I noticed that in 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 try it 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.
Asterisk staying in the media path after the transfer is unacceptable, so please do not suggest setting can reinvite=no.