diff options
| author | Richard Mudgett <rmudgett@digium.com> | 2015-03-24 19:41:36 +0000 |
|---|---|---|
| committer | Richard Mudgett <rmudgett@digium.com> | 2015-03-24 19:41:36 +0000 |
| commit | 4c2fc5b81103f7942277bf3dd6481742500ddb2d (patch) | |
| tree | 5049be51e380fb66367d0efe47fed02c71642940 /CHANGES | |
| parent | 60f01520e74c7c043e506dfb65c787148205fa93 (diff) | |
chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages.
Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens. If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.
Consequences of these unnecessary messages:
* The caller can start hearing ringback before the far end even gets the
call.
* Many phones tend to grab the first connected line information and refuse
to update the display if it changes. The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.
When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled. When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.
* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages. The default is "no" to disable sending the
unnecessary messages.
ASTERISK-24781 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4473/
........
Merged revisions 433338 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 12 |
1 files changed, 12 insertions, 0 deletions
@@ -107,6 +107,18 @@ res_musiconhold application (e.g. Queue or Dial) specified music. ------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------ +------------------------------------------------------------------------------ + +chan_pjsip +------------------ + * New 'rpid_immediate' option to control if connected line update information + goes to the caller immediately or waits for another reason to send the + connected line information update. See the online option documentation for + more information. Defaults to 'no' as setting it to 'yes' can result in + many unnecessary messages being sent to the caller. + +------------------------------------------------------------------------------ --- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------ ------------------------------------------------------------------------------ |