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1 files changed, 78 insertions, 2 deletions
diff --git a/CHANGES b/CHANGES
index 55ec3292a..95354791b 100644
--- a/CHANGES
+++ b/CHANGES
@@ -9,6 +9,84 @@
==============================================================================
------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.11.0 to Asterisk 13.12.0 ----------
+------------------------------------------------------------------------------
+
+app_voicemail
+------------------
+ * Added "tps_queue_high" and "tps_queue_low" options.
+ The options can modify the taskprocessor alert levels for this module.
+ Additional information can be found in the sample configuration file at
+ config/samples/voicemail.conf.sample.
+
+res_pjsip_mwi
+------------------
+ * Added "mwi_tps_queue_high" and "mwi_tps_queue_low" global configuration
+ options to tune taskprocessor alert levels.
+
+ * Added "mwi_disable_initial_unsolicited" global configuration option
+ to disable sending unsolicited MWI to all endpoints on startup.
+ Additional information can be found in the sample configuration file at
+ config/samples/pjsip.conf.sample.
+
+chan_pjsip
+------------------
+ * A new dialplan function, PJSIP_SEND_SESSION_REFRESH, has been added. When
+ invoked, a re-INVITE or UPDATE request will be sent immediately to the
+ endpoint underlying the channel. When used in combination with the existing
+ dialplan function PJSIP_MEDIA_OFFER, this allows the formats on a PJSIP
+ channel to be re-negotiated and updated after session set up.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.10.0 to Asterisk 13.11.0 ----------
+------------------------------------------------------------------------------
+
+chan_dahdi
+------------------
+ * Added "faxdetect_timeout" option.
+ The option determines how many seconds into a call before faxdetect
+ is disabled for the call. Setting the value to zero disables the timeout.
+
+chan_sip
+------------------
+ * Session-Timers (RFC 4028) work for TCP (and TLS) transports as well now.
+ Previously Asterisk dropped calls only with UDP transports. However with
+ longer international calls via TCP, the SIP channel might break, because
+ all hops on the Internet route must stay online (have not a single power
+ outage, for example). Therefore with Session-Timers enabled (which are
+ enabled at default), you might see additional dropped calls. Consequently
+ please, consider to go for session-timers=refuse in your sip.conf.
+
+res_pjsip
+------------------
+ * Added "fax_detect_timeout" to endpoint.
+ The option determines how many seconds into a call before fax_detect
+ is disabled for the call. Setting the value to zero disables the timeout.
+
+ * Added "subscribe_context" to endpoint.
+ If specified, incoming SUBSCRIBE requests will be searched for the matching
+ extension in the indicated context. If no "subscribe_context" is specified,
+ then the "context" setting is used.
+
+res_rtp_asterisk
+------------------
+ * The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS).
+ Enabling PFS is attempted by default, and is dependent on the configuration
+ of the module using TLS.
+ - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
+ specify a ECDHE cipher suite in sip.conf, for example:
+ dtlscipher=AES128-SHA
+ - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
+ into the private key file, e.g., sip.conf dtlsprivatekey. For example:
+ openssl dhparam -out ./dh.pem 2048
+ - Because clients expect the server to prefer PFS, and because OpenSSL sorts
+ its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
+ Consider re-ordering your cipher suites in the respective configuration
+ file. For example:
+ dtlscipher=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256
+ which forces PFS and requires at least DTLS 1.2.
+
+------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.9.0 to Asterisk 13.10.0 -----------
------------------------------------------------------------------------------
@@ -75,8 +153,6 @@ res_pjsip
"contact_deny" - List of Contact header addresses to deny
"contact_permit" - List of Contact header addresses to permit
- * Added new status Updated to AMI event ContactStatus on update registration
-
* Added "reg_server" to contacts.
If the Asterisk system name is set in asterisk.conf, it will be stored
into the "reg_server" field in the ps_contacts table to facilitate