Propably this is a cisco router issue.
I discovered that if a put
play line in the extensions.conf
so that it can play something before
the call is done, even
for one second, the call is working
normally.

I also played with other variables
in the sip.conf but have not succeeded
except with the play line in extensions.conf



exten => 419,1,Playback(pbx-transfer)
exten => 419,2,Dial(SIP/419)


_______________________________________________ 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

Reply via email to