Hi,

is it possible to let asterisk issue a SIP redirect?

A SIP invite command by a SIP client should be answered
by 30X Temporarly moved to SIP/....

Is this possible with asterisk, maybe from within the dialplan?

(reinvite is not what I'm looking for, because it does not
completely release the originally called SIP server, e.g. if
reinvite fails, ...)

Roger.

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