FYI - FOP2 License Revoke for moving to a new machine

(Greg Snover) #13

Really? Is that how business works? Learn something new every day! :wink:

Because I had about a 15-Minute conversation with Ryan about it a couple of months ago - they (Sangoma) had talked about it internally, but no approach (that he knew of) had been made.

I got back into FOP2 here: FOP2 WebRTC with Asterisk 18.3.0 and FreePBX15 working settings and a question - FreePBX / Tips and Tricks - FreePBX Community Forums because I was trying to use the WebRTC phone in FOP2 and because I was trying to get my customer base off Asterisk 13 - many of them had iSymphony, but it does not work with anything beyond 13 (and not well with that…).

I have quite a few installs and would like to see the app flourish - I am not talking about hijacking it away from Asternic - quite the opposite, I think they should make him a sweet offer and get it and fix it - as you can see from the previous post, to use the WebRTC phone in FOP2, you have to configure the extension and then basically it no longer works with a Hard Phone.

But UCP’s webphone works perfectly with FreePBX - so merge the two (FOP2 using UCP’s Phone) and ta-da! You have a perfect Web-Phone for FreePBX.

I am already paying Asternic for it - I would pay MORE for it if the WebRTC phone worked without having to booger the extension to make it work.

(Tom Ray) #14

Did you try setting the transport to auto so it would use the WebRTC transport when needed and UDP when needed?

(Greg Snover) #15

Yes - Auto doesn’t work:

(Simon Telephonics) #16

The Sangoma operator panel is called XactView. Older guide, not sure whether there’s a newer one:

(Greg Snover) #17

Looks like that is only for PBXact - I can find no way to look at/purchase it for FreePBX.

(Simon Telephonics) #18

Might be worth exploring PBXact and presenting some options to your customers including this operator console. Especially since your customers seem to be fond of the softphones Zulu/Connect (which are included in PBXact). Anyway, yes, looks like this is off-topic if you’re strictly FreePBX.

(TheJames) #19

There is no cost benefit for Sangoma to buy it. I am not sure they have the engineering resources either. Ultimately if you need/want a call panel they will happily sell you switchvox which they actively spend engineering money on.


So @GSnover did Nico come through and allow your revocation ? (if not, be be sneaky and call his Argentinian number and ask in your best Argentinian Spanish for help with your " 'Ouse Internet", know you tricked him, be prepared . . .)

Would you be prepared to pay about 10x his price for an alternative plus another monthly commit xN for your N users’ HTML5/webrtc based softphones ? I’m sure he would if he got some of that action.

(Greg Snover) #21

Sure did - That is why I did this post - that was from him.

Absolutely! I am happy to pay for things that I can easily sell - Just like Sangoma Connect here: Sangoma Connect is Awesome! - Commercial Modules / Sangoma Connect - FreePBX Community Forums

I am the dialtone provider for 90% of my customers - If I could have a similar deal on FOP2 that included a WebPhone that worked with FreePBX without modifying the extension, I would jump on it!

I can think of at least 10 customers off the top of my head that would take it tomorrow!


So to understand, whilst it works well , you can’t make a buck out of his Chrome FOP2 webrtc extension ?

(Greg Snover) #23

No, just the opposite - if you configure the extension for FOP2 to use it, you CAN’T use it at the same time with a Hard-Phone - That’s the problem.

I am not looking to make a buck on this at all - I am trying to provide a missing feature - a Web-Based SoftPhone with an interface for watching the other extensions + Chat and presence.

If for some reason it got bundled (working) into FreePBX, I would happily give it away.

All I am saying is that if I have to pay something to make this work the way it should, I would be happy to - and I think a lot of other people would too!

(Tom Ray) #25

Then have you tried the route FreePBX has gone, two endpoints that share the same state and presence? Just would need to Dial() both at once for incoming calls.

(Greg Snover) #26

Yes, I have used that method in several places and it works fine - but it would be way better if (like Connect and UCP) it was an invisible pair of the primary extension and all the plumbing was automatic.

(TheJames) #27

This by the way is a holdover from chan_sip and those who can’t just let it die… If you are using pjsip you can just bump max contacts.

(Tom Ray) #28

Except that’s not what is being talked about here. As the quote from Josh pointed out, you can’t have an IP phone and WebRTC on the same endpoint config because of the DTLS-SRTP requirements for WebRTC. So in order to have a WebRTC phone and a desk phone you need two endpoints which would need to be in the Dial() command for incoming calls to hit them both. Bumping max contacts doesn’t solve this issue.

(Greg Snover) #29

Yeah - it would be really great if you COULD set up the contacts with different settings - not only would it fix this problem, it would allow you to do E911 on the contacts with different Caller-ID’s - very handy to have the same extension at your office and house, but different 911’s for each phone - same extension, but different settings.

That would be such a huge help!

(Tom Ray) #30

Contacts have nothing to do with outbound calls. Contacts are there to tell the system where to send requests to, so in this case incoming calls to the devices. Contacts don’t need to exist for an outbound call to happen.

I set unique callerid names on each of the devices that share the same endpoint. I use that to determine which E9-1-1 details to send.

(David55) #31

I think there must be a terminology problem here. My understanding of the term outgoing, for a SIP, is a call in which Asterisk acts as the SIP client, e.g. a typical B leg, whether trunk or extension.

PJSIP contacts are only for such calls, although they can be bypassed by providing extra information in the dial string.

(Tom Ray) #32

That’s not always the case. I can initiate a call straight from Asterisk via different methods. So I send an call via PJSIP to an endpoint’s contacts that is just a single leg call.

But you are correct that you don’t even need an AOR or Contacts to send a call to a destination URI, the only thing you need is a valid Endpoint.

(system) closed #33

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