[Asterisk-Users] Meetme codec translation and callerID library.

[Alexander Lopez]

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.