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branches/12/res/res_pjsip_send_to_voicemail.c <https://reviewboard.asterisk.org/r/3245/#comment20573> You don't need to pass size in, it's optional. branches/12/res/res_pjsip_send_to_voicemail.c <https://reviewboard.asterisk.org/r/3245/#comment20575> Silly use of RAII_VAR. branches/12/res/res_pjsip_send_to_voicemail.c <https://reviewboard.asterisk.org/r/3245/#comment20577> I think it would also be useful to have the endpoint name in here as well. branches/12/res/res_pjsip_send_to_voicemail.c <https://reviewboard.asterisk.org/r/3245/#comment20574> No need for ast_channel_cleanup, just use ast_channel_unref. branches/12/res/res_pjsip_send_to_voicemail.c <https://reviewboard.asterisk.org/r/3245/#comment20572> Pfft, using RAII_VAR here is silly. - Joshua Colp On Feb. 20, 2014, 11:01 p.m., Kevin Harwell wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3245/ > ----------------------------------------------------------- > > (Updated Feb. 20, 2014, 11:01 p.m.) > > > Review request for Asterisk Developers. > > > Repository: Asterisk > > > Description > ------- > > Added the ability for transferring directly to voicemail on digium phones. > Added a new module that checks for the presence of a custom header and/or > diversion header within a sip REFER. If either is found and they specify a > sending to voicemail action then variables are added to the channel allowing > the user access to them in the dialplan. Dialplan can then be written that > branches based upon these values allowing, for instace, for a single number > to be used for dialing and/or accessing voicemail directly. > > Also fixed a problem where the PJSIP_HEADER function was allowing non pjsip > channels through (checked to make sure it has the correct channel type before > proceeding). > > > Diffs > ----- > > branches/12/res/res_pjsip_send_to_voicemail.c PRE-CREATION > branches/12/res/res_pjsip_header_funcs.c 408442 > > Diff: https://reviewboard.asterisk.org/r/3245/diff/ > > > Testing > ------- > > Ran various scenarios manually with digium phones to make sure user were able > to transfer callers directly to voicemail. Also wrote a testsuite test that > checks the presence of those headers/values in the dialplan: > https://reviewboard.asterisk.org/r/3246/ > > > Thanks, > > Kevin Harwell > >
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