From 52148d93f4f672433a28f0f60cca9f20055a8c5c Mon Sep 17 00:00:00 2001 From: Matt Jordan Date: Sun, 15 May 2016 12:22:42 -0500 Subject: CHANGES: Update formatting of items * Provide consistent indenting of lines in bulleted paragraphs * Respect the 80 character column width * Group all like items together, e.g., all dialplan applications under "Applications", etc. * Use a single blank line to break up functionality changes within a larger section * Use two blanks lines to delineate larger sections Change-Id: I0488554f5cb7c51da70003d69288a21c9aab9647 --- CHANGES | 90 +++++++++++++++++++++++++++++++++++++---------------------------- 1 file changed, 52 insertions(+), 38 deletions(-) diff --git a/CHANGES b/CHANGES index 15b4d0c67..9c60b2443 100644 --- a/CHANGES +++ b/CHANGES @@ -15,13 +15,13 @@ ARI ----------------- * A new ARI method has been added to the channels resource. "create" allows for - you to create a new channel and place that channel into a Stasis application. This - is similar to origination except that the specified channel is not dialed. This - allows for an application writer to create a channel, perform manipulations on it, - and then delay dialing the channel until later. + you to create a new channel and place that channel into a Stasis application. + This is similar to origination except that the specified channel is not + dialed. This allows for an application writer to create a channel, perform + manipulations on it, and then delay dialing the channel until later. - * To complement the "create" method, a "dial" method has been added to the channels - resource in order to place a call to a created channel. + * To complement the "create" method, a "dial" method has been added to the + channels resource in order to place a call to a created channel. * All operations that initiate playback of media on a resource now support a list of media URIs. The list of URIs are played in the order they are @@ -32,6 +32,7 @@ ARI back to the resource. The "PlaybackFinished" event is raised when all media URIs are done. + Applications ------------------ @@ -73,6 +74,17 @@ Playback provided, including the file extension. Currently, on HTTP and HTTPS URI schemes are supported. +Queue +------------------- + * Added field ReasonPause on QueueMemberStatus if set when paused, the reason + the queue member was paused. + + * Added field LastPause on QueueMemberStatus for time when started the last + pause for a queue member. + + * Show the time when started the last pause for queue member on CLI for command + 'queue show'. + SMS ------------------ * Added the 'n' option, which prevents the SMS from being written to the log @@ -80,20 +92,6 @@ SMS providers to not log SMS content. -CDRs ------------------- -cdr_odbc ------------------- - * Added a new configuration option, "newcdrcolumns", which enables use of the - post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'. - ------------------- -cdr_csv ------------------- - * Added a new configuration option, "newcdrcolumns", which enables use of the - post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'. - - Channel Drivers ------------------ @@ -101,6 +99,7 @@ chan_dahdi ------------------ * The CALLERID(ani2) value for incoming calls is now populated in featdmf signaling mode. The information was previously discarded. + * Added the force_restart_unavailable_chans compatibility option. When enabled it causes Asterisk to restart the ISDN B channel if an outgoing call receives cause 44 (Requested channel not available). @@ -110,6 +109,7 @@ chan_iax2 * The iax.conf forcejitterbuffer option has been removed. It is now always forced if you set iax.conf jitterbuffer=yes. If you put a jitter buffer on a channel it will be on the channel. + * A new configuration parameters, 'calltokenexpiration', has been added that controls the duration before a call token expires. Default duration is 10 seconds. Setting this to a higher value may help in lagged networks or those @@ -120,9 +120,11 @@ chan_sip * New 'rtpbindaddr' global setting. This allows a user to define which ipaddress to bind the rtpengine to. For example, chan_sip might bind to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10). + * DTLS related configuration options can now be set at a general level. Enabling DTLS support, though, requires enabling it at the user or peer level. + * Added the possibility to set the From: header through the the SIP dial string (populating the fromuser/fromdomain fields), complementing the [!dnid] option for the To: header that has existed since 1.6.0 (1d6b192). @@ -132,17 +134,22 @@ chan_sip chan_pjsip ------------------ * New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter - to the request URI and From URI if the user is determined to be a phone number. - * New 'moh_passthrough' endpoint setting. This will pass hold and unhold requests - through using SIP re-invites with sendonly and sendrecv accordingly. + to the request URI and From URI if the user is determined to be a phone + number. + + * New 'moh_passthrough' endpoint setting. This will pass hold and unhold + requests through using SIP re-invites with sendonly and sendrecv accordingly. + * Added the pjsip.conf system type disable_tcp_switch option. The option allows the user to disable switching from UDP to TCP transports described by RFC 3261 section 18.1.1. - * New 'line' and 'endpoint' options added on outbound registrations. This allows some - identifying information to be added to the Contact of the outbound registration. - If this information is present on messages received from the remote server - the message will automatically be associated with the configured endpoint on the - outbound registration. + + * New 'line' and 'endpoint' options added on outbound registrations. This + allows some identifying information to be added to the Contact of the + outbound registration. If this information is present on messages received + from the remote server the message will automatically be associated with the + configured endpoint on the outbound registration. + Core ------------------ @@ -190,6 +197,7 @@ Core context. If enabled then a hint will be automatically created with the name of the device. + Functions ------------------ @@ -208,8 +216,9 @@ CURL DTMF Features ------------------ * The transferdialattempts default value has been changed from 1 to 3. The - transferinvalidsound has been changed from "pbx-invalid" to "privacy-incorrect". - These were changed to make DTMF transfers be more user-friendly by default. + transferinvalidsound has been changed from "pbx-invalid" to + "privacy-incorrect". These were changed to make DTMF transfers be more + user-friendly by default. Resources @@ -250,6 +259,7 @@ res_pjsip_outbound_registration outbound registration, registration is retried at the given interval up to 'max_retries'. + CEL Backends ------------------ @@ -262,6 +272,7 @@ cel_pgsql configurable for cel_pgsql via the 'schema' in configuration file cel_pgsql.conf. + CDR Backends ------------------ @@ -272,15 +283,18 @@ cdr_adaptive_odbc names. This setting is configurable for cdr_adaptive_odbc via the quoted_identifiers in configuration file cdr_adaptive_odbc.conf. -Queue -------------------- - * Added field ReasonPause on QueueMemberStatus if set when paused, the reason - the queue member was paused. - * Added field LastPause on QueueMemberStatus for time when started the last - pause for a queue member. - * Show the time when started the last pause for queue member on CLI for command - 'queue show'. +cdr_odbc +------------------ + * Added a new configuration option, "newcdrcolumns", which enables use of the + post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'. +cdr_csv +------------------ + * Added a new configuration option, "newcdrcolumns", which enables use of the + post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'. + + +------------------------------------------------------------------------------ --- Functionality changes from Asterisk 13.9.0 to Asterisk 13.10.0 ----------- ------------------------------------------------------------------------------ -- cgit v1.2.3