long time ago I added the "SIP_CODEC" variable that you can set from within the dialplan, eg:
exten => s,1,Set(SIP_CODEC=alaw) exten => s,n,Answer exten => s,n,whatever now if the remote side actually supports the chosen codec Asterisk will try to use that one ... there's no error reporting as far as I know Martin On Tue, Oct 20, 2009 at 5:26 PM, Eric Chamberlain <[email protected]> wrote: > Hello, > > I'd like to implement some public sip uri's that poeple can call into > and get an echo test. Is there a way to force a codec so that users > can test various codecs? > > Something like: > > [email protected] (negotiates whatever codec, is there a way to > figure out what codec was negotiated and tell the user) > [email protected] (forces g711) > [email protected] (forces g729) > [email protected] (forces gsm) > ... > [email protected] (forces ilbc) > > -- > Eric Chamberlain, Founder > RF.com - http://RF.com/ > > > > > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
