i’d like to have 2 xlite stations communicate w/ each other directly w/o packets passing through the
asterisk. (of course, after sip does it’s thing.)
i set canreinvite=yes, nat=no, dtmfmode=sip info.
works fine w/ codecs supported by both xlite and asterisk.
doesn’t work w/ codecs not supported on asterisk.
in the later case, i get ‘failed to extablish call’.
since the codec comes into play only for rtp, why do need a codec that is supported also on the
with the setting described above, rtp packets are going directly between the 2 xlite stations.
i’m using xlite v4.5.5
asterisk v 1.8.6