dashrender said:
$25K is for the whole solution - drop in appliance running Digium's version of Asterisk (Digium invented Asterisk, right? so everything they do is Asterisk based) and 85+ phones and PRI port support.
Don't be fooled Switchvox is only running on top of Asterisk. There's a lot of proprietary stuff with their phones that go with it. You can still use any desk phone but want get the fancy features they market on with them, and will need to manually configure the end points. in which case freepbx really makes more sense.
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Another sticking point was conference calling. We need the ability to conference 3-4 calls together without using a conference bridge. I don't recall the problem, but I don't think I was able to get more than 3 parties onto a single call without using a conference bridge. That situation was/is unacceptable. We needed to be able to conference people onto a mega call in a moments notice, the thought that we would give people information on how to dial into a bridge would be to time consuming and expecting to much from people that don't work in our office that are part of these calls.
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@Dashrender said:
Another sticking point was conference calling. We need the ability to conference 3-4 calls together without using a conference bridge. I don't recall the problem, but I don't think I was able to get more than 3 parties onto a single call without using a conference bridge. That situation was/is unacceptable. We needed to be able to conference people onto a mega call in a moments notice, the thought that we would give people information on how to dial into a bridge would be to time consuming and expecting to much from people that don't work in our office that are part of these calls.
I believe that is a function of the phone supporting enough lines not nessecirly the phone system itself.
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so you didn't complete the transition to FreePBX, you restored your Mitel VOIP system because you found FreePBX limited and not meeting your need ?
i'm sorry to ask you this question because i didn't well understand your story because my english is not that good so i was confused, did what i understand is correct ?? -
He ended up going with Mitel. FreePBX did not have some functionality that was needed.
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Did you audition Symphony instead of FOP2? FOP2 is not part of FreePBX, but Symphony is their official add on for that type of use and has a lot more functionality.
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I learned about Symphony in the 11'th hour so I never tried it out.
What I remember was that ran into a road block that I felt I could not resolve in the 15 days before the go live date I had looming.
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@IT-ADMIN said:
so you didn't complete the transition to FreePBX, you restored your Mitel VOIP system because you found FreePBX limited and not meeting your need ?
i'm sorry to ask you this question because i didn't well understand your story because my english is not that good so i was confused, did what i understand is correct ??I left my existing Mitel system in place, and expanded it with more Mitel equipment for my new location.
My plan to just adding more phones to the new location and not installing a new PBX was not the way we installed because the pricing savings of installing a new PBX and bridging it into the old system was less expensive.
