please turn on all the debug, warning, error etc messages in the console, see logger.conf, then type sip peer <peer1> debug and sip peer <peer2> debug to see the SIP messages.
How are you testing if asterisk is in the media path? Regards On 1/23/06, Steve Gladden <[EMAIL PROTECTED]> wrote: > been testing with a rather simple setup. > > The mission is to actually get a reinvite to work on the lan. > > I am trying with two sipura phones G.711 codec forced on both > both on the lan no nat no fancy options suchs as tT or H > > No matter what we do asterisk hangs on to the media path, how > in the world do I get a reinvite to work where the media path > is actually handled by the two phones on the lan? > > Any pointers greatly appreciated! > > Steve > > > Pretty simple extensions, on lan no nat > > <sip.conf> > [4785] > > type=friend > username=4785 > secret=test > host=dynamic > canreinvite=yes > > [4786] > > type=friend > username=4786 > secret=tesst > host=dynamic > canreinvite=yes > > <extensions.conf> > exten => 4785,1,Dial(SIP/4785,66) > exten => 4785,3,hangup > > exten => 4786,1,Dial(SIP/4786,66) > exten => 4786,3,hangup > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
