I have been fighting various flavors of re-invite periodically. I have seen re-invite issues on both incoming and outgoing based on the route. Has anybody seen this and have bene able to trap what is going on?
When all works – my phone sitting behind my PBX gets invited to the destination SIP server and the two talk nicely without having any network traffic on my server.
When I select the Alcazar outgoing route Asterisk connects the phone to Alcazar and it works for about one second. When the re-invite happens audio drops from the phone behind the PBX. Since this works with most of the Anveo Direct routes I am very puzzled. Under normal circumstances I will only see RTP traffic on the Asterisk server for a second and the re-invite fixes that and all works.
My theory is there is some from of reinvite data that Asterisk is getting which it doesn’t handle and in reality it shouldn’t do the re-invite but that is just a theory.
I had a second issue that I will submit under separate cover on incoming SIP RTP to Anveo Direct.
Thoughts on my theory would be appreciated along with any suggestions.