i’d like to establish a session between 2 phones using codecs supported on both phones, but not supported on asterisk.
i tried various suggestions and combos thereof (canreinvite yes, dtmf sip info, no nat, media direct).
no luck yet.
seems to me (an others) that it should be doable though.
no problem getting all rtp traffic directly between the phones w/o asterisk in between ONLY IF
it’s a codec also supported by asterisk. i want to use codecs NOT supported by asterisk.
i’m using asterisk 1.8.6 w/ freepbx, x-lite 4.5.5, snom300
i’m willing to try alternative freeware softphone since i’m told that xlite may not be handling
canreinvite properly (or at all).