Damon Estep wrote:
Does it work? I am having trouble getting it to work that way.
Yes.
Is the sip signaling all handled by asterisk in this case? - required by my providers session border controller.
Yes. It is not possible for a SIP UA to remove itself from a SIP DIALOG without using REFER or INVITE/Replaces to transfer the call to another party. Just moving the media around does not impact the path that the signaling follows.
I guess what I am asking is can asterisk function as a SIP PROXY when configured correctly?
No, Asterisk cannot be configured in any way to act as a proxy. It is _always_ a back-to-back UA.
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