On Thu, 13 Aug 2015 10:41:31 +0200 "Stefan Viljoen" <viljo...@verishare.co.za> wrote: > Have you checked your RTP port ranges (I'm sure you have), and also
Yes. The ATA is using a range well within the range open on the server. > that the server IP for RTP as specified in the initial SIP is correct? Both the server and client are outside of NAT so I don't know what this might mean. They both have public IPs. > Not sure how this will relate to your setup, but we had something > similar here using Asterisk 1.8.11.0 on both sides of the connection, > via a VOIP service provider in the middle. This is an Asterisk server talking to an ATA. > We had slightly different parameters, e. g. that we would have no RTP > at all, but a call that did connect to total silence, dialed from > either side. Was NAT involved? > Also check what RTP port ranges are being used - I have had this > one-directional problem where the port range > in /etc/asterisk/rtp.conf was too broad, and the firewall on my > server was only allowing a smaller subset of RTP ports. rtpstart=10000 rtpend=20000 which is exactly what my packet filter allows through. > It might require some careful tracing of SIP messages, maybe you can > try this? Specifically try to determine what RTP port number is being > negotiated when you have your zero-audio back from the remote party - > what RTP port and RTP server IP is he using at that moment on his > side? I will check that. Thanks for your suggestions. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users