So, when I run the debug, I call from one cell, which calls an inbound route on my server that has a misc. dest. to another cell. No RTP packets show, and the call drops after 30 seconds or so due to that. BUT, if I add a blank Announcement in the route, which then has a destination to the misc dest., it works!
It’s like asterisk kind of answers the call for that blank Announcement, and then forwards it to the misc. dest. so that it works. But if the route is directly to the misc dest., it doesn’t work.
The same behavior is true for dialing my desk phone directly. If I answer it, great, but if find-me forwards it to my cell, there’s no RTP traffic and thus no audio. BUT, if I use an IVR, announcement, or anything else that makes asterisk “answer” first, RTP flows.
That is a network issue? How?
edit: with progressinband=yes, it DOES work with find-me, but not for a misc. dest. I had gotten used to using that for testing, as it was rendering the same results. Again, I can overcome the misc. dest. problem by routing the call through a blank announcement first, but there must be a better way. And what’s strange is that I was definitely able to use misc. destinations like this before, but I can’t think of any changes that would have caused this. All three of our servers are showing the same symptoms, so it points to Anveo, but I just cant be sure.
edit 2: progressinband=yes AND prematuremedia=no BOTH have to be set in Asterisk SIP Settings for the find-me to work. Still no direct route to misc destination though (from an outside call).