All my experience with Asterisk has come with SIP VOIP providers and digital PRI so I’m a bit of a newbie in the analog world.
I have a client that has 4 analog lines from Bell Canada. They want to install Asterisk and a TDM400P card. Will this support direct inward dialing the same way as SIP or PRI? Can Asterisk determine the number that is being dialed so that I can route it to the correct internal extension with analog lines?
Well, my previous post isn’t correct, seems there are methods to have dids on analog lines, I just did a search with Google and found something but using a digital line is much simpler.
An analog line is only ever associated with a single number(RFCs not included). As such, you simply need to create a different context for each incomming analog line and have that context ring the appropriate person.
So, as each channel (otherwise known as a port on the card) receives a call, it will ring the appropriate person. Then you can have each person ring a different internal context (or base it on ${EXTEN}) to make sure that they use their individual port on the TDM card.
That said, yes, a digital BRI/PRI implementation is far easier.
The question I’ve been asked by Bell is whether Asterisk supports “Analog DID”. This client is pretty small and it doesn’t really make much sense to move up to PRI at this time.
To me, after doing some reading on the web, “analog DID” sounds like a setup where the phone company give me 4 analog lines but they would connect to FXS ports instead of FXO…
No you would still connect them via FXO. Analog DID does exist its usually just almost as expensive as setting up an ISDN or partial T1 line so its not very common.
I do not know if Asterisk will support it but if you have (or plan on buying) a Digium card it would be worth sending an email to support@digium.com.
It may be worthwhile getting some quotes on partial T1’s as well - we have a 8 channel partial T1 for our fax server with DID’s and its less than we were paying for 8 analog lines.
group=1
signaling=xxxx
param=xxxx
param=xxxx
context=DID_1
channel=1
context=DID_2
channel=2
context=DID_3
channel=3
…and so on and so forth.
Read the zapata.conf example file or voip-info. Each time you define a channel it takes the variables above it for it’s config. If you change a variable inbetween channel declarations, then the most recent param entry is used for the next declared channel. That’s how you have mulitple analog DIDs.
Thats not the same, that directing based on what line was used NOT what number was dialed. DID (whether analog or digital) allows you to have more numbers than lines and the original poster wants to route based on the number dialed not the line used.