I’m using succesfully a sip trunk between freepbx and SipToSis to manage Skype calls
All is installed on same machine and works fine for months until the previous Skype needed to be replaced due to changes in authentication.
Done this, all calls are placed correctly but I have no audio both directions.
I have modified nothing other than new Skype installation (just a .bz2 decompressed on new folder and made run) ; usual FreePbx modules upgrade.
It seems to be an RTP issue:
chan_sip.c:28883 check_rtp_timeout: Disconnecting call ‘SIP/Skype-0000001d’ for lack of RTP activity in 31 seconds
(then Asterisk drops the call)
Since RTP packets are managed between Asterisk side of SipToSis and FreePbx trunk itself (where nothing has been modified) it’s quite tricky to understand what is happened.
I think it isn’t a NAT issue or something like, as both peers reside inside the same machine, with the same IP address on different ports.
SipToSis trace reports 0 RTP packets received.
RTP debug on FreePbx traces a lot of packets between its IP address and the related IP phone address , nothing more.
Does anybody experienced something like this ?
Where can I start to dig deeper for solving the issue ??