Fixed issue with using CALLID at beginning of recording filename
Added partial experimantal Asterisk 13 compatibility
git-svn-id: svn://192.168.202.10@2817 3d104415-ff17-0410-8863-d5cf3c621b8a
reports that are not included in Vicidial
Added Status Groups and the ability to override Campaign Statuses on a per-In-Group or per-List basis
Added Minimum and Maximum Seconds options for all status settings. This allows you to prevent the use of a status by an agent if they have
talked to a customer less than X seconds or more than Y seconds.
Fixed issue with email Customer Gone message in agent screen
Added Asterisk 1.8 and higher voicemail.conf sample file
Added listed support in install.pl for Asterisk 11
git-svn-id: svn://192.168.202.10@2403 3d104415-ff17-0410-8863-d5cf3c621b8a
Fixed several small bugs
Added new optional extras/did_recent_call.php script for use with DID Filter URL
git-svn-id: svn://192.168.202.10@2042 3d104415-ff17-0410-8863-d5cf3c621b8a
Altered agi-phone_monitor script to not hangup after agent hangs up customer, requires small change to custom dialplan entries
Added new Dial Log report
Added Disable Auto Dial system setting
git-svn-id: svn://192.168.202.10@1999 3d104415-ff17-0410-8863-d5cf3c621b8a
Added User Group multi-selection option to the real-time report
Added HTML bar chart display options to most of the reports
Removed password logging from admin scripts failed login attempts
Made webroot-writable system setting function the same on all admin scripts
Fixed list lastcalldate for manual dial calls
git-svn-id: svn://192.168.202.10@1789 3d104415-ff17-0410-8863-d5cf3c621b8a
Added voicemail emailbody to voicemail.conf that will show callerIDnumber after the name
git-svn-id: svn://192.168.202.10@1710 3d104415-ff17-0410-8863-d5cf3c621b8a
Added preprocess leads in script to filter leads and generate standard format lead files before being imported
changed realtime links to new realtime report
git-svn-id: svn://192.168.202.10@1577 3d104415-ff17-0410-8863-d5cf3c621b8a
- Added MIX directory auto mixing option to CRON audio step 1
- Added outbound callerIDnumber to DROP and other AGI transfers for in-groups
- Changed called_count to increment on alt-dial calls
- Fixed variable mis-assignment in several AGI and bin scripts
- Fixed Pause code issue with QueueMetrics
git-svn-id: svn://192.168.202.10@1254 3d104415-ff17-0410-8863-d5cf3c621b8a
Added a process to update statuses in the vicidial_list table by batch, with web report
git-svn-id: svn://192.168.202.10@1169 3d104415-ff17-0410-8863-d5cf3c621b8a
Added script to allow for batch loading of leads into Vtiger
Several other Vtiger integration changes
see docs/VTIGER.txt for more info
git-svn-id: svn://192.168.202.10@1039 3d104415-ff17-0410-8863-d5cf3c621b8a
Made install.pl install features.conf.sample if the user selects to install Asterisk config samples.
git-svn-id: svn://192.168.202.10@813 3d104415-ff17-0410-8863-d5cf3c621b8a
Also, updated conf file generation utility in install.pl and the default conf file examples with the new blind monitoring settings and reduced-complexity multi-server extens
git-svn-id: svn://192.168.202.10@785 3d104415-ff17-0410-8863-d5cf3c621b8a
added hopper insertion logging to AST_VDhopper scripts
added better DMTF filtering in agi-IVR script as well as a one hour timeout
git-svn-id: svn://192.168.202.10@773 3d104415-ff17-0410-8863-d5cf3c621b8a
mattf