[Asterisk-Users] channel manipulation

[JS]

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
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