We're trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number.
UK Landline->voipfone->asterisk 1.4->voipfone->UK landline About 1 in 3 times the call at the final landline is silent and we see "RTP Read too short" scrolling on the console log. Where do we start working out what's going on? Other than that the server is working well John
-- _____________________________________________________________________ -- 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