[asterisk-users] Speex Problem

[Brent Davidson]

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.