Hello, I have an asterisk setup using pjsip. Everything used to work correctly until one remote site changed internet provider and thier router does not support sip protocol algorithms.
It works for some time, but then suddenly audio stops working both directions. When this happens I see RTP responses going out to the local address of the natted phone, not to the natted address. The problem appears for the phones independently. The asterisk is connected to the internet with public static IP address. The pjsip config contains: [aor] type=aor qualify_frequency = 60 max_contacts=1 remove_existing = yes [endpoint] type = endpoint context = internal dtmf_mode = rfc4733 disallow = all allow = alaw allow = ilbc allow = g729 allow = gsm allow = g723 direct_media = no allow_subscribe = yes subscribe_context = blf rewrite_contact = yes rtp_symmetric = yes force_rport = yes Am I missing something? Why the communication breaks suddenly? Thanks Marek -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
