diff options
author | Matthew Jordan <mjordan@digium.com> | 2015-02-12 20:32:48 +0000 |
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committer | Matthew Jordan <mjordan@digium.com> | 2015-02-12 20:32:48 +0000 |
commit | 1995baad71412b740743f2be82d7323f7c15de12 (patch) | |
tree | 8648c6d938fda51030f4ce669f3d2cdf507a2bc4 /CHANGES | |
parent | e8ec15a9ef019d0dcfa875a5f08b9e1277fe701d (diff) |
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
Diffstat (limited to 'CHANGES')
-rw-r--r-- | CHANGES | 19 |
1 files changed, 19 insertions, 0 deletions
@@ -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 ------------ ------------------------------------------------------------------------------ |