[asterisk-commits] CHANGES: correct version for a new option 'refer blind progr... (asterisk[master])

[SVN commits to the Asterisk project]

Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5772 )

Change subject: CHANGES: correct version for a new option 'refer_blind_progress'
......................................................................


CHANGES: correct version for a new option 'refer_blind_progress'

Change-Id: If4817d26a8974610827624fb8a4e56d681d6bf97
---
M CHANGES
1 file changed, 9 insertions(+), 9 deletions(-)

Approvals:
  George Joseph: Looks good to me, approved
  Jenkins2: Approved for Submit
  Joshua Colp: Looks good to me, but someone else must approve



diff --git a/CHANGES b/CHANGES
index 442f59d..8290865 100644
--- a/CHANGES
+++ b/CHANGES
@@ -21,6 +21,15 @@
 --- Functionality changes from Asterisk 14.5.0 to Asterisk 14.6.0 ------------
 ------------------------------------------------------------------------------
 
+res_pjsip
+------------------
+ * A new endpoint option "refer_blind_progress" was added to turn off notifying
+   the progress details on Blind Transfer. If this option is not set then
+   the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted".
+   On default is enabled.
+   Some SIP phones like Mitel/Aastra or Snom keep the line busy until
+   receive "200 OK".
+
 res_agi
 ------------------
  * The EAGI() application will now look for a dialplan variable named
@@ -37,15 +46,6 @@
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 14.4.0 to Asterisk 14.5.0 ------------
 ------------------------------------------------------------------------------
-
-res_pjsip
-------------------
- * A new endpoint option "refer_blind_progress" was added to turn off notifying
-   the progress details on Blind Transfer. If this option is not set then
-   the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted".
-   On default is enabled.
-   Some SIP phones like Mitel/Aastra or Snom keep the line busy until
-   receive "200 OK".
 
 res_rtp_asterisk
 ------------------

-- 
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Gerrit-MessageType: merged
Gerrit-Change-Id: If4817d26a8974610827624fb8a4e56d681d6bf97
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Alexei Gradinari <alex2grad at gmail.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>