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

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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
........

Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matthew Jordan
2015-02-12 20:34:37 +00:00
parent 9d081ed06c
commit 29f66b0429
11 changed files with 333 additions and 5 deletions
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18
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@@ -100,6 +100,24 @@ res_musiconhold
over the channel-set musicclass. This allows separate hold-music from
application (e.g. Queue or Dial) specified music.
------------------------------------------------------------------------------
--- 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 ------------
------------------------------------------------------------------------------