Another newbie question

[clifford.gormley]

Ok, this UC540 is pretty cool.

I created my first Prompt, used x397 to record it, use x398 to listen and react to it, and it all works pretty well.

Here is my next question, which has already been addressed on a previous discussion, but I am starting it new here for simplicity.

Now for the newbie question.

Can I use just one PSTN line (I think 'PSTN' is the correct term.  I am using the term 'PSTN' to refer to the phone line that comes into my home office which I am using to demo my UC540 right now) to allow external users to join a conference call?  I have set up MeetMe, and see how that will work with internal users, but am not sure how to set it up to work with external users.

My common sense tells me that this will NOT work for one external PSTN line, because my thinking is that after the first user calls in, any subsequent other users will get a busy signal.

I think David told me that I may needed to get a SIP provider.   I have heard of SIP, but am not sure what it does or what it provides, nor do I know who is a good SIP provider in the Sacramento area.

Thanks from UC-Newbie.

[SteveOrfanos]

Point the FXO port to the meet me pilot number and when they call in they will join the conference. Internally seize dial tone and press meet me soft key. Do the later first for testing.

[clifford.gormley]

I will try this tonight.

Thanks!

[David Trad]

Hi Cliff,

I think David told me that I may needed to get a SIP provider.   I have 
heard of SIP, but am not sure what it does or what it provides, nor do I
 know who is a good SIP provider in the Sacramento area.

I'm not from the states so I cant help you with providers, however just think of SIP as the same as your PSTN line, with the exception that your calls are coming in and going out over IP and in the majority of cases, over the public Internet. I liken SIP to a digital form of your analogue line, I guess that is the closest to laymen's terms I can get

The suggestions above is a brilliant one, however in doing this it would mean ever single call that comes in will be directed to the Meet-Me conference, and if you have not logged in and setup the rooms, all calls into the system will get rejected, so my idea of using an AA is your safest bet.

If you Google SIP providers in Sacramento, I am sure something will pop up

This might help you or confuse you, but you could look at this site, they list some of the more common/mainstream providers (I think)

http://siptrunkingcomparison.com/ia/sac-city.html

Cheers,

David.