I have a registered extension which is being used to make and receive outbound calls. I have a one way audio on outbound external calls, incoming external calls have two way audio.
when investigating the issue i have found that on calls with successful two way audio the RTP media is always sent to and received from 192.168.116.25 as shown in the Asterisk RTP debug screenshot below.
On calls with one way audio i saw that media was received from 192.168.116.25 but sent to 192.168.115.25 as shown in the Asterisk RTP debug screenshot below.
I asked the customer about the two addresses 192.168.116.25 and 192.168.115.25. They are Italian so i didn’t really understand but i managed to understand that 192.168.116.25 was the correct address to send media to and 192.168.115.25 was not being used anymore.
I have done some research to find out why Asterisk is sending media traffic to 192.168.115.25 so i looked at the SIP packets. In the screenshots below i can see that a 183 Session Progress (SDP) packet is received from the server and the c= field contains 192.168.115.25 as the media address as shown in the screenshot below.
The First 183 Session Progress (SDP) packet is quickly followed by a second 183 Session Progress (SDP) packet which there the c= field contains the correct 192.168.116.25 media address as shown in the screenshot below.
Can anybody help me understand my my Asterisk configuration is sending media to the first incorrect address and not the one contained in the second packet which is correct ?
I am using Asterisk 13.23.0
I have tried registering the same extension to a generic VoIP phone and i am able to make and receive calls with two way audio no problem.
All help is Appreciated ! Thanks in advance.