diff options
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 45 |
1 files changed, 45 insertions, 0 deletions
@@ -12,6 +12,14 @@ --- Functionality changes from Asterisk 13.14.0 to Asterisk 13.15.0 ---------- ------------------------------------------------------------------------------ +AMI +------------------ + * The 'PJSIPShowEndpoint' command's respone event of 'IdentifyDetail' now + contains a new optional parameter, 'MatchHeader', mapping to the new + configuration option 'match_header' for the corresponding 'identify' object. + It should be noted that since 'match_header' takes in a key: value pair, the + event parameter will contain a ':' as well. + app_record ------------------ * Added new 'u' option to Record() application which prevents Asterisk from @@ -33,6 +41,22 @@ app_voicemail * Added 'fromstring' field to the voicemail boxes. If set, it will override the global 'fromstring' field on a per-mailbox basis. +res_pjsip +------------------ + * A new transport parameter 'symmetric_transport' has been added. + When a request from a dynamic contact comes in on a transport with this + option set to 'yes', the transport name will be saved and used for + subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's + saved as a contact uri parameter named 'x-ast-txp' and will display with + the contact uri in CLI, AMI, and ARI output. On the outgoing request, + if a transport wasn't explicitly set on the endpoint AND the request URI + is not a hostname, the saved transport will be used and the 'x-ast-txp' + parameter stripped from the outgoing packet. To facilitate recreation of + subscriptions on asterisk restart, a new column 'contact_uri' needed to be + added to the ps_subcsription_persistence table. Since new columns were + added to both transport and subscription_persistence, an alembic upgrade + should be run to bring the database tables up to date. + res_pjsip_transport_websocket ------------------ * Removed non-secure websocket support. Firefox and Chrome have not allowed @@ -41,6 +65,27 @@ res_pjsip_transport_websocket when Asterisk attempts to send SIP requests to do something like initiate call hangup. +res_pjsip_endpoint_identifier_ip +------------------ + * A new option has been added to the 'identify' configuration object, + 'match_header'. The 'match_header' attribute should contain a SIP + header: value pair that, When set, will cause inbound requests that contain + the matching SIP header/value pair to be associated with the corresponding + endpoint. This option is cumulative with the 'match' option, so that if + either option matches the request, the request is associated with the + endpoint. + + In a future release, this module will be renamed to something more + appropriate, as it now matches inbound requests on more than just IP + address. + +res_rtp_asterisk +----------------- + * The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP + Data and Control Packets on a Single Port." So far, the only channel driver + that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on + a PJSIP endpoint in pjsip.conf to enable the feature. + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 13.13.0 to Asterisk 13.14.0 ---------- ------------------------------------------------------------------------------ |