I have read TAPI Click-To-Dial possible? - #24 by Drrehak
and that is for if a number is to be dialed as well as an ext but i just need the call the ext not a number how do i do this?
It’s not clear whether the A side is your phone, or someone else’s. In the latter case, you probably don’t want the AddHeader. Also the way that code does AddHeader will only work for chan_sip, and should be retiring chan_sip, in favour of chan_pjsip. In addition, FreePBX is not limited to SIP extensions.
Also, under FreePBX, it is arguably better to use a local channel, on A the side, to use the FreePBX logic on that side. You should be able to use “*80” in the number, to get an an autoanswer, if calling yourself.
(It doesn’t make sense to dial a number on the A side, and do nothing on the B side, but the B side doesn’t have to be a trunk or extension.)
(If you want to reopen an old topic, you can flag it and ask for it to be reopened. However, in this case, I think that would be the wrong action.)
Can you please describe your scenario in more detail, as a call from a web page, to me, means that the web server makes a SIP call, or the browser makes a WebRTC call, both of which would be handled as normal incoming calls by FreePBX.
should then automattically call my pbx ext 2000 (no auto answer) when i pick up i can then communicate with the caller that is using a browser. understand? there is no outside line needed, just internet.
That’s not click and call as normally understood. Although I’ve never used WebRTC, I believe it is a WebRTC call from the browser. To Asterisk there is one incoming call leg and one outgoing one.
As far as I know it is not sensible and probably not possible to do a click to call scenario to a browser, but if you could the channel parameter in the originate, would be the address of the browser based WebRTC client, and the num parameter would be your extension 2000.
Although I haven’t validated it in detail, this topic seems to be about what you actually want to do:
Be warned that WebRTC is a moving target. You should make sure you have the latest versions at both ends (e.g. it would be unwise to try to continue to use chan_sip with it). You should be prepared to have to read up on protocol and do low level debugging.