Asterisk conflict with phone provider

Hello there!

I’m encountering an obvious issue with asterisk. Here is my configuration:

PSTN lines > Grandstream FXO Gateway > Asterisk server > IP phones

Each extension of my asterisk system is linked to a specific pstn line (thus, a specific number and a specific port on my gateway).

So, let’s say someone is on the phone with an external number. i.e: Extension #75 is using port #1 on the gateway which is connected to the 3102 1275 number. So this number is now busy, and when someone tries to call it, it rings as busy, and the caller is transferred to the pstn provider voicemail. Which is bad! Is there anyway I can get the call to my asterisk system and then handle it from there. I want to be able to allow calls in, even when a line is busy and let asterisk handle it. I don’t want my phone provider to do anything!

Is there anyway of doing that? How are people doing at the moment? Obviously, they don’t use two voicemails… Everyone should have this issue, and I’m sure someone found a way around it…!?

Please help, thanks a lot!

And I’m not trying to get multiple calls on a single line, I’m just looking for a solution to go around the problem…

Your issue is not an asterisk one. You want to get multiple calls for a single number, which for a standard phone line is only possible if you use call waiting. If you are on a Centrex system you can ask your carrier to set up hunt groups/roll over which basically says if this line is busy, roll over to this one etc… The problem with this is when the 2nd call into the 1st number rings (Into the 2nd hunt line), you don’t know what number was originally dialed which may or may not be an issue for you.
If you have more than 8 or so lines, you may want to look into a PRI line which would resolve this issue.

I like your hunt groups/roll over solution, that’s exactly what I need. But loosing the originally dialed number will be an issue for me as I want to keep direct in dial. So in asterisk, I need to retrieve the originally dialed number and route the call to the corresponding extension…

Any idea how I could do that?

Without switching technologies I don’t think you can do what you want to do. Either change to a PRI or strictly SIP setup and get the functionality you want. Or stay with the analog lines and deal with the limitations.

I see…
Thanks a lot for your help anyway!