On 11.06.2013, at 0:24, Sean Darcy wrote: > Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no > success:
Silk is enabled only after asterisk restart. for silk work need codecs.conf with silk configuration res_format_attr_silk.so - loaded codec_silk.so - loaded please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats > > [Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164 > ast_channel_make_compatible_helper: No path to translate from SIP/ng-00000000 > to Motif/+12025551...@voice.google.com-da3c > [Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032 dial_exec_full: > Had to drop call because I couldn't make SIP/ng-00000000 compatible with > Motif/+12025551...@voice.google.com-da3c > == Spawn extension (BaseDial, s, 4) exited non-zero on 'SIP/ng-00000000' > > core show translations doesn't include any SILK. > > SILK is installed: > > core show codec 100018 > 100018 SILK Custom Format 8khz > 100018 SILK Custom Format 12khz > 100018 SILK Custom Format 16khz > 100018 SILK Custom Format 24khz > > sean > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
