is there a trick to get asterisk (1.6.2.13) to propagate codec-changing sip reinvites when directrtpsetup=yes?
i'm trying to route calls to a gateway without keeping asterisk in the rtp stream. the gateway is first routing the call to a media server. when connecting the call to the downstream carrier a different codec is selected. the reinvite makes it to asterisk but asterisk isn't sending it along to the originator so the transmit/receive codecs are mismatched causing one-way audio. thanks, gene -- _____________________________________________________________________ -- 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
