hi all ! just a question: suppose the following route: sip phone (ulaw) -> asterisk -> internet <- asterisk <- sip phone (ulaw)
it is possible to force the two asterisk to convert the codec from ulaw to, say, gsm ? i mean, without touching the two sip phones thanks for your time .mike _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
