[Asterisk-Users] SIP signaling and canreinvite=yes

[Kevin P. Fleming]

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.