Brian J. Murrell wrote: > On Wed, 2008-04-09 at 17:21 -0500, Brent Davidson wrote: > > > Does anyone know if Asterisk will convert an inband DTMF from one sip > channel to an info or rfc2833 DTMF on another (i.e. bridged) SIP > channel? You might also try "canreinvite=no" for both your phone and the sip peer. I think it's normal procedure for Asterisk to drop out of the call path once the call is established between two peers. The canreinvite directive forces asterisk to remain as an intermediary, and it will probably do the transcoding that way. If I'm not mistaken this is also useful for making calls between two system that have no common codecs.
-Brent _______________________________________________ -- 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
