ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app

This patch adds a new feature to ARI to redirect a channel to another server,
and fixes a few bugs in PJSIP's handling of the Transfer dialplan
application/ARI redirect capability.

*New Feature*
A new operation has been added to the ARI channels resource, redirect. With
this, a channel in a Stasis application can be redirected to another endpoint
of the same underlying channel technology.

*Bug fixes*
In the process of writing this new feature, two bugs were fixed in the PJSIP
stack:
(1) The existing .transfer channel callback had the limitation that it could
    only transfer channels to a SIP URI, i.e., you had to pass
    'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is
    still supported, it is somewhat unintuitive - particularly in a world full
    of endpoints. As such, we now also support specifying the PJSIP endpoint to
    transfer to.
(2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by
    updating its Contact header. Alas, that resulted in the forwarding
    destination set by the dialplan application/ARI resource/whatever being
    rewritten with very incorrect information. Hence, we now don't bother
    updating an outgoing response if it is a 302. Since this took a looong time
    to find, some additional debug statements have been added to those modules
    that update the Contact headers.

Review: https://reviewboard.asterisk.org/r/4316/

ASTERISK-24015 #close
Reported by: Private Name

ASTERISK-24703 #close
Reported by: Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431717 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 19 insertions, 0 deletions

diff --git a/CHANGES b/CHANGES
index 15080405d..749338a4c 100644
--- a/CHANGES
+++ b/CHANGES
@@ -7,6 +7,25 @@
 === and the other UPGRADE files for older releases.
 ===
 ==============================================================================
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------
+------------------------------------------------------------------------------
+
+chan_pjsip/app_transfer
+------------------
+ * The Transfer application, when used with chan_pjsip, now supports using
+   a PJSIP endpoint as the transfer destination. This is in addition to
+   explicitly specifying a SIP URI to transfer to.
+
+res_ari_channels
+------------------
+ * The ARI /channels resource now supports a new operation, 'redirect'. The
+   redirect operation will perform a technology and state specific redirection
+   on the channel to a specified endpoint or destination. In the case of SIP
+   technologies, this is either a 302 Redirect response to an on-going INVITE
+   dialog or a SIP REFER request.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.1.0 to Asterisk 13.2.0 ------------
 ------------------------------------------------------------------------------