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Hi, I found the reason. Asterisk did not recognize DTMF-event because my
SIP phone sent DTMF-event with wrong rtp payload type.
In short, Asterisk is not guilty. When the SIP phone calls, it will advertise RTP payload type 96 for
DTMF-Event; Asterisk answers with 96 and expects 96. So everything is OK. When the SIP phone is called, Asterisk advertises RTP payload type
101 for DTMF-Event; my SIP phone answers with 96. Asterisk expects 101, but my
SIP phone sends DTMF-event with RTP payload type 96. Asterisk complains “unknown
rtp payload type 96”. My college will fix the phone. Before they start, I changed
Asterisk instead. Now Asterisk takes 96 as the default value and everything is
OK. BR Younger Wang -----Original Message----- 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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