TIPS Lots of General Questions

SipnEat

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Hey All -

I have worked with PIAF for a month now, and I have a pretty good handle on things in general, however now that I have a system in place (waiting on my Sangoma B600DE card) Im starting to get nervous.. :)

So I have a number of questions, I HOPE its not too much and the flames start coming my way. I'm new to the forum so please be nice.. :rolleyes:

My plan:
The customer is using a SwitchBox that is EOL, and needs to upgrade. As I support all their systems they came to me. Given the replacement costs for the new TalkSwitch (Fortinet) is ~$2000.00, with a smaller number of options I think this is a better way to go. What I want to do in the first round is create a direct replacement of the TalkSwitch to the PIAF system using the POTS, then add VoIP trunks and move that way. Bandwidth is good, and the Firewall/Switch environment is pretty solid and supports shaping.

Current system (being replaced):
1 x TalkSwitch
4 x POTS
1 x 4 set Cordless phone system 2 lines

Proposed Replacement system:
1 x MSI C847IS-P33 (Great littel ITX Board BTW)
2 x 2 DDR3 RAM
1 x 500GB SATA drive
1 x Sangoma B600DE (waiting for arrival)
PIAF - Asterisk 11, FreePBX 2.11
1 x Sipstation trunk with one (1) DID (so far)
10 x Phones (not selected yet)

The questions:

Q1. Will the above system from a general sense, like using the current POTS lines, be a direct replacement? i.e. I can move the POTS over and get them up and running with the new system with the same functions. Not worries about IVR stuff.. that I know is good.

Q2. The overall idea is to get the system up with PIAF with the SAME functions, basically a direct replacement, then add a VoIP trunk (already up and working) and slowly move to a VoIP keeping a single POTS for the main line. My understanding here is that we either need one (1) trunk per line or a multi-channel type setup. Am I correct in this?

Q3. I see phones out there that state "One sip account 2 line" and "4 line phone", I have TRIED to sort out what the difference is, but still a bit confused. The customer needs to be able to answer a call, put that call on hold, press a second "line" and make an outgoing call while that call is on hold. As noted I'm using a single trunk with a single DID so right now, its one.. but moving to the other line how do the phones affect this?

Well not as bad though Im sure I will have more as I move forward!

Can't thank everyone enough for all the replies already!

Brian
 

tm1000

Schmoozecom INC/FreePBX
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1. I don't know anything about Talkswitch. So I don't know the features. But the feature set of Asterisk and FreePBX rivals that of Cisco Call Manager and NEC

2. This is up to the VoIP provider. 1 trunk means 1 connection (for both inbound and outbound). It's a data pipe so technically the provider could allow you to have unlimited inbound and outbound. As I said it's up to them. Best advice is when dealing with 'trunks' don't think of PSTN/POTS it does NOT work the same.

3. All SIP phones can do multiple inbound and outbound and on hold. '2 line' '4 line' is in reference to how many accounts you can register to the phone. Eg. extension 101,102,103 can all be on one phone.
 
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POTS lines can be an issue. If they are long term advertised numbers I would not port the main number but would forward it. At least for now. If there is an issue you can quickly forward it somewhere else. The local telco can set the number of "paths" that the forwarded number can have so your not limited to one.

I realize you didn't say that your going to port all the POTS lines but what you will find is that the call quality using the interface card is never quite as good as pure SIP trunks. The other thing is there is a slight delay in ringing because the interface card needs to wait for the caller-id info to be sent after the first ring.

Having a POTS line is good for a fax in case that is an issue.

Some people use the forwarded POTS (to a SIP trunk) for incoming and then do all outgoing on a SIP trunk rather than using the POTS lines.

The phones are really a critical choice in terms of acceptance and usability so buy one or two you can return and test them out in the same way that they will be actually be used.
 

leemason

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SIPSTATION does not limit the number of inbound/outbound calls. When you get more confident with the system I see no reason to keep any POTS lines at all. In the longer term it's best to migrate all numbers to a SIP service. In the intervening times as suggested above you could just forward POTS numbers to the SIP service.

In terms of phones just decide what you really need to do with them, what you want in terms of facilities programmed onto keys, and make sure that you get phones that support the functions needed AND are well supported by End Point Manager (EPM) so that you can control your phones via config on the PIAF server. Have a look at phones from Yealink, Aastra, Snom, or Digium.

Do you need wireless/DECT phones? There are not too many of these support by EPM. They are not supported by EPM but I quite like the Gigaset range.
 

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