[Asterisk-Users] Softphone - Freeware?!

[Maurizio Marini]

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 >You do need to have this enabled in the dialplan dial strings to
 >enable transfers. 
u should use something like this:

[from-sip]

exten => 101,1,Dial(SIP/sip1,20,tTr)

from http://www.voip-info.org/wiki-Asterisk+cmd+Dial:

The options parameter, which is optional, is a string containging zero or more 
of the following flags and paramters: 
t: Allow the called user to transfer the call 
T: Allow the calling user to transfer the call 
r: Generate a ringing tone for the calling party, passing no audio from the 
called channel(s) until one answers. Use with care and don't insert this by 
default into all your dial statements as you are killing call progress 
information for the user. 


- -- 
Maurizio Marini		GSM +39-335-8259739
Work: +39-0721-855285	Fax +39-0721-859609
Home: +39-0721-950396	IAXTel: (700) 350-1234
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