FOP2 Community Forum

@Edi said:
do you have latest fop2

Yep, latest FOP2, latest FreePBX and latest versions of Chrome and Firefox.

Final update for today:
It would seem as though it DOES work! Just not in Google Chrome. For the heck of it I just tried it in Firefox and it worked first try.
What would cause this to work in Firefox but not Chrome?

Another minor update:
I got calls going through now by delecting a different Media Encryption and transport setting, but one way audio only, whether it's in out outbound calls. I can hear everything via the WebRTC client, but the other side doesn't hear anything. RTP debug in asterisk shows the right IPs and ports, and everything is being passed by the firewall. My browser is configured to allow microphone access as well.

The settings within FreePBX I applied to the extension I am testing with are as follows, everything else is default settings:
Transport: 0.0.0.0-wss
Enable AVPF - Yes
Enable ICE Support - Yes
Enable rtcp Mux - Yes
Enable WebRTC defaults - Yes
Media Encryption - DTLS-SRTP (not recommended)
Enable DTLS - Yes
Use Certificate - I selected the real, not self-signed SSL cert configured for my server

So what am I missing?

Minor update:
I was loading FOP2 in HTTP but WebRTC needs HTTPS, so I fixed up the ssl configuration in fop2.cfg and reloaded in HTTPS. Now the WebRTC phone behaves slightly differently. Instead of getting stuck on "Trying ##########" it just immediately flips to Call Ended when trying to dial a number.
When receiving a call it rings, I see the callerID, but when answering it immediately flips to Call Ended as well.
So something is still not right... Any ideas?

Does anyone know how to get the WebRTC plugin working on the latest FreePBX? The instructions provided in the WebRTC Guide seem to be specific to IssabelPBX and don't line up completely with the options available on FreePBX.

I have the plugin installed and the extension configured as close as possible to the example in the guide. When I load up FOP 2, the extension appears to register, as shown in FOP2 as well as in the asterisk logs, however it is not able to send or receive calls in any way. It just says "Trying ##########" in the UI and nothing at all in the asterisk logs. I cannot even cancel the call in the WebRTC UI, it just stays kind of frozen until I close the FOP2 page entirely and reload.

I guess I was a little trigger happy on my posting of this issue...

Looks as though disabling websockets in presence.js has resolved the issue for me.

I am still curious as to how or why this would have only affected certain users. I guess ultimately it doesn't matter too much, as long as it works my users are happy, which in turn makes me happy

I am running the latest FOP2 on PIAF Purple.

For most users, log in and usage is fast and everything is great.

For certain users however, logging in to FOP2 takes a very long time (5+ minutes). It seems to be affecting about 10 of my 40 users and there does not seem to be any common denominator between them. All on different hardware with different browsers and even different OS versions.

For all of these users, once they log in, everything is fine however the login process is excruciatingly long.

Any ideas?

Nothing actually. I would put in the value and then click on "Submit Changes". As soon as the page reloaded after the save, the field would be blank again. No errors or warnings anywhere.

I've worked around the issue for now by including the following in buttons_custom.cfg but this is not reflected in the FreePBX module.

[DAHDI/1]
type=trunk
label=DAHDI/Analog
channel=DAHDI/1
channel=DAHDI/2
channel=DAHDI/3
channel=DAHDI/4
channel=DAHDI/5
channel=DAHDI/6
channel=DAHDI/7
channel=DAHDI/8

I apologize if this has been covered in another post, I was not able to find anything after extensive searching...

I have FOP2 installed on my PIAF (purple) server and I am using a combination of SIP and Analog lines. Asterisk is working fine, I can make and receive calls on both SIP and DAHDi.

I am using a Digium 8 port card and it is configured with all 8 ports in g0.

FOP2 displays calls in the SIP trunk perfectly, however it does not display anything in the DAHDi trunk button.

I have tried adding every permutation of "ZAP/DAHDi Channels" to the configuration in the FreePBX module for FOP2 but it never seems to "stick". As soon as I hit "Submit" it goes blank the Dahdi trunk button never shows any calls in or out.

I'm sure I am missing something simple here but for the life of me I can't seem to figure it out.

Any help is greatly appreciated!

Thanks!