No errors, asterisk just immediately sends the other call to voicemail if there is already a call in progress. There's a busy message or something on the CLI (I'm not near the server right now so I can't get the exact text) and then the voicemail macro runs.
Did you do anything funky when assigning the lines to the Cisco? I basically copied the config for a single extension to 4 of the 6 lines. When I do a "sip show peers" I only see one listing... do you get multiple listings for each extension (i.e. a separate registration for each instance of the line?) Thanks, Pat Quoting Joseph <[EMAIL PROTECTED]>: > On Mon, 2005-05-02 at 21:51 +0200, Kristof Hardy wrote: > > Patrick M. Gray, Jr. wrote: > > > In google'ing around a bit, it seems I should be able to assign the same > > > extension to several of the SIP lines on the 7960, and asterisk should > > Works fine for us. > > Do you get errors? > > > -- > respectfully, Joseph =============== > ---------------------= ********** = > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
