Oh, wise ones, ponder with me over two of the surprises that populate the universe!
I have a phone, that I sometimes cannot reach, connected via pjsip. It can call other extensions just fine, it can call out over a trunk to my cell, all is well, but getting a call? Forget it most of the time.
None of my other phones have two contacts listed…. and this phone, a cisco-spa-514, has just one sip account…
The trouble is, when I try to call it…. sometimes the INVITE is directed to the “Unavail” entry, and the call never completes. The phone doesn’t even ring then. Any ideas? I tried to get the “Unavail” entry out… I
removed it from the db, I rebooted the phone, restarted asterisk, and it is still there.
The above cisco-spa, when it calls out over the trunk, all is well, wonderful 2-way audio. But when I do the same operation from my yealink phones, I get my cell with one-way audio. It’s a classic NAT situation: the phone system is in a droplet at digital ocean, but my phones are here at home behind a NAT. I see only 3 NAT
force_rport rtp_symmetric rewrite_contact
and I set them all to “yes”, and they can call each other, but as explained, in dialing out thru a trunk, the yealinks get one-way audio…