Hello, I am trying to get to Jain Sip softphones to call one another via an Asterisk server. When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways. When I look at the asterisk log file it has an entry which says: "Oooh, format changed to 2".
Would anyone know why this is occuring one way and not the other, and more importantly, how would I fix this. After some examination I see that when I send the OK to the INVITE, this SDP body should have a 0 for the codec which is ulaw. When this Ok message gets to the other pc after going through asterisk it seems like asterisk adds a codec because the SDP body now contains the codecs 0 and 3. I believe the problem has something to do with this but I am not sure why it would work one way but not the other. Any help would be greatly appreciated. Thanks very much, Denis Kutman _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
