[Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515

[Lea]

I noticed that in the file "chan_sip.c" there is a constant 
SIP_REFER. 

Does this mean that Asterisk can initiate SIP-SIP transfers with 
the REFER / NOTIFY methods according to RFC-3515 ? 

If not, why not ? ...after all RFC-3515 is almost 3 years old 
and I thought Asterisk is on the cutting edge of VoIP... 

Anyway, if somebody has some code to make Asterisk compliant 
with this RFC-3515 or some hidden #defines or maybe compiler 
options to enable this crucial transfer functionality, please 
scribble something back. 

The SIP reinvite method simply does not work with some providers 
such as Vonage, see my post on the Asterisk Users list/forum. 

You can experience this problem yourself (I can send you the 
Vonage host, login and password for testing). 
The REFER / NOTIFY methods work every time with all the 
providers I have ever tried. 


Regards, 
Daniel 

P.S. 
Asterisk staying in the media path after the transfer is 
unacceptable, so please do not suggest setting can reinvite=no. 


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