You said you disabled the pjsip channel driver, but it is required for webrtc. See this post:
It is about Zulu but he says that WebRTC requires the same.
If you do not want pjsip to listen for SIP, it looks like according to that thread, you could enable pjsip for WS and WSS only, which should satisfy the webrtc requirement.
Once you can get the PA system responding to an ATA, such that the PA auto-answers calls to the ATA, then it should be a simple matter to create a short custom dialplan to send the DTMF after answer to initiate the page.
Ok then, if i wan’t the pjsip to be active it’s only with the change of the SIP port (5060 to 5160) and SIP channel Driver set to “both”?
I will Run back over and verify this but if I remember right when the phone plugged into the 5100 Goes off hook the PA sends a dial tone. Would this interfere with the ATA at all? Almost like the PA side is the FXS and the ATA would need to be the FXO?
Thanks @lgaetz
So I have added the below into extensions_custom.conf
[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()
Where would I define the extensions of type custom?
Hi, I want to use two inbound Routes. One default and one just for a single Number.
Let assume the single number is +49123456789.
I have two inbound routes.
DID/CID: ANY/ANY --> IVR
DID/CID ANY/+49123456789 --> DISA
but also the call from the single number is routed to IVR.
So, I thought ANY may bound all - regardless any other one configured. So I tried:
DID/CID: ANY/_X! --> IVR
DID/CID ANY/+49123456789 --> DISA
Now, my single Number is routed correctly, but all other does not work anymore.
I used this wiki page https://wiki.freepbx.org/display/FPG/Inbound+Route+User+Guide and the linked https://wiki.freepbx.org/display/FPG/DIAL+PATTERN+INFO)
Nvm, i fixed the pjsip extension creation.
The only way to do this in the GUI is to daisychain call flow controls. But that means which ever ones first if it’s turned on nothing else matters. But it will get the job done if that’s all you want.
I didn’t say you needed to change your extensions or create new ones with pjsip. Just enable the pjsip driver specifically the WS and WSS part so that webrtc can use it.
i’m sorry but i don’t know what this mean
You could also look at using a sip paging adapter such as one from Snom or Cyberdata
Anyone have any ideas?
I’d hate to have to waste credits to contact Tech Support for something like this.
So my endpoint will register to 2004?
I have not used custom extensions before, always PJSIP/ ChanSIP. Is the principle the same? My endpoint will register using the extension as the username and secret as the password? and the port?
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Seems like you have gone through and modified many settings from original, which has made the troubleshooting very confusing and difficult. Websocket mode should be “auto” or “pjsip”.
in Asterisk SIP Settings - pjsip - you can set UDP, TCP, TLS to No, and set WS and WSS to yes.
