How to issue a SIP REINVITE

I have a SIP trunk provider that does not support a REFER. They said I can use reinvite to transfer a call, but I am unclear how to initiate that. Transfer() does a REFER, what does a reinvite? Or is that not possible for a call transfer? I am trying to send a incoming call directly to a cell phone without having to have the rtp traffic go threw our equip or the call use up 2 trunks. Thanks! - Jeremy

directmedia (bypassing the RTP) is not the same as a transfer. Asterisk can use re-invites for that, as long as you meet the preconditions (neither side can have directmedia=no, codecs must match, and you mustn’t do any recording, or anything that requires listening for DTMF).

I’m not aware of any way in which Re-INVITE can be used for true transfers.

I think they are saying we do not support transfer, but we do support direct media.

PS. I suspect trunk is a commercial concept here. direct media will stop your IP bandwidth being used, but might still be counted as two trunks for any limit on the number of simultaneous calls, or charging purposes.

A true transfer would probably be charged as two calls.

That is what I kind of thought but was not sure. Thanks for clarifying it! - Jeremy