snip > > Can Meetme be made to work with G.729? (I gather not) > IIRC, MeetMe does it 'mixing' using SLIN (Signed Linear, * should transcode to/from g.729 to SLIN.
> If a call comes in (internally or externally), the call comes in as a > G.729 call, which then re-negotiates to a G.711u call when if > gets transferred to a MeetMe room. It sould be transcoded into SLIN from g.729. If you phone is requesting the use of G.711 you may want to use a disallow/allow in your sip.conf. > > Is there a way to set up asterisk that will allow me to have > internal phones renegotiate to G.711, with the external lines > instead transcoding within asterisk. > (runtime is more available than bandwidth). > I don't follow, You want the inside (high-bandwidth) phone to use G.711. If so use the allow/disallow in sip. > Also, does anyone know if there's a way to dynamically alter > incoming Caller-IDs to add Caller ID text to them. > > ie. call comes in with ID 01234 567890 gets changed to "A Company" > <01234 567890> ? > > This will be with a lot of numbers, so a pile of conditionals > in the extensions.conf file wouldn't really be a good solution. > Set(CALLERID(name)="What ever you want") If you want Dbfunctions you could write an AGI script, or you can use the DB function within Asterisk (ASTDB and *SQL) to set the variable. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users