- September 5, 2019 at 8:55 pm
My VitalPBX version is 2.3.6-1, Asterisk 16.5.0-1 and DAHDI 2.11.1-7
I’m looking into sngrep, but for the time being I think I’ve noticed something strange, while checking the sip flow in wireshark: While in both successful and unsuccessful cases the connection establishment flow is the same, there seems to be a small difference in the sip flow between the PBX and the extension which picks up the call. I’ve attached two outputs, one working and one not working, as produced by wireshark. In the case where there is one-way audio, the PBX sends a packet in g711U and then switches to g711A. I have even tried to remove 711U from the accepted codecs, but this still happens.
By the way: 192.168.1.110 is the PBX. 192.168.2.1 is a softphone I am using for testing, behind a VPN (regular extensions are in the range 192.168.1.111 and above, and exhibit the same behaviour)0