> From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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. _______________________________________________ -- 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
