> On March 24, 2015, 10:32 p.m., rnewton wrote: > > Tested with 4488. Pretty much worked fine. > > > > You will need to add a timing interface to modules.conf. > > res_timing_timerfd.so is probably fine.
When testing with 4488 I ran through the following tests: *patch1 - internal stuff* internal user to internal user (audio): PASS internal user to internal user (voicemail-unavail): PASS internal user to internal user (voicemail-busy): PASS internal user can check voicemail: PASS deskphone displays MWI indication: PASS *patch2 - outside connectivity* registration to ITSP(DCS) comes up: PASS internal user dials out ITSP with 7 digit number: PASS internal user dials out ITSP with 10 digit number: PASS internal user dials out ITSP with 10+1 digit number: PASS internal user dials main IVR internally: PASS restricted number patterns work successfully: PASS inbound calls reach the main IVR: PASS external user can reach external voicemail feature via DID: PASS external users can dial internal users directly via DID match: PASS *patch3 - queues with external and internal access* sales queue reached internally: PASS externally: PASS sales queue rings Terry, Garnet and Franny in ring-all: PASS customer advocate queue reached internally: PASS externally: PASS customer advocate Queue rings Maria, Dusty and Tommie in ring-all: PASS *patch4 - conferences* employee conference can be dialed by internal users: PASS at least two parties in employee conference with audio: PASS customer conference can be dialed into by internal user and transfer in external users: PASS at least two parties, including an external party in customer conference with audio: PASS *ALL PATCHES COMBINED* All IVR options go to the correct feature/extension: PASS CDR Master.csv does not record any intra-office calls: PASS CDR Master.csv records calls to/from the ITSP account: PASS - rnewton ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4504/#review14821 ----------------------------------------------------------- On March 16, 2015, 5:48 p.m., Jonathan Rose wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4504/ > ----------------------------------------------------------- > > (Updated March 16, 2015, 5:48 p.m.) > > > Review request for Asterisk Developers and rnewton. > > > Repository: Asterisk > > > Description > ------- > > From: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company > > SAC requires two conference rooms, one for use by employees only and one for > use by employees and customers (outside connectivity still needs to be > established so that 555-6500 can be added and customers can actually dial > into said conference) > > > Diffs > ----- > > /branches/13/configs/basic-pbx/modules.conf 432996 > /branches/13/configs/basic-pbx/extensions.conf 432996 > /branches/13/configs/basic-pbx/confbridge.conf PRE-CREATION > > Diff: https://reviewboard.asterisk.org/r/4504/diff/ > > > Testing > ------- > > Made sure app_confbridge loaded and internal users were able to dial into the > conferences. > > > Thanks, > > Jonathan Rose > >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
