I am currently running an Asterisk version 13.1.0 on a Ubuntu 16.04 server.
We are facing one way voice only in certain outgoing calls.
I was able to reproduce the issue and it appears only in the following scenario:
I configured on my VoIP provider a number that transfers incoming calls to 3 numbers(simultaneously)
Then I called this number with my extension, the VoIP provider sends “183 session progress” ergo ring back was send via early media(negotiated codec and port for RTP)
One of the ring group members picks up the phone, source port from the RTP proxy of my VoIP provider for transmitting RTP packets changes(in 200 OK SIP packet)
Asterisk does not send the RTP packets to the in the 200 OK transmitted destination port
I retested the same scenario with my test VoIP Phone Yealink T4G VoIP without the PBX bewtween and the issue does not appear.
Respectively the VoIP phone was able to change the destination port after the call receiver has responded.
First of all, I attempted to search an option in the Asterisk config that could help out, but wasn’t able to find a matching setting.
Is it possible that this issues seems to be an Asterisk bug? If yes, could someone post a link(didn’t find a bug report).
If no, can someone give me a hint what I could try to solve this issue?
Any helpful input would be greatly appreciated.