Displaying 1 result from an estimated 1 matches for "07f2055c".
2007 Sep 11
1
Chan_sip Entry
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