> On March 12, 2015, 8:41 p.m., Matt Jordan wrote: > > 1. For this to go into Asterisk 13, tests will need to be provided that > > cover the new parameter. (Really, those tests should be written regardless) > > 2. The CHANGES file will need to get updated with the new option. > > rmudgett wrote: > Actually I'd prefer that the rpid_immediate option not exist at all and > the code it controls to just be removed. Sending connected line updates back > to the caller _before_ getting connected doesn't really make sense. This is > what the REDIRECTING information is supposed to be doing.
I disagree, providing connected line updates pre-answer makes perfect sense. Why would I want to know the connected-line name only after the call has been answered? That assumes the call even is answered, sending the connected-line information immediately allows the phone to include the name in it's call history. If no call-forwarding and/or re-addressing has taken place why would REDIRECTING be used? - gareth ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4473/#review14669 ----------------------------------------------------------- On March 10, 2015, 11:48 p.m., rmudgett wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4473/ > ----------------------------------------------------------- > > (Updated March 10, 2015, 11:48 p.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-24781 > https://issues.asterisk.org/jira/browse/ASTERISK-24781 > > > Repository: Asterisk > > > Description > ------- > > This patch builds on https://reviewboard.asterisk.org/r/4472/ > > When that patch is committed this patch reduces to simply adding the > rpid_immediate option to skip generating INVITE response messages as a > result of connected line updates. > > > 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. > > > Diffs > ----- > > /branches/13/res/res_pjsip_caller_id.c 432761 > /branches/13/res/res_pjsip/pjsip_configuration.c 432761 > /branches/13/res/res_pjsip.c 432761 > /branches/13/include/asterisk/res_pjsip.h 432761 > /branches/13/configs/samples/pjsip.conf.sample 432761 > /branches/13/channels/chan_pjsip.c 432761 > > Diff: https://reviewboard.asterisk.org/r/4473/diff/ > > > Testing > ------- > > * Ran the tests/channels/pjsip testsuite tests. They still pass. > > * Made a call chain as follows: 100 -> * -> * -> * -> 200. With the patch > there are no unnecessary messages. Without the patch there were several > "180 Ringing" messages sent back to the caller. > > > Thanks, > > rmudgett > >
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