DISQUS

WinnP · 1 month ago

Great tutorial/summary. However, despite having followed it I am still not able to get any incoming or outgoing calls. Calls from one computer to another (using X-Lite) within my network are fine, and the PBX seems to be running fine. Google Voice calls to the gizmo5 client also work fine. However, it seems like the gizmo5 forwarding to SIP isn't working.

I have a static IP, so didn't have to mess with dynamic DNS stuff. I forwarded port 5060 for TCP and UDP on my router... what's the best way to test whether that is working? Know of any other tricks to troubleshoot this problem?

I know you're not customer support for any of these programs or anything, but I figure it wouldn't hurt to ask.

GeekLad · 1 month ago

I would check the Asterisk call logs to see if the calls are coming in at
all. Just click on the Reports tab at the top of the FreePBX administration
window. If Gizmo5 is forwarding properly, you should see entries being
added to the call log. If not, then the problem is either with the router
port forwarding or with the Gizmo5 forwarding.

I've messed it up a few times on the Gizmo5 forwarding, because I would
click the SIP radio button but forgot to click on the Forwarding All Calls
radio button.

WinnP · 1 month ago

Thanks... I assume you mean the call logs I can access on /admin/reports.php via the web interface? I just tried two calls (one incoming, one outgoing) and they both show up as "Answered" on that log. However, the incoming call rang for awhile then went to my GV voicemail and the outgoing call just gave me the hold music for 50 seconds then said "Goodbye."

GeekLad · 1 month ago

Something else you can try, is to go into the General Settings screen and
setting Allow Anonymous Inbound SIP Calls? to yes. This could potentially
have security implications (for example, if you have an IVR). I'm somewhat
of an Asterisk noobie myself, so use this feature with caution.

It may be a good way to test things out to see if the box is indeed
receiving the calls. Ultimately, I think you want to get things set up
properly with the trunks, routes, and extensions so that you are able to
properly receive calls without having to set that option. I'm still a bit
fuzzy on setting these things up the right way, which is why I turned to
Orgasmatron since he already had all the trunks, routes, and extensions all
set up.

I'm still puzzled as to why it isn't working for you though. Everything
should be all set w/ the trunks, routes, and extensions so it shouldn't be
necessary to open up the system to anonymous inbound SIP calls.

WinnP · 1 month ago

Alright, thanks for the help -- I think I'm going to try out the PIAF forums, perhaps one of their gurus will know what I've done wrong. I'm sure I just messed up a single keystroke somewhere along the way.

GeekLad · 1 month ago

Yeah, the PBX in a Flash forum may be of better help than I have been.
Sorry I wasn't able to help you out. If you do finally find a fix in the
PIAF forums, please feel free to post the solution here. Best of luck!

WinnP · 1 month ago

I got my IPKall DID after only waiting about 14 hours (I'd heard sometimes it takes days or weeks), so I just switched to that. Now it works fine. Have no clue what was wrong with Gizmo5!

GeekLad · 1 month ago

Very odd... Yeah, the IPkall wait period was why I went about using Gizmo.
Well, I'm glad you got it working!

vigneshwar · 1 week ago

What is the difference between this method and the one using googlevoice, gizmo and linksys PAPA2T ATA device. Is the outbount call time unlimited on this method?

I read about this on http://googlevoiceandgizmo.blogspot.com/

GeekLad · 1 week ago

Yes. There's no limitation to the amount of time that can be spent on the
outbound call.