The company I work for is planning to replace its POTS lines with a PRI
line. The plan is to buy a block of phone numbers, some of which will
be used for phone, and the rest for fax.
We are now using a Norstar modular ICS switch (0X32), which routes
incoming phone calls according to the extension the caller dials. I
understand that a call coming in over a PRI line carries the called
number and that this feature is called DID. So, once we get PRI, we will
route calls based on DID. We would like to replace the Norstar switch
with Asterisk.
I have several questions:
Must we convert the company’s internal telephone infrastructure from
POTS to IP, or can we continue to use the old internal lines? (Using the
old lines, we won’t have to buy VoIP phones, thus saving us a lot money.
It would be even better if we could continue to use our Meridian phones.
But, see next question).
Our company has two separate buildings with a wireless IP connection
between them. Is there a way for Asterisk to route calls to the local
building using the old internal POTS lines, but to use VOIP to route
calls to the remote building?
Is there additional hardware we need to buy in order to allow Asterisk
to route a call based on DID, or is something like a Digium PRI card all
we need?
We would like to route certain dialed numbers to a Hylafax fax server
sitting behind (from the point of view of someone outside the company)
Asterisk. Would the DID information of a call be visible to Hylafax,
or would it be stripped off by Asterisk? (If we can use our old phone
lines instead of VOIP for the phone network, I suppose we will have to
signal Hylafax the called number using a method other than DID. Would
Asterisk need to do something like DID to DTMF translation in that case?)
I will try to help out with the issues I am familiar with…
No, you can use hardware from digium, along with Zaptel in order to use traditional pots handsets on an Asterisk box.
This is all possible through the dial plan (extensions.conf).
Everything you want here - apart from fax since i don’t know about that - is definately possible. Sounds like you need to find a suitable Asterisk consultant to fully scope and plan out the system. But Asterisk can easily handle what you want, and more.
[quote=“Berhanie”]we could even go
fully VoIP, if we can find a decent SIP or IAX service provider.
Very exciting stuff![/quote]
That is one of the hardest things to find…. I’m trying to do the same at the moment. Out of interest, where are you located?
Also, with regards to going fully VoIP, remember to keep one traditional line in case of power or internet failure. Devices such as the Sipura 3000 will automatically switch to the local PSTN connection if VoIP is down.