Hi,
I encountered something i can't understand. I've setup 2 extensions.
[100]
type=friend
host=dynamic
nat=yes
secret=100
[101]
type=friend
host=dynamic
nat=yes
secret=101
and on extensions.conf
exten => _1XX,1,Dial(SIP/${EXTEN}|30|t)
exten => _1XX,n,Hangup
This dial plan is ok, audio connects both ways.
but when i had a typo error, i forgot the 't' option, only one way audio
when i call, 't' option is used to transfer call how come it affected
the audio?
thank you in advanced
regards
nhadie
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users