On 11-05-16 09:13 AM, Alex Balashov wrote: > On 05/16/2011 09:00 AM, Mohammad Khan wrote: > >> Is there way I can use two Asterisk box, one to maintain SIP packets and >> other for RTP traffic? > > No, the signaling and bearer plane are integrated in Asterisk. > > But you can use reinvites to hand off RTP processing to third-party endpoints > and bypass Asterisk, in qualifying call scenarios and network topologies.
You could try directrtpsetup=yes which is similar to directmedia, except the audio is redirected in the initial INVITEs rather than reinviting the media a few RTP packets in. Leif. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
