asterisk users - Jul 2005 - Attended transfer works for caller, not for callee

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 root@router on a i686 running Linux on 2005-06-27 06:07:18".
 
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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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-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Younger
Wang
Sent: 2005?7?1? 16:25
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Attended transfer works for caller, not for
callee
 
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 root@router on a i686 running Linux on 2005-06-27 06:07:18?.
 
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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