I have upgraded FreeSWITCH several times recently for testing purposes. Also, my router's configuration has changed slightly as I have moved from tunneled IPv6 to a new native IPv6-over-ADSL trial.
However, the problem now is related to my ISP's IPv4-only SIP service, and the symptoms are as follows. 1. If I call a test number, sometimes it all works perfectly. 2. On other occasions (with no discernible pattern) the call connects but no audio is received from the remote end. When this occurs, tshark shows that rtp packets are being sent out to the correct IPv4 address of the server. I am using Stun to handle nat, as my router does not support any of the nat configuration protocols. I want to establish whether it's a router issue or a FreeSWITCH problem. The router is going to be replaced eventually with a small form-factor Linux box and an ADSL2+ card from Traverse Technologies (http://www.traverse.com.au/), but given my priorities at the moment, it won't happen until next year. I can compare SIP traces of that would help. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
