Small office phone setup
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@JaredBusch said:
@coliver said:
Agreed, although remember that every call is at this point is an external call. So it would count twice as far as bandwidth is concerned.
Every call is external, but with only 10 phones how many in house calls are happening? Those are the only ones that take double bandwidth.
Exactly. With that few of people, the chances of lots of intra-office calls taking place is slim.
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@JaredBusch said:
Every call is external, but with only 10 phones how many in house calls are happening? Those are the only ones that take double bandwidth.
Additionally re-invite can be enabled to let the RTP streams talk to each other directly.
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@JaredBusch said:
@coliver said:
Agreed, although remember that every call is at this point is an external call. So it would count twice as far as bandwidth is concerned.
Every call is external, but with only 10 phones how many in house calls are happening? Those are the only ones that take double bandwidth.
Good point, just thought it would be something to be made aware of.
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@coliver said:
Good point, just thought it would be something to be made aware of.
Also, calculating calls on 100kb per call means you have at most 10 active calls * 100 kbps = 1 mbps with QoS on your router, there should not be any problems.
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@JaredBusch said:
@coliver said:
Good point, just thought it would be something to be made aware of.
Also, calculating calls on 100kb per call means you have at most 10 active calls * 100 kbps = 1 mbps with QoS on your router, there should not be any problems.
Considering he's only got 2 or 3 phone numbers, that shouldn't be an issue.
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OK Coliver Likes FreePBX.. what others are worth looking at?
Also looking to the future - my main location is looking to replace it's phone system next year.
Currently it's two Intertel systems bridged together, one digital one VOIP.We have 40 VOIP handsets and 65 digital handsets.
We have a T1 (23 trunks), I'd guess we have more than half busy at any one time, of course we could bust to the whole 23, but I'm not sure we ever have.We probably have 10-15 intraoffice calls going at a time.
Assuming I want both systems to be the same - does this change the desire to use FreePBX?
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@ajstringham said:
Considering he's only got 2 or 3 phone numbers, that shouldn't be an issue.
He has 10 phones. This has nothing to do with external calls.
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@Dashrender said:
OK Coliver Likes FreePBX.. what others are worth looking at?
Also looking to the future - my main location is looking to replace it's phone system next year.
Currently it's two Intertel systems bridged together, one digital one VOIP.We have 40 VOIP handsets and 65 digital handsets.
We have a T1 (23 trunks), I'd guess we have more than half busy at any one time, of course we could bust to the whole 23, but I'm not sure we ever have.We probably have 10-15 intraoffice calls going at a time.
Assuming I want both systems to be the same - does this change the desire to use FreePBX?
Nope. FreePBX will scale quite well for that. In reality, you could use this one office as a test environment and then eventually incorporate their server into your main FreePBX server if you roll that out for the main site.
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@JaredBusch said:
@ajstringham said:
Considering he's only got 2 or 3 phone numbers, that shouldn't be an issue.
He has 10 phones. This has nothing to do with external calls.
I'm mixing up stuff in my head. He wont' have that many concurrent calls though.
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@JaredBusch said:
@ajstringham said:
Considering he's only got 2 or 3 phone numbers, that shouldn't be an issue.
He has 10 phones. This has nothing to do with external calls.
But I only have 4 employees - so unless a patient picks up a phone I can't see us ever having more than 4 off hook at once.
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@Dashrender said:
@JaredBusch said:
@ajstringham said:
Considering he's only got 2 or 3 phone numbers, that shouldn't be an issue.
He has 10 phones. This has nothing to do with external calls.
But I only have 4 employees - so unless a patient picks up a phone I can't see us ever having more than 4 off hook at once.
Ah ok. So likely never more than 2 calls at a time, on average.
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@ajstringham said:
@Dashrender said:
But I only have 4 employees - so unless a patient picks up a phone I can't see us ever having more than 4 off hook at once.
Ah ok. So likely never more than 2 calls at a time, on average.
Yep. They don't intra office call right now - they just yell down the hallway to pick up the phone.
They'd only have 3 if the fax was in use at the same time as the the phone lines. But having two phones busy would be a near constant thing.
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Something else to mention - my carrier currently forwards all calls that would overflow the two lines back to my main office. Can SIP trunks do that?
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@Dashrender said:
Something else to mention - my carrier currently forwards all calls that would overflow the two lines back to my main office. Can SIP trunks do that?
Yeah, that's doable.
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@ajstringham said:
I believe the average bandwidth for calls is 100kbps both up and down for each concurrent call.
100Kb/s is just above the theoretical maximum, not average. We use it as a buffered number to account for all possible overhead. Average is well below 80Kb/s for uncompressed audio and as low as like 15Kb/s for some compressed options. Using 100Kb/s gives you more than enough safety margin and is really easy to calculate. With 100Kb/s you can do high def audio even.
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@ajstringham said:
@Dashrender said:
Something else to mention - my carrier currently forwards all calls that would overflow the two lines back to my main office. Can SIP trunks do that?
Yeah, that's doable.
Actually no, a SIP trunk does not do that. A provider may have an add on service, but a trunk does not do that.
All the providers I have worked with only send calls to the fail over number when the trunk is unreachable. Not when the trunk has reach a concurrent call limit. There may be a provider that does it, but I do not know of one.
The issue here is your call flow not the SIP trunk. You only have 2 phones available to answer a call, but what about call waiting or having multiple lines programmed on the phone to allow more than one inbound call at a time?
You need to think differently. Using a trunk from VoIP.ms has no realistic limit to concurrent calls. You send the calls in to a ring group and have the fail for that ring group be to send the call to your main office.
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@ajstringham said:
The latency difference really isn't noticeable. NTG hosts their PBX out of Toronto and I used it both from Upstate NY and Dallas and didn't have issues either time.
We did. More recently we moved it to Chicago.
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@scottalanmiller said:
We did. More recently we moved it to Chicago.
FYI, VoIP.ms must be in the same facility as RackSpace because my trunks to their Chicago servers have 1ms response times.
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@JaredBusch said:
@ajstringham said:
@Dashrender said:
Something else to mention - my carrier currently forwards all calls that would overflow the two lines back to my main office. Can SIP trunks do that?
Yeah, that's doable.
Actually no, a SIP trunk does not do that. A provider may have an add on service, but a trunk does not do that.
All the providers I have worked with only send calls to the fail over number when the trunk is unreachable. Not when the trunk has reach a concurrent call limit. There may be a provider that does it, but I do not know of one.
The issue here is your call flow not the SIP trunk. You only have 2 phones available to answer a call, but what about call waiting or having multiple lines programmed on the phone to allow more than one inbound call at a time?
You need to think differently. Using a trunk from VoIP.ms has no realistic limit to concurrent calls. You send the calls in to a ring group and have the fail for that ring group be to send the call to your main office.
As long as I can 'send these calls to a traditional LEC' that's fine. Can I?
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@JaredBusch said:
@coliver said:
Good point, just thought it would be something to be made aware of.
Also, calculating calls on 100kb per call means you have at most 10 active calls * 100 kbps = 1 mbps with QoS on your router, there should not be any problems.
And that is the maximum with every line engaged, all at once, all talking. Any silence suppression or compression brings that number down. You'd never be able to hit 1Mb/s (and that is rounded up again on top of the buffer already built in) even in a test purposefully trying to hit that. A more reasonable "you'll never hit it limit" is more like 800Kb/s. And there is a very good chance that 600Kb/s will never actually be hit even after years of use.
