Lister Account wrote:> I'm a newbie here. I have a general question that can help drive how
> exactly I'm going to get started.
>
> Say I have a single inbound number (1-800-my-number).
>
> When a call is connected on that number, and another call comes in,
> will asterisk answer it, will a call waiting signal be triggered, or
> will a busy signal occur?
>
Asterisk will answer the call, as long as there are available lines.
> I'm thinking about this for several reasons.
>
> 1) for pbx, I want to make sure all inbound calls are picked up.
> 2) for conferencing, I want to be able to just give someone a single
> number to call every time (for regularly scheduled meetings).
>
> As I understand it with traditional telco equipment, you'd have a t1
> or a group of lines, and the inbound call would get answered as long
> as there was an open line available. (it would rotate through the
> available lines)
>
> If this is still the case with voip in the asterisk world, does this
> mean I'd have to have a set of inbound lines? And if so, could you
> point me to the wiki and call me names, or send a link to
> documentation about how to set up this rollover behavior?
What you point out would be correct. Plug the T1 line into one of
Digium's T1 cards, configure your dialplan appropriately and you're all
set.
>
> The way I'm planning on things is to set up my server in a data center
> with high bandwidth availability (all voip, no pots or telco t1).
> Inbound calls will be for voicemail, routed to my landline or cell
> phone, or conference calls.
>
> This might sound like a stupid question, but I'm wondering if setting
> up asterisk is the way to go, or if I should use a virtual pbx
> service/conf. call provider. I prefer asterisk because I'd have
> control over everything (and I'm a geek who is addicted to OSS). If I
> have to maintain 10 or more dial-in numbers at a cost of
> $10/month/line, it might not make sense though. (figuring max conf.
> call of 6 visitors, and 2 inbound calls that might be routed to land
> lines)
All these can be done with *, just depends on how you've got your
dialplan configured.
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>