Brent Davidson wrote: > I'm trying to test out Speex for our branch to branch connections, but > am running in to a problem. I downloaded the Speex source code for > 1.2rc1, did a ./configure, make and make install then went to my > asterisk folder did a ./configure, make clean make menuconfig verified > that speex is enabled, saved config then did make, stopped asterisk then > make install and start asterisk. > > Did the exact same steps on one of the branch machines, then went into > sip.conf on both machines and set the codec between the branches to > speex and restarted asterisk on both machines. > > When I try to call between the branches I get the following message: > > [Oct 14 12:26:09] WARNING[23308]: chan_sip.c:3024 sip_call: No audio > format found to offer. Cancelling call to 42 > > I remember seeing a post somwhere along the way stating that the new > version of speex requires a change to the Asterisk code to link against > a new library or something, but I couldn't find the post again. Is > there something I'm missing that is keeping Speex from working? > > Thanks, > Brent Daivdson Well, I'll answer my own question... Needed to do an ldconfig.
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