Newbie Needs Help - Simple Routing

Brand new PBX user - been using VOIPO with a Grandstream HT502 for 5 years, one of the lines is getting noisy. So, wanted to take this opportunity to enter the world of personal PBX. I have two “numbers” with VOIPO.

Here’s what I want to do - when someone dials our PSTN (VOIPO) number, I want them to hear a recording that says, “Press 1 to reach Clay, Press 2 to reach Jeannie” and then have their call routed to our cell phones, so we can carry just one device, and receive calls :“at home” while anywhere. I’ve got some other bells and whistles I want to add later, but want to crawl, walk run.

Just spent several hours trying to get FreePBX to even ANSWER my VOIPO line (instead of the HT502); with zero luck - figured I just try to answer and route first, and then put the IVR logic in place.

Can someone give me a basic idea of what I need to do, or point me at a “FreePBX for Dummies”?

I’m assuming something like (but, getting really lost in the details)

Set up an Inbound SIP trunk on one of the VOIPO numbers
Set up an outbound SIP trunk on the other VOIPO number
Set up inbound and outbound routes
Somehow cause the inbound call to be routed to the outbound trunk, which would dial my cell

This is a pretty good guide Sangoma Documentation

You can do that two ways.

  1. Setup Misc Destinations for each cellphone.
  2. Setup two (virtual) extensions with FollowMe to each cellphone.

Now, create an IVR which points 1&2 either to the Misc Destinations or Extentions w FM.
Then, point your Inbound route to the IVR.

Here is a thread on VOIPO with Asterisk, this fourm will be much more helpful with getting your VOIPO trunk online:

Set up an Inbound/Outbound SIP trunk:
https://wiki.freepbx.org/display/FPG/Trunks+Module+-+User+Guide

Inbound Routes:
https://wiki.freepbx.org/display/FPG/Inbound+Routes+Module

Outbound Routes:
https://wiki.freepbx.org/display/FPG/Outbound+Routes+Module

As PitzKey mentioned, the quickest way to do this is to make your cellphone a misc. destination, then make the inbound route point to your misc destination.
https://wiki.freepbx.org/display/FPG/Misc+Destinations

You could make a custom recording:
https://wiki.freepbx.org/display/FPG/System+Recordings+Module

Then make an IVR:
https://wiki.freepbx.org/display/FPG/IVR+Module

Have option 1 point to misc destination 1, option 2 point to misc 2.

Thanks, itzik and comtech!

Did the following:
set up inbound and outbound trunks
set up inbound and outbound routes
set up a misc destination for my cell and pointed the inbound route to that

rebooted (I’m running on a Raspberry Pi)

from the log, it looks like the trunks registered (although there are some timeout messages that bother me); but, when I call the number, I see nothing in the “full” log, and the call goes to the ATA
Here’s a snip from the “:full” log
[2018-08-29 21:55:14] VERBOSE[1062] loader.c: app_queue.so => (True Call Queueing)
[2018-08-29 21:55:14] NOTICE[1149] chan_sip.c: Peer ‘VOIPO-in’ is now Reachable. (1077ms / 2000ms)
[2018-08-29 21:55:14] VERBOSE[1062] asterisk.c: Asterisk Ready.
[2018-08-29 21:55:16] NOTICE[1149] chan_sip.c: Peer ‘VOIPO_OUT’ is now Lagged. (3093ms / 2000ms)
[2018-08-29 21:55:30] NOTICE[1149] chan_sip.c: Peer ‘VOIPO_OUT’ is now Reachable. (1096ms / 2000ms)
[2018-08-29 21:55:33] NOTICE[1149] chan_sip.c: – Registration for ‘[email protected]’ timed out, trying again (Attempt #3)
[2018-08-29 21:55:53] NOTICE[1149] chan_sip.c: – Registration for ‘[email protected]’ timed out, trying again (Attempt #4)
[2018-08-29 21:56:13] NOTICE[1149] chan_sip.c: – Registration for ‘[email protected]’ timed out, trying again (Attempt #5)
[2018-08-29 21:56:33] NOTICE[1149] chan_sip.c: – Registration for ‘[email protected]’ timed out, trying again (Attempt #6)
[2018-08-29 21:56:53] NOTICE[1149] chan_sip.c: – Registration for ‘[email protected]’ timed out, trying again (Attempt #7)
[2018-08-29 21:57:13] NOTICE[1149] chan_sip.c: – Registration for ‘[email protected]’ timed out, trying again (Attempt #9)
[2018-08-29 21:57:17] NOTICE[1149] chan_sip.c: Peer ‘VOIPO-in’ is now Lagged. (2090ms / 2000ms)
[2018-08-29 21:57:28] NOTICE[1149] chan_sip.c: Peer ‘VOIPO-in’ is now Reachable. (1164ms / 2000ms)

Those numbers are terrible. This will never work, and when it does, it will be unusable.