Why didn't I think of using that command... It shows all "-" for G729a which is presumably why I'm having a problem
I have purchased 20 licenses from Digium, downloaded binary, registered the binary correctly, placed it in the correct directory and it is listed specifically in SIP.conf I'm sure that I have had some calls between SIP phones using G729a via Asterisk (not re-invited) How can I be sure that the G729a codec is working correctly? Very much appreciate your response >>> [EMAIL PROTECTED] 19/07/05 14:31:57 >>> What does your 'show translation' look like? Can you copy/paste the specific *.conf entries for the sip devices and capi? > Message is "no translator path exists for channel type CAPI (native 8) to 256" > > >>> [EMAIL PROTECTED] 19/07/05 13:44:29 >>> > > > I though that Asterisk would transcode between codecs! All my SIP devices > > support G729a & > 711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to > accept a call from a SIP > device using G729a and then complains that it can't translate into G711 to go > onto the ISDN > network. Does anybody know if there is some setting somewhere or if this is > how it is supposed > to work > > > What does the sip debug show? > > Any CLI data to give us a clue? _______________________________________________ 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 Scanned by MailDefender - managed email security from intY - www.maildefender.net _______________________________________________ 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
