Do you agree something is blocking the audio in one direction? Can you do a 'rtp debug' and then initiate a SIP call and see if there is two way audio traffic. Also make sure these extensions have 'canreinvite=no'.
Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-06-01 7:02 PM, "Gary Baribault" <g...@baribault.net> wrote: As I stated, the incoming calls are on TDM DS0s connected to the Digium card, and the extensions are on the same local network as the Asterisk server. There is currently no NAT anywhere. Gary Baribault On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote: > > Output of 'iptables -L -n' would also be helpfu... -- _____________________________________________________________________ -- 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
-- _____________________________________________________________________ -- 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
