VOIP.MS to Legacy PBX
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I have a customer that wants to keep their legacy pbx but use SIP for phone service. Currently they use some NEC piece of crap that has 4 pots lines connected to it. My plan was to purchase either a Grandstream FXS or 2 Cisco SPA112 to convert the SIP trunks to analog lines. I am not completely sure what the correct path would be for handling incoming DIDs. Would I purchase 4 DIDs and create a hunt group using the failover options in VOIP.MS or can I accomplish everything with a single DID? If I went with only 1 DID then I am not sure how I would configure the hunt group on the ATA. If I go with 4 DIDs can I limit each DID to 1 channel?
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@syko24 Why does the customer want to keep the legacy pbx?
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@eddiejennings said in VOIP.MS to Legacy PBX:
@syko24 Why does the customer want to keep the legacy pbx?
They don't want to spend the money. I would like them to upgrade but at this point they just want the benefit of saving on their phone bill.
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@syko24 said in VOIP.MS to Legacy PBX:
@eddiejennings said in VOIP.MS to Legacy PBX:
@syko24 Why does the customer want to keep the legacy pbx?
They don't want to spend the money. I would like them to upgrade but at this point they just want the benefit of saving on their phone bill.
To make sure I understand, they have analog phone lines connecting the PBX to every phone at the location, but they want to connect to the PSTN via SIP trunking, correct?
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@eddiejennings said in VOIP.MS to Legacy PBX:
@syko24 said in VOIP.MS to Legacy PBX:
@eddiejennings said in VOIP.MS to Legacy PBX:
@syko24 Why does the customer want to keep the legacy pbx?
They don't want to spend the money. I would like them to upgrade but at this point they just want the benefit of saving on their phone bill.
To make sure I understand, they have analog phone lines connecting the PBX to every phone at the location, but they want to connect to the PSTN via SIP trunking, correct?
Correct, replacing AT&T with VOIP.MS.
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I wonder what the costs are for supporting and maintaining that PBX. Perhaps you can find some leverage with that to just move to VOIP completely.
Another question would be do they need 4 DIDs? If not, then that would help make the solution less complex.
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The more I think about it, I wonder if the ATA has a maximum channel setting. If I go with 4 DIDs and set each account on the ATA to 1 channel, that would cause a busy signal when there is a second call. That would cause the failover option on VOIP.MS to kick in and then I could send the call to the next DID.
Guess I am not sure if this is an actual option or if there is a better method here.
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@eddiejennings said in VOIP.MS to Legacy PBX:
I wonder what the costs are for supporting and maintaining that PBX. Perhaps you can find some leverage with that to just move to VOIP completely.
Another question would be do they need 4 DIDs? If not, then that would help make the solution less complex.
Where I am getting a little confused is if I only had 1 DID and an incoming call happens, which ports on the ATA are going to be active? I would think all 4 would be unless I had some kind of separation. That is why I thought 4 DIDs. One for each port on the ATA.
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@syko24 said in VOIP.MS to Legacy PBX:
@eddiejennings said in VOIP.MS to Legacy PBX:
I wonder what the costs are for supporting and maintaining that PBX. Perhaps you can find some leverage with that to just move to VOIP completely.
Another question would be do they need 4 DIDs? If not, then that would help make the solution less complex.
Where I am getting a little confused is if I only had 1 DID and an incoming call happens, which ports on the ATA are going to be active? I would think all 4 would be unless I had some kind of separation. That is why I thought 4 DIDs. One for each port on the ATA.
Without reading some documentation on the ATA device, I can't answer you since I've never used those devices before. I am curious though, how many endpoints are connected to the PBX?
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I have a SPA112 laying around somewhere. I need to play around with it and see how it handles 2 calls when registered with a single DID. Not sure if both ports are active or if it checks port 1 first and then goes to port 2 if 1 is in use. Maybe there is a hunt option on the ATA.
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@eddiejennings said in VOIP.MS to Legacy PBX:
@syko24 said in VOIP.MS to Legacy PBX:
@eddiejennings said in VOIP.MS to Legacy PBX:
I wonder what the costs are for supporting and maintaining that PBX. Perhaps you can find some leverage with that to just move to VOIP completely.
Another question would be do they need 4 DIDs? If not, then that would help make the solution less complex.
Where I am getting a little confused is if I only had 1 DID and an incoming call happens, which ports on the ATA are going to be active? I would think all 4 would be unless I had some kind of separation. That is why I thought 4 DIDs. One for each port on the ATA.
Without reading some documentation on the ATA device, I can't answer you since I've never used those devices before. I am curious though, how many endpoints are connected to the PBX?
I believe 12 phones
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@syko24 said in VOIP.MS to Legacy PBX:
@eddiejennings said in VOIP.MS to Legacy PBX:
@syko24 said in VOIP.MS to Legacy PBX:
@eddiejennings said in VOIP.MS to Legacy PBX:
I wonder what the costs are for supporting and maintaining that PBX. Perhaps you can find some leverage with that to just move to VOIP completely.
Another question would be do they need 4 DIDs? If not, then that would help make the solution less complex.
Where I am getting a little confused is if I only had 1 DID and an incoming call happens, which ports on the ATA are going to be active? I would think all 4 would be unless I had some kind of separation. That is why I thought 4 DIDs. One for each port on the ATA.
Without reading some documentation on the ATA device, I can't answer you since I've never used those devices before. I am curious though, how many endpoints are connected to the PBX?
I believe 12 phones
The ring group would have to happen on the PBX.
I am going to be on the road all morning so I can’t get into detail right not.
The PBX expense is like $600 in labor assuming FreePBX.
Just have them buy cheap used IP phones off eBay and setup free PBX. You can find them for less than $50 if you dig.
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@jaredbusch said in VOIP.MS to Legacy PBX:
@syko24 said in VOIP.MS to Legacy PBX:
@eddiejennings said in VOIP.MS to Legacy PBX:
@syko24 said in VOIP.MS to Legacy PBX:
@eddiejennings said in VOIP.MS to Legacy PBX:
I wonder what the costs are for supporting and maintaining that PBX. Perhaps you can find some leverage with that to just move to VOIP completely.
Another question would be do they need 4 DIDs? If not, then that would help make the solution less complex.
Where I am getting a little confused is if I only had 1 DID and an incoming call happens, which ports on the ATA are going to be active? I would think all 4 would be unless I had some kind of separation. That is why I thought 4 DIDs. One for each port on the ATA.
Without reading some documentation on the ATA device, I can't answer you since I've never used those devices before. I am curious though, how many endpoints are connected to the PBX?
I believe 12 phones
The ring group would have to happen on the PBX.
I am going to be on the road all morning so I can’t get into detail right not.
The PBX expense is like $600 in labor assuming FreePBX.
Just have them buy cheap used IP phones off eBay and setup free PBX. You can find them for less than $50 if you dig.
@JaredBusch I totally agree with what you are saying. I would choose to put in FreePBX and all new or used phones. I have setup a FreePBX for this customer in the past. They have multiple businesses and are aware of the benefits and features that it offers. For this particular business they don't want to spend the money on replacing all the equipment. I get the impression that they are thinking of selling in the not so distant future and don't want to spend money where they feel they don't need to.
My goal is to replace the pots lines with sip trunks. The only solution I know of would be to put an ATA/FXS device in front of their current PBX. I just don't know how to create a hunt group similar to how it is setup with the pots lines.
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I feel like there should be a hunt feature on the ATA itself that would allow me to start with port1 and then move on to the next if port1 is occupied.
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@jaredbusch said in VOIP.MS to Legacy PBX:
Voip.ms can be the PBX.
Just not the easiest to use.
I've played around with this as well. The configuration is just not as smooth as working with FreePBX.
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The company I work for has done SIP trunks to legacy PBX systems. Typically we do this with either their Total Access 900e devices or their Netvanta equivalents. If your PBX supports PRI, this is the basic config document from AdTran:
https://supportforums.adtran.com/docs/DOC-3404 -
@syko24 said in VOIP.MS to Legacy PBX:
I feel like there should be a hunt feature on the ATA itself that would allow me to start with port1 and then move on to the next if port1 is occupied.
How would this work when one ATA has no idea about the next one?
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@jaredbusch said in VOIP.MS to Legacy PBX:
@syko24 said in VOIP.MS to Legacy PBX:
I feel like there should be a hunt feature on the ATA itself that would allow me to start with port1 and then move on to the next if port1 is occupied.How would this work when one ATA has no idea about the next one?
Use a Grandstream HT701 with 4 sub accounts on VoIP.ms.
Each sub account will be an extension in the VoIP.ms ring group or hunt group.
Simplest and easiest thing to do.
