|
Hi, I have been
trying to enable attended transfer for callee. When
the callee pressed *2, DTMF tone was heard by the
caller. But when the caller pressed *2, attended transfer started. It’s
strange. I used two
SIP phones. My Asterisk version is “Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running
Linux on 2005-06-27 In features.conf, I have: [featuremap]
blindxfer => #1
; Blind transfer disconnect => *0 ;
Disconnect ;automon =>
*1
; One Touch Record atxfer => *2
; Attended transfer My extensions.conf is like this: exten => _8XXX,1,Dial(SIP/${EXTEN},30,Ttm) Another
problem is, when caller started the transfer, no dial tone is given. The log
said “NOTICE[11245]: app.c:67 ast_app_dtget: Huh....? no dial
for indications?”. Anybody has
the same problem as I do? BTW, can I have more precise control of transfer behavior?
If yes, will anybody show me the document? Thank you
very much! BR Younger Wang |
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