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WebRTC call button do nothing
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WebRTC call button do nothing
You can test the WSS from your browser. Browse to https://yourname:8089/ws
You should see a message from Asterisk saying “Upgrade required” (it means you are supposed to connect with websocket and not HTTP, but it is useful for a test)
If you don’t see this, maybe your browser will show you a certificate error or if the connection times out altogether you still need to investigate your firewall further.
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WebRTC call button do nothing
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Multiple “night modes”
OK so a few things here.
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How do the five cell phones and home phones log into the queue? Wouldn’t you have these just as static agents so that the person doesn’t have to log in? More importantly they don’t have to log in two devices/agents into the queue.
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Why are calls going to a cell phone along with their IP Phone? The queue cannot track the state of a PSTN number (cell/landline/etc) so it will never know when they are on a call and just continue to send calls to the agent because it sees two available agents.
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Having both their IP phone and the cell phone as agents in the queue means there is always at least one free agent when calls come to them. So doesn’t matter which device they have the call on. The queue will continue to send them new calls while on the phone because it sees available agents. And as pointed out in #2, a PSTN number can’t have a PSTN device state tracked so it can send a call to the cell phone when they are already on it.
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Despite all of the above, this is going to require you to write custom dialplan for your queue changing as nothing in the GUI has something with multiple options for the same Call Flow Control like this.
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WebRTC call button do nothing
It seems like a browser problem then. Asterisk is open and responding to you. Could you try a different browser? Perhaps your Ad Block Plus is stopping the connection.
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WebRTC call button do nothing
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WebRTC call button do nothing
Good. Now check your Asterisk console again. This time the error is probably related to SIP and not webrtc.
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WebRTC call button do nothing
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Modulo Custom Context
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From-internal-custom wildcard extensions
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From-internal-custom wildcard extensions
Yes because now you pushing it a context that has no method of actually dialing the device. You’re initiating a Statis and then hanging up. You need to push this back into proper dialplan if you’re going to do that.
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WebRTC call button do nothing
Try also calling only to Asterisk such as “*43” which is Echo Test. Then you can be sure it is not the softphone’s problem.
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From-internal-custom wildcard extensions
Basically, what I want is my extensions listed in the PBX: 2001, 2002, 2003.
When using the stasis application, I want calls to show in the event logs when being dialled,
Is there any way to do so with the extension being present in extensions and in the extensions_custom.conf?
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From-internal-custom wildcard extensions
Thanks for your help here…
Would it be possible to add the below onto the bottom of my from-internal-custom:
include => from-internal
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WebRTC call button do nothing
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Integrating SIP with Simplex 5100 PA System
Hello,
We are looking into integrating an old Simplex 5100 PA with our Current FreePBX system, I know this is not 100% FreePBX related but I figured I would consult some more experienced people here 
So Our 5100 PA uses and admin station phone that you can Dial #10 in order to do an All-Page. Do any of you know the best way I could tie this into a SIP phone? It uses a proprietary amp so integrating directly with the amp with a SIP paging adapter is a no go for now.
I didn’t know if I could configure an ATA to auto answer the call and then send a DTMF tone over to initiate the page better yet if that’s possible is there way to confine that all to one button on a SIP phone. Almost like script the call? If that’s a thing.
We are looking to RIP this entire system out in the future however the speakers use a nonstandard voltage so that bumped it out of the budget range for now. So I am just looking for the best method to accomplish this.
Thanks!
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WebRTC call button do nothing
And BTW my softphone can call my personnal phone through the trunk sip.
So my softphone is working good i think
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Wallpaper_file | Digium Phones
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WebRTC call button do nothing
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From-internal-custom wildcard extensions
I would undo most of what you have already done, and start with:
[from-internal-AnthKay92]
exten => _XXXX,1,NoOp(entering ${CONTEXT} as defined in extensions_custom.conf)
same => n,Answer()
same => n,Stasis(Tess_Asterisk,PJSIP/${EXTEN}, 45)
same => n,Hangup()
And then define extensions of type “custom” with a dial string of
xxxx@from-internal-AnthKay92
replace the x’s with the ext number.
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