Hi: I am working on a scenario where I need to 1) create outgoing SIP channel 2) send re-INVITE 3) bridge the outgoing channel with an incoming channel scenario: user1 and user2 are in call with each other. (end-to-end RTP traffic) (when this call was placed, sip header values were dumped in a file) user3 calls user2, asterisk follows above 3 steps to establish call between user2 and user3. (transfer user2 to the new call) Does anybody know how to create a new channel and bridge two channels manually? Thanks, Jim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060313/4ea88676/attachment.htm