Hi,
I have asterisk connected to PSTN via H.323 gateway via chan_oh323.
Incoming calls to SIP extensions work, but SIP message "486 busy here" from a busy extension isn't correctly forwarded to H.323.
As a result, a caller from the H.323 side calling a busy SIP extension gets some rings and then an irritating timeout with H.323 message 'no user responding' instead of 'user busy'.
Asterisk knows the user is busy and jumps to the prio+101 extension. The CDR also logs the call as 'busy'.
With Zap ISDN channels the following works nicely:
exten => 12345,1,Dial(SIP/${EXTEN},45,r)
exten => 12345,102,SetVar(PRI_CAUSE=17)
exten => 12345,103,Hangup
Can we do something similar with chan_oh323?
Many thanks, Jan _______________________________________________ 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
