Phone setup - what to assign to what?


#1

Hi

I am new to Asterisk and need help putting the pieces together. I have a Yealink T28P phone and I’m able to connect it properly to Asterisk. My question is a more about the fundamentals of assigning lines and extension to the phone.

With a traditional PBX, I would have buttons on the phone to indicate the POTS lines and when I wanted to dial out, I would choose the line that is available. Conversely, if a call came in, I would see which line it was and answer the call.

For the purpose of my question, lets assume I have 2 POTS lines, 2 SIP lines and 5 office extensions. How would I go about configuring the phone. Do I need to move away from the old thinking and just assume that Asterisk will handle the call based on a dial plan and all I need to do is pick up the phone and dial?

The Yealink has a facility to assign 6 SIP accounts, but they in turn relate to an extension number in Asterisk. How does that fit together. Do I assign extensions to the users (1001 - 1005) and separate extensions for the SIP lines (2001 and 2002)

I would obviously need a FSO/FSX card for the POTS lines but how do they get linked to the phone? Is that all done for me by Asterisk?

I hope my question make sense - just want someone to give me a quick overview of how to set up a basic system - or point me to a good article/post.

All help is greatly appreciated
Thanks


#2

Part of your problem is that your idea of a traditional PBX is not actually a traditional PBX and certainly not a PABX. Whilst it might actually be a PABX emulating one, what you actually describe is a key and lamp unit where every line is connected to every phone.

Asterisk is modeled more on the sort of PABX that would be used by an organisation with 50 to 1,000+ extensions.

It can, with suitable phones go someway to emulating what you describe (key word BLF - busy lamp field), but that should be considered advanced configuration, and you should analyse your deep requirements to see if you really need that behaviour. On traditional PABX systems, that sort of feature is often only emulated at the department level with individual managers replacing the outside lines.

You defnitely need to read Asterisk: The Future of Telephony, but I think you may also need to find a more general text on PABXes.

PBX - private branch exchange (possibly manual)
PABX - private automatic branch exchange.


#3

Hey dude,

I understand your question, as at some point I was asking the same questions :smile:

That’s not a must. If you’re using more budget combination of PBX and/or phones you won’t have any buttons and it would be pretty much like the phone you got at home. Looking at the phone you have gotten it seems it supports few SIP accounts, so one way would be to configure it with different accounts for external lines, as well as for external. However this is not necessary, as you could do just as well with one account and dialplan configuration using different patterns for internal and external dialing.

Now another convenient feature, especially for the case with just a few POTS lines, would be SLA, where you could see which lines are in use. I’m not quite sure how should you configure this on this phone, but I suppose you’re gonna have to read the phone manual and search around for Asterisk SLA (it’s in the Asterisk wiki for sure).

In asterisk you actually call the users, not the extensions numbers, so it’s up to how you want it. An example scenario in your case would be to have 1 account for the internal calls and 1 for access to the PSTN. This way you could select an outbound line just like with a traditional PBX. Of course this is not required, as, for example you could have just 1 account and press 9 for example to dial outside your PBX.

Stoyan


#4

AFAICT, SLA never worked on Asterisk. Part of the problem is the fact,in Asterisk, only one device can be registered per account. There are other PBX implementations where SLA actually works. It is probably worth checking the following links:

wiki.freeswitch.org/wiki/Shared_Line_Appearance
wiki.sipfoundry.org/display/sipX … Appearance


#5

I personally haven’t tried to get the SIP SLA to work, but I’ve had no problems with chan_sccp-b and SLA.