El Fri, 04 Apr 2014 08:18:22 +0200 Rainer Piper <rainer.pi...@soho-piper.de> escribió:
> Hallo, > my guess is the audio codec opus > > asterisk can NOT do transcoding from opus to pcmu. > > The opus codec in asterisk is (just) a path through codec. > > your trace right at the end: > !!! Failed to parse SessionDescription. Failed to parse audio codecs > correctly !!! > Just in case you don't know the patch: https://github.com/meetecho/asterisk-opus cheers, Jon _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
