[asterisk-users] FW: transcoder

[Steve Langstaff]

> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Khaled Chehab
> Sent: 07 February 2008 12:33

> What I am asking for is something to take an incoming SIP 
> INVITE, change the codecs listed in the SDP, forward the 
> (new) INVITE to a media gateway, perform the reverse codec 
> handling for the 200 OK and perform RTP transcoding on the 
> resulting 2 legs of the call.

[Eliza, is that you?]

> -How can asterisk do that !

Have sip.conf entries for your phones that have:

disallow=all
allow=g711

and an entry for your media gateway that has:

disallow=all
allow=g723
allow=ilbc
allow=g729

You will also need some extensions.conf stuff to forward calls from the
phones to the media gateway and vice-versa.