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SpinnerOffline



Joined: Sep 14, 2007
Posts: 1

Status: Offline
Posted: Sep 14, 2007 - 12:30 PM Reply with quote Back to top
Hi everyone,

Our office is getting set up with VoIP; we have a Sipgate number. The planned setup is as follows:

Two people use the phone for outgoing calls. When a call comes in, the secretary will most often answer it, but may have to transfer the call to the boss.

Is there a simple way of setting up such a system? What hardware will we need? I was thinking of ordering a Linksys PAP2T and connecting up two analogue phones. Does anyone know whether such a system would have the necessary features to facilitate our requests?

Thanks for your help.

Spinner
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polajooOffline



Joined: Sep 11, 2007
Posts: 19

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Posted: Sep 15, 2007 - 10:20 AM Reply with quote Back to top
The Linksys wouldn't let you transfer from one port to the other, that'd be the switch that did that for you.

Not sure what you'd need to use though, but you'd need *something* to 'run' your system.

Perhaps one of the other people can help you out. Sorry for being vague - I'm still learning too... Smile
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ianplainOffline
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Joined: Jul 05, 2004
Posts: 2844
Location: Bath UK
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Posted: Sep 15, 2007 - 11:09 AM Reply with quote Back to top
Hi

For such a small office I would look at the Voip "centrex" hosted PBX solutions. Most ITSPs offer such a service. This way you will have all the features of a PBX with out the electricity Bill Wink

Otherwise you would need to look a PBX for your office. There are many avalible and it depends on how deep you want to get into the guts depends on the way to go.

Ian
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alkeshrsOffline



Joined: Oct 17, 2007
Posts: 1

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Posted: Oct 17, 2007 - 11:58 AM Reply with quote Back to top
Dear Spinner,

You can have a Digitial Receptionist which will be played whenever the call comes to your office and then the caller can be directed to the desired destination depending upon his requirement.



Thanks.
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ysinc88Offline



Joined: Sep 27, 2007
Posts: 1

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Posted: Oct 29, 2007 - 08:26 PM Reply with quote Back to top
As the incoming is not a problem as said before. The tricky part would be to dial from line to line both reside on PAP.
I do not know a VoIP provider that does not provide FREE member to member as calls are routed inside the network the provider does not encour any termination charges.
You may write a simple dial plan that will connect you from one line to another with 3 digits or so.

something like <123:12345678> where the 123 is the other line and the 12345678 is your Auth ID on the network. so when dialing out it will translate 123 to 12345678.
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