[Asterisk-Users] SIP signaling and canreinvite=yes
Kevin P. Fleming
kpfleming at digium.com
Mon Nov 14 23:08:45 MST 2005
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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