[asterisk-users] Speex Problem

[Brent Davidson]

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