Well…here is the news!
I was suspicious of Zoiper and a codec issue. Zoiper wants you to pay for H.264 otherwise it will only use VP8. FreePBX/Asterisk does support VP8, I made sure it was enabled and at the top of the codec list. Still nothing worked. I put down the iPhone and installed Linphone on my PC with the H.264 codec.
Two things happened:
- A call from the DX-80 to the PC resulted in two way audio/video(BOOM!)
- A call from Linphone resulted in the PC receiving video from the DX-80 but the DX-80 displayed no video from the PC. Place the call on hold on the DX-80, pick it back up and poof. Bidirectional audio and video. Every time. That’s a problem I will chase down later. At least it works!
Thank you everyone for trying to help me figure this out.
Now back to the original question that got us here. If you saw my original post, how can I add URI dialing to my dial?
When I try to dial "firstname.lastname@example.org" or "email@example.com" etc. it fails. I could paste the whole log here but I think this is a problem someone knows the answer to…ultimately the ending errors are:
[2021-01-07 18:33:37] NOTICE : res_pjsip_session.c : 4007 new_invite : 2100: Call (TCP:192.168.1.12:42659) to extension ‘wbdemo’ rejected because extension not found in context ‘from-internal’.
[2021-01-07 18:33:37] DEBUG : res_pjsip_session.c : 4007 new_invite : 2100: Call (TCP:192.168.1.12:42659) to extension ‘wbdemo’ rejected because extension not found in context ‘from-internal’.
[2021-01-07 18:33:37] DEBUG : res_pjsip_session.c : 4479 handle_outgoing_response : 2100: Method is INVITE, Response is 404 Not Found
So the INVITE fails with the 404 Not Found error.
@Stewart1 - I could live with the dedicated extension as previously suggested If I had to. When setting the extension to “SIPfirstname.lastname@example.org” I get audio only. I don’t know if there is supposed to be video.
@billsimon If I paste “SIPemail@example.com” into the dial field, I get the screen and audio that you showed above. If I try to use “PJSIP/anonymous/sip:firstname.lastname@example.org” as you have it typed, it fails and doesn’t even show up in my sngrep list.
There is a fairly old article on how to accomplish this task and it is for a much earlier version of Asterisk. Here is the article https://www.voip-info.org/asterisk-tips-sip-uri-dial so i would think there still has to be a way to make it work.
I am using FreePBX so you’ll have to scroll half way through the article where the author says what to do if you’re using FreePBX. I tried it(i think correctly) but it’s beyond my understanding of asterisk.
Thank you again to everyone who is chiming in to help!