Hi again,
always latest CVS from 27/06/04. Calling to a SIP gateway I receive:
Unable to find a path from G723 to ALAW Unable to find a path from ULAW to G723 Asked to transmit frame type 4, while native format is 1 (read/write = 8/4) Unable to forward voice [last messages repeated lot of times] Acked pending invite 102 <- My phone number ... No path to translate from SIP/... to SIP/... Had to drop call because I couldn't make SIP/... compatible with SIP/...
Even if I force my sip.conf to use only g723.1 I have the same result.
BTW, if I want to modify my codecs in a sip context, it's not taking in account by asterisk. Is'it normal behaviour?
-- dash _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
