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I checked again and that's the exact URL syntax I use on my SPA phones. Maybe try it without http auth and see if it provisions properly just so you know that part of it works. Also try putting the URL directly into your browser and make sure it works ie: http://FUSION PBX...

Frank maybe can convince Mark to move his operation ;)

Maybe try creating a ring group with your cell number as the only destination instead of a dummy extension?

Have G729 as the first codec in your Sofia Internal profile and then only have G729 provisioned to the phone, or at least make it preferred on the phone. Honestly I've had better luck with GSM in low bandwidth situations. As an aside, a compressed codec isn't going to do anything to fix jitter...

Hi, This is what should be in your profile rule after provisioning is complete: [--uid myUser --pwd myPass]http://mydomain.com/app/provision/?mac=$MA This is what you would put in a browser to provision the phone: http://IP ADDRESS OF...

Can you not just have five fake extensions that just ring without having them registered to a phone? As long as you don't terminate somewhere wouldn't they just ring forever?

The easy answer is different ring tones for each IVR option with a call display prefix, ie prefix the call display with "Sales". As far as I know (I'm a Fusion PBX noob) you can set the ring tone by adding an alert-info statement to the incoming call as per here...

OK Not sure why i keep working on the Cisco/Linksys SPA templates when the phones are EOL. I guess its because these where my go to phones for a long time. Anyway, there's a bit of code in the Cisco templates that should be in the Linksys templates if you want the provisioning button to work...

Thanks for this. I fixed some issues with provisioning templates by guessing, but now I know whats going on.

I'm really sad to see this as well. Such great workhorse phones with quality Sipura DSPs inside them, and provisioning was awesome. Customers definitely loved the Cisco branding, Yaelink not so much even though they're a great phone as well.

Ok a few simple things to check for. First yes most of the Caribbean falls in the North American dial plan so anything not filtered with 011x is going to make it through. Some of those rate centers can be expensive. It's worth your time to put together a Caribbean route and deal with it...

This kind of a dumb work around, but depending on how many IP's you have to spare and how many domains you want to set up maybe just make the domain name an IP address. For example assign a bunch of IP aliases to the box, have sofia listen on them and then name your domains accordingly. Not sure...

Just taking a wild guess, but maybe throw the domain in this line: <processNodeName>sip.xyz</processNodeName>

I'm looking for a billing solution for selling aggregate channels. So I want to do sell a local/or aggregate on net sip channel. Obvioulsy anything off net or international would still be billed by the minute. I'm looking at ASTPP right now, but it's kind of clunky for this purpose. Does...

In your dial plan settings anywhere that you want to dial out immediately append the it with S0 (es zero) For example 10 digit prefixed with a 1 would be: 1xxx[2-9]xxxxxxS0 without: xxx[2-9]xxxxxxS0 I think the default dial plan has ten digit dialing with no S0 and therefore you have to wait...

Awesome! Thanks!

Sorry if I'm being dense, but I don't see it in outbound routes. Is it a condition variable?

Is there a way to limit simultaneous channels on a gateway or an outbound route?

Thanks I figured it out after a while. Just saying maybe a newb forum would be cool.

Thats great! Glad it was something simple. PS, buying the pro version of ping plotter is the best money you'll ever spend.