FYI - FOP2 License Revoke for moving to a new machine


(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!


#22

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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