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2006-06-16 snapshot

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git-svn-id: svn://192.168.202.10@9 3d104415-ff17-0410-8863-d5cf3c621b8a
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mattf
2006-07-07 15:27:12 +00:00
parent 7b2622b826
commit ca8e640156
30 changed files with 4284 additions and 450 deletions
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docs/APP_CONFERENCE.txt
+25 -8
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@@ -23,8 +23,8 @@ tested it in production yet.
If you want to use app_conference instead of meetme for VICIDIAL then follow these instructions:
cd /usr/src/asterisk
wget http://www.eflo.net/files/VD_app_conference_0.5.zip
unzip VD_app_conference_0.5.zip
wget http://www.eflo.net/files/VD_app_conference_0.6.zip
unzip VD_app_conference_0.6.zip
cd app_conference
make clean
make
@@ -34,25 +34,25 @@ make install
In your extensions.conf file you would replace these lines:
exten => 8600051,1,Meetme,8600051
exten => 8600052,1,Meetme,8600052
...
...continue through 8600100...
exten => 78600051,1,Meetme,8600051|q
exten => 78600052,1,Meetme,8600052|q
...
...continue through 78600100...
exten => 68600051,1,Meetme,8600051|mq
exten => 68600052,1,Meetme,8600052|mq
...
...continue through 68600100...
with these lines:
exten => 8600051,1,Conference(8600051)
exten => 8600052,1,Conference(8600052)
...
...continue through 8600100...
exten => 78600051,1,Conference(8600051|q)
exten => 78600052,1,Conference(8600052|q)
...
...continue through 78600100...
exten => 68600051,1,Conference(8600051|Lq)
exten => 68600052,1,Conference(8600052|Lq)
...continue through 68600100...
NOTES:
Here are the possible feature flags for this version of app_conference:
@@ -67,3 +67,20 @@ Here are the possible feature flags for this version of app_conference:
o i: use inband dtmf broadcast.
o t: use rfc dtmf signal broadcast.
COMPILE NOTES:
Some Makefile alterations may be necessary for your system:
From Alek Barkley:
I am running FC4 with latest updates ( 2.6.16 kernel )
My CPU is AMD K6 and GCC 4.0.2.
I edited Makefile in app_conference distribution.
To enter CPU architecture in my case -march=k6
========================================================
PROC = k6
# this is fun for x86
#CFLAGS += -march=pentium3 -msse -mfpmath=sse,387
CFLAGS += -march=k6
===========================================================
docs/README.txt
+7 -3
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@@ -1,5 +1,5 @@
+------------------------------------------------------------------------------+
| Asterisk GUI client - astguiclient - twenty-fourth public release 1.1.12 |
| Asterisk GUI client - astguiclient - twenty-fifth public release 1.1.12 |
| created by astGUIclient group <astguiclient@eflo.net> |
| project started 2003-10-06 http://sourceforge.net/projects/astguiclient/ |
+------------------------------------------------------------------------------+
@@ -681,13 +681,17 @@ UPGRADE NOTES:
* if upgrading from 1.1.10 run the upgrade_1.1.11.sql script in MySQL
(\. /home/cron/astguiclient/upgrade_1.1.11.sql)
1.1.12 - Twenty-Fifth public release - 2006-0X-XX
1.1.12 - Twenty-Fifth public release - 2006-06-XX
- Completely redesigned local call time system in VICIDIAL for more flexibility
- Added Drop-call and safe-harbor campaign and inbound options
- Added option for allowing wrapup time between calls in vicidial.php
- Multi-language translation rewritten for more flexibility and easy of use
- Added more user-permission options/restrictions for both agents and admins
- Many other small changes listed in the TODO.txt file
- Added option of using app_conference instead of meetme engine
- Added internal DNC list for Do-Not-Call entries across the system
- Added automatic lead recycling to call back Busy calls at definable intervals
- Added listID override and timezone lookup to all lead importing scripts
- Many other changes and bug fixes listed in the TODO.txt file
UPGRADE NOTES:
* if upgrading from 1.1.11 you need to:
- download the new version from the project website
docs/REQUIREMENTS.txt
+1
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@@ -7,6 +7,7 @@ Requirements for installing astGUIclient/VICIDIAL on an Asterisk server:
- Zap trunks(T1/E1 or PSTN lines), IAX2 trunks or SIP trunks required
- If using VOIP trunks and VICIDIAL auto-dial, you must have trunks registered
- You must have a zaptel timer like a PSTN/T1/E1 card or ztdummy
or you can use the experimental app_conference module
- Client phones can be SIP/IAX/Zap
- In extensions.conf you must use the 'o' Dial flag on your outbound extensions
- In extensions.conf you must have thee following line in your default context:
docs/SCRATCH_INSTALL.txt
+43 -25
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@@ -314,12 +314,30 @@ in and out as separate files
- you are done
LAME is an MP3 encoder used to convert audio files from WAV to MP3. We prefer GSM
usually, but some users have standardized on MP3 so they would need this utility
to be loaded to use that option.
*OPTIONAL*
2. Go to http://lame.sourceforge.net/ and download the lame package
- to install this directly on the command line type:
- cd /usr/local
-wget http://umn.dl.sourceforge.net/sourceforge/lame/lame-3.96.1.tar.gz
-gunzip lame-3.96.1.tar.gz
-tar xvf lame-3.96.1.tar
-cd lame-3.96.1
-./configure
-make
-make install
- you are done
mpg123 is the mp3 player for asterisk and is used for music-on-hold
applications. We do not usually use it on our system, but to be more
comprehensive we will give the directions to install it.
*OPTIONAL* (only if you want to use Astrisk's music-on-hold)
2. Go to http://www.mpg123.de/ and download the mpg123 package
3. Go to http://www.mpg123.de/ and download the mpg123 package
- to install this directly on the command line type:
- cd /usr/local
- wget http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz
@@ -340,7 +358,7 @@ the ability to log output and still attach to the screen that executed asterisk
originally.
*REQUIRED*
3. Go to http://www.gnu.org/software/screen/ and download the screen package
4. Go to http://www.gnu.org/software/screen/ and download the screen package
- to install this directly on the command line type:
- cd /usr/local
- wget http://ftp.gnu.org/gnu/screen/screen-4.0.2.tar.gz
@@ -360,7 +378,7 @@ graphical time-based scrolling graph. We use it to view how loaded the system is
and it visualizes load spikes very well
*OPTIONAL* (only for obsessive admins like me)
4. Go to http://www.daveltd.com/src/util/ttyload/ and download the ttyload
5. Go to http://www.daveltd.com/src/util/ttyload/ and download the ttyload
package
- to install this directly on the command line type:
- cd /usr/local
@@ -378,7 +396,7 @@ with the time of a master server somewhere in the world. We use it to make sure
the time is the same on our client computers and our servers.
*MANDITORY FOR VICIDIAL USERS* (install on server and all clients)
5. Go to http://www.ntp.org/ and download the ntpd package
6. Go to http://www.ntp.org/ and download the ntpd package
- to install this directly on the command line type:
- cd /usr/local
- wget http://www.eecis.udel.edu/~ntp/ntp_spool/ntp4/ntp-stable-4.2.0a-20060224.tar.gz
@@ -401,7 +419,7 @@ aftpd is a tftp server which is a file transfer method used by many VOIP phones
to get their firmware and/or configuration files. This is optional.
*OPTIONAL* (only if you want it for a VOIP phone you are going to use)
6. Go to ftp://ftp.mamalinux.com/pub/atftp and download the aftpd package
7. Go to ftp://ftp.mamalinux.com/pub/atftp and download the aftpd package
- to install this directly on the command line type:
- cd /usr/local
- ncftpget ftp://ftp.mamalinux.com/pub/atftp/atftp-0.7.tar.gz
@@ -421,7 +439,7 @@ iftop is a good console bandwidth visualization tool that shows you active
connections, where they are going to/from and how much of your precious bandwidth
they are using. *OPTIONAL*
7. Go to http://www.ex-parrot.com/~pdw/iftop/ and download the package
8. Go to http://www.ex-parrot.com/~pdw/iftop/ and download the package
- to install this directly on the command line type:
- cd /usr/local
- wget http://www.tcpdump.org/release/libpcap-0.8.3.tar.gz
@@ -448,7 +466,7 @@ within PNG files simply by creating a config file and a data file. We use this
package along with the included PHP script to generate server performance graphs
that can be displayed real-time on a web page. *OPTIONAL*
8. Go to http://ploticus.sourceforge.net/ and download the package
9. Go to http://ploticus.sourceforge.net/ and download the package
- to install this directly on the command line type:
NOTE: you may have to edit the Makefile to remove X11 if you don't have it
- cd /usr/local
@@ -462,7 +480,7 @@ that can be displayed real-time on a web page. *OPTIONAL*
- you are done
NOTE: for the graphics to work on the AST_server_performance page you will
need the 'pl' script to be linked or copied into your htdocs/vicidial/ploticus directory
NOTE: you may need to edit the Makefile for ploticus if you do not have X11
MySQL is a fast database system that is very easy to integrate with any
application. You can either install it on the local Asterisk system or have one
@@ -473,7 +491,7 @@ so we will need to install mysql on this machine.
on another machine)
NOTE: a minimum of MySQL 4.0.X is required
9. Go to http://www.mysql.com/ and download the mysql package
10. Go to http://www.mysql.com/ and download the mysql package
- to install this directly on the command line type:
- cd /usr/local
- wget http://mirror.trouble-free.net/mysql_mirror/Downloads/MySQL-4.0/mysql-4.0.26.tar.gz
@@ -533,7 +551,7 @@ updated on your system, so we're going to install a new version now.
*OPTIONAL* (only updated as a precaution, not manditory)
[NOTE: newer zlib is needed before installing]
10. Go to http://www.openssh.org/ and download the linux source for openssh
11. Go to http://www.openssh.org/ and download the linux source for openssh
- to install this directly on the command line type:
- cd /usr/local
- wget http://www.zlib.net/zlib-1.2.3.tar.gz
@@ -559,7 +577,7 @@ locally and have it work with apache, you need it installed on your machine
*OPTIONAL* (only install openssl if you want to use SSL secured web pages on
your locally installed copy of Apache web server)
11. Go to http://www.openssl.org/ and download the linux source for openssl
12. Go to http://www.openssl.org/ and download the linux source for openssl
- to install this directly on the command line type:
- cd /usr/local
- wget http://www.openssl.org/source/openssl-0.9.8b.tar.gz
@@ -578,7 +596,7 @@ functionalities we need to have Apache and PHP installed on this machine.
*OPTIONAL* (only install Apache and PHP locally if you don't want to use an
installation on another machine)
12. Go to http://www.apache.org/ and download the apache unix source
13. Go to http://www.apache.org/ and download the apache unix source
Go to http://www.php.net/ and download the php unix source code
- to install this directly on the command line type:
- cd /usr/local
@@ -616,7 +634,7 @@ the load of your web traffic across many servers. If you are running more than
another cheap web server to handle the extra load.
*OPTIONAL*
13. Go to http://balance.sourceforge.net to download the most recent source version
14. Go to http://balance.sourceforge.net to download the most recent source version
- to install this directly on the command line type:
- cd /usr/local
- wget http://voxel.dl.sourceforge.net/sourceforge/balance/balance-3.34.tgz
@@ -636,7 +654,7 @@ you want to use the latest development code of Asterisk you will need to have
this loaded on your system.
*OPTIONAL*
14. Go to http://subversion.tigris.org to download the most recent source version
15. Go to http://subversion.tigris.org to download the most recent source version
- to install this directly on the command line type:
- cd /usr/local
- wget http://subversion.tigris.org/downloads/subversion-1.3.0.tar.gz
@@ -653,10 +671,10 @@ PHASE 4: INSTALLING ASTERISK
OK, all the prep work is done, now it's time to start having fun with Asterisk.
There are four basic ways to install Asterisk, an official release(at the time of
this writing the official release is 1.2.8), the CVS_v1-0 branch(1.0
this writing the official release is 1.2.9.1), the CVS_v1-0 branch(1.0
with bug fixes not yet packaged and released) and the SVN_DEV version(development
branch). Because there are bugs that affect astGUIclient in releases 1.0.8 and
1.0.9, we recommend using Asterisk 1.2.8 or 1.0.10. These instructions are how we
1.0.9, we recommend using Asterisk 1.2.9.1 or 1.0.11.1. These instructions are how we
get our Asterisk system with it's T1 line installed with our 2 SIP VOIP devices
and one IAX2 softphone.
@@ -665,27 +683,27 @@ CVS_v1-0_Stable branch where the issues are fixed, or patching your 1.0.8/1.0.9
with the following patch:
(http://astguiclient.sourceforge.net/experimental_code/localmasq.patch)
- If you do patch your system make sure you put the asterisk version
field for the server on the admin pages as '1.0.10'
field for the server on the admin pages as '1.0.11.1'
1. follow these command line steps:
- mkdir /usr/src/asterisk
- cd /usr/src/asterisk
A. if you want 1.2 release (reliable with new features):
- wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.8.tar.gz
- wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.9.1.tar.gz
- wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.6.tar.gz
- wget http://ftp.digium.com/pub/libpri/releases/libpri-1.2.3.tar.gz
- gunzip asterisk-1.2.8.tar.gz
- tar xvf asterisk-1.2.8.tar
- gunzip asterisk-1.2.9.1.tar.gz
- tar xvf asterisk-1.2.9.1.tar
- gunzip zaptel-1.2.6.tar.gz
- tar xvf zaptel-1.2.6.tar
- gunzip libpri-1.2.3.tar.gz
- tar xvf libpri-1.2.3.tar
B. If you want latest 1.0 tree stable release:
- wget http://ftp.digium.com/pub/asterisk/old-releases/asterisk-1.0.10.tar.gz
- wget http://ftp.digium.com/pub/asterisk/old-releases/asterisk-1.0.11.1.tar.gz
- wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.0.10.tar.gz
- wget http://ftp.digium.com/pub/libpri/releases/libpri-1.0.9.tar.gz
- gunzip asterisk-1.0.10.tar.gz
- tar xvf asterisk-1.0.10.tar
- gunzip asterisk-1.0.11.1.tar.gz
- tar xvf asterisk-1.0.11.1.tar
- gunzip zaptel-1.0.10.tar.gz
- tar xvf zaptel-1.0.10.tar
- gunzip libpri-1.0.9.tar.gz
@@ -720,7 +738,7 @@ with the following patch:
- make clean
- make
- make install
- cd ../asterisk-1.2.8
- cd ../asterisk-1.2.9.1
- (1.2 tree) If you want to include Answering Machine Detection ability
you will need to download app_amd.c and amd.conf and alter the
apps/Makefile to compile it properly
@@ -1544,7 +1562,7 @@ effected extension groups are show):
##### you will have problems if it is not in your dialplan!
;exten => h,1,AGI(call_log.agi,${EXTEN}) ; use regular AGI for older versions
exten => h,1,DeadAGI(call_log.agi,${EXTEN}) ; DeadAGI is new
exten => h,2,DeadAGI(VD_hangup.agi,${EXTEN})
exten => h,2,DeadAGI(VD_hangup.agi,PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
; Extension 3429 - Inbound 800 number (1-800-555-3429)
exten => _**3429,1,Ringing
docs/TODO.txt
+49 -18
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@@ -74,16 +74,40 @@
- DONE Fix active calls stats for multiple CLOSER campaigns in Realtime Campaign screen.
- DONE Add internal DNC list of phone numbers(vicidial_dnc table) that would be scrubbed against by the AST_VDhopper.pl script while putting leads into the hopper to be dialed and it would kick the internal DNC matches out with a special status as DNCL(Do-Not-Call-Load) so they would be in the system but unable to be dialed. Would need to add a mechanism to add a lead to this list upon agent-dispo of a call as DNC within vicidial.php. Would need to create a new admin page section to manually add leads to DNC list, ADD TO DNC link in the LISTS section. Also, would add a parameter to vicidial_campaigns(dnc_list_enabled ENUM('Y','N')) where a campaign could be exempted from the system's DNC list restrictions if set to N(would not be active by default 'N').
- DONE Add option for PRI T1 system usage with VICIDIAL outbound to automatically grab Busy and Disconnect information from PRI call termincation codes and use those for NA dispositions. VD_hangup.agi and AST_VDauto_dial.pl modified.
- DONE Remove gmt time validation of leads from AST_VDauto-dial because AST_VDhopper already removes leads that are outside of the appropriate range of GMT offsets when it is run every minute. Removing this redundancy should speed up the dialing slightly.
- DONE Fix VD_hangup to work with CLOSER transfer-from-fronter calls, different log contents than from CLOSER inbound call.
- DONE Add option for disposition-based lead recycling on an automated and timed basis per campaign. It will be some sort of method to be able to auto-insert Busy/Ring-no-answer/other-NA calls back into the hopper after a certain timeout. there will be a minimum of 120 seconds before the lead will be allowed to go back into the hopper and a lead can only go back into the hopper a maximum of 10 times before the list would need to be reset. The vicidial_list.called_since_last_reset field needs to be altered to allow for counting of recycle attempts. All scripts need to be able to keep the state of call_since_last_reset. A new table: vicidial_lead_recycle that will function somewhat like HotKeys on the campaign screen. admin.php and AST_VDhopper.pl will need to be modified.
- DONE Change ADMIN_area_code_populate.pl script to ignore the header row of the phone codes gmt file DB import
- DONE Add database GMT lookup to the VICIDIAL_IN_new_leads_file.pl lead loader lead import process.
- DONE Change ADMIN_adjust_GMTnow_on_leads.pl script to use database instead of flat text file.
- DONE Fix display issues with LIST MODIFY page on admin.php for statuses because of lead recycling and with GMT offset for positive GMT offsets.
- DONE Add listID override feature to the VICIDIAL_IN_new_leads_file.pl lead loader lead import process to force all leads being loaded into the same list_id.
- DONE Add listID override feature to the basic web-based lead loader lead import process to force all leads being loaded into the same list_id.
- DONE Add database GMT lookup to the basic web-based lead loader lead import process.
- DONE Add listID override feature to the super web-based lead loader lead import process to force all leads being loaded into the same list_id.
- DONE Add database GMT lookup to the super web-based lead loader lead import process. This one is more difficult because of the many scripts and sections used to parse and insert leads: listloader_super.pl, listloader.pl, new_listloader_superL.php
- DONE standardize the query results variable in PHP scripts to $rslt.
- HIGH Add internal DNC list of phone numbers(vicidial_dnc table) that would be scrubbed against by the AST_VDhopper.pl script while putting leads into the hopper to be dialed and it would kick the internal DNC matches out with a special status as DNCL(Do-Not-Call-Load) so they would be in the system but unable to be dialed. Would need to add a mechanism to add a lead to this list upon agent-dispo of a call as DNC within vicidial.php. Would need to create a new admin page section to manually add leads to DNC list. Also, would add a parameter to vicidial_campaigns(dnc_list_enabled ENUM('Y','N')) where a campaign could be exempted from the system's DNC list restrictions(would be active by default 'Y').
- HIGH Add optional set of scripts for PRI system usage with VICIDIAL outbound to automatically grab Busy and disconnect information from PRI call termincation codes and use those for NA dispositions. The problem with these is that they are not 100% accurate.
- HIGH Add option for disposition-based lead recycling on an automated and timed basis per campaign. It will be some sort of method to be able to auto-insert Busy/Ring-no-answer/other-NA calls back into the hopper after a certain timeout. Not sure on a method or formula yet.
- HIGH Fix call_log entries for Zap client phones where call information is in different places than if calls are placed from SIP or IAX client phones. call_log AGI scripts.
@@ -91,9 +115,11 @@
- MEDIUM fix MySQL error on campaign modify page for dialable leads when no active lists for campaign.
- MEDIUM add option to disable detailed logging for VICIDIAL and astGUIclient server apps.
- MEDIUM add new ALLFORCE to recording method for campaigns. will force recording of all calls and disable the button for agents no matter their setting.
- MEDIUM add new ALLFORCE to recording method for campaigns. will force recording of all calls and disable the button for agents no matter their vicidial_users setting.
- MEDIUM add a basic predictive/adaptive algorithm to VICIDIAL to automatically speed-up and slow-down dialing by altering the dial_level at regular intervals. Would need more settings in the vicidial_campaigns table for adaptive like maximum lines per agent, dropped call ratio, dropped call rule to guide the predictive app. Would also need to change dial_level to be a DOUBLE or VARCHAR field to allow for different fractional increments. The admin.php page would have to be altered to ignore auto_dial_level changes when adaptive dialing is activated. The minimum dial_level would be set to '1' for adaptive dialing
auto_dial_level VARCHAR(6), # allow for just about anything. precision decimal too
@@ -102,13 +128,19 @@
adaptive_dropped_percentage SMALLINT(3), # percentage of accaptable dropped calls
adaptive_dropped_rule ENUM('HARD_LIMIT','TAPERED','AVERAGE'), # method for adhearing to % limit
- MEDIUM room manager documentation - manual. Probably a free 2 chapter black and white manual and a full color print book available for sale.
- MEDIUM Rewrite the inbound and closer call handling to allow for music on hold per in-group and a single queue app instead of multiple AGIs each running their own queries. Currently with large queues and long wait times a lot of load is generated with calls on hold in the queue. Switching to a central queue application would reduce the load and speed things up.
- MEDIUM Add optional field to phones table to allow for different recording file names in astguiclient.php, Similar to vicidial options.
- MEDIUM make 3rd party consultative transfers work for Local extensions, also for internal/local transfer to vicidial closers if possible.
- MEDIUM Add ChanSpy functionality to listen-in on VOIP channels in astguiclient.php. Maybe not since Chanspy can cause problems like crashing the Asterisk server.
- MEDIUM Attempt to make some kind of holiday schedule function for national and state holidays that would be able block out states on specific no-call days. This is rather difficult because of the way some holidays are scheduled, I.E. Easter.
- MEDIUM Create a script to allow vicidial agents to dial into an AGI script that would ask for user/pass and then pass them onto their vicidial.php-defined sessionid.
- MEDIUM Add ChanSpy functionality to listen-in on VOIP channels in astguiclient.php. Maybe not since Chanspy can cause problems like crashing the Asterisk server. Need to do more testing.
- MEDIUM Create a script to allow vicidial agents to dial into an AGI script that would ask for user/pass and then place them onto their vicidial.php-defined sessionid.
- MEDIUM add link in astguiclient to login and logout from Asterisk Queues
@@ -117,27 +149,23 @@
- MEDIUM add new tab to astguiclient.php script to show status of Asterisk queues
- MEDIUM Convert all scripts to DBD::MySQL perl library in place of Net::MySQL. DBD::MySQL is faster because it uses the MySQL client libs on the machine. One problem is that you need the mysqlclientlibs installed on every
Asterisk server and the default installation of only the MySQL client libs will result in having to do a force install of cpan DBD::MySQL(because it tries to test with the local DB). Not to mention all of the code changes that would need to be made. There are also MySQL client licensing issues for certain installation circumstances(Net::MySQL doesn't use mysql client libs so it doesn't have those issues) but I really don't want to get into that discussion.
Asterisk server and the default installation of only the MySQL client libs will result in having to do a force install of cpan DBD::MySQL(because it tries to test with the local DB). Not to mention all of the code changes that would need to be made. There are also MySQL client licensing issues for certain installation circumstances(Net::MySQL doesn't use mysql client libs so it doesn't have those issues) but I really don't want to get into that discussion. We will start by converting at least one perl script for each release until we have converted then all, then we will no longer support Net::MySQL. This will probably take some time, but the scripts that are added to the DBI-scripts folder will be drop-in replacements for the Net versions that are installed by default.
- MEDIUM add ability to call CLOSER campaigns "BLEND" or "INBND". would require many script changes from server side to client side, not set in stone yet, but considering it.
- MEDIUM add new method for capping the maximum number of attempts to try leads of a specific status. This would be set up like HotKeys are, per campaign where you would select the status and define the maximum number of attempts for that type of status for that campaign and the VDhopper wouldn't allow that kind of lead into the hopper if it was over the max value of call_count.(having filters reduced the priority of this item since you can do a lot of custom coding to duplicate this feature in vicidial_lead_filters)
- MEDIUM add some kind of intelligent call-routing to astguiclient so that you can auto-forward calls to another extension from within astguiclient.php instead of using your phone or something like Queues which can mess up other things. This will need it's own table in the DB and probably some extensions.conf entries and an AGI script or two to do the actual call routing.
- MEDIUM add a new script for using SoX to analyze the first 5 seconds of a ring-time recording within 15 minutes of the NA call ending and detecting if it has the SIT tone at the beginning of it so the lead can be taken out of the list as a DC
- MEDIUM add skills-based routing to give the ability for a manager to set an agent at a higher or lower level per campaign or in-group. Defaults to level 0 possible values will be -9 through 9
- MEDIUM add all possible fields to the SCRIPT tab to auto-populate like in-group for inbound/closer, campaign and others
- MEDIUM add ability to specify list_id in lead loader at time of picking fields
- MEDIUM add ability to scrub leads being imported in the lead loader against the vicidial_dnc list as they are being imported.
- MEDIUM add ability upon vicidial.php login to take a number to dial as a password and a reserved word(something like MANUAL in all caps) that would allow the use of an EXTERNAL phone without setting it up as a aasterisk.phones entry.
- MEDIUM add ability upon vicidial.php login to take a number to dial as a password and a reserved word(something like MANUAL in all caps) that would allow the use of an EXTERNAL phone without setting it up as an asterisk.phones entry.
- MEDIUM rebuild the vdremote.php remote agents pages to use AJAX(PHP/Javascript/XMLHTTPRequest) for real-time popups of call data.
- MEDIUM allow custom audio welcome message for vicidial campaigns.
- MEDIUM allow custom audio welcome message for vicidial campaigns. This might be an app_conference only feature.
- MEDIUM create a cron script and method for creating new extensions in the sip.conf/iax.conf files as well as dialplan entries(extensions.conf). This will require more fields in the phones table to account for variables available to the protocols. The conf files would need to be altered before to have a flag for the beginning and ending of the auto-generated content. This would also necessitate a TEXT area for the extensions.conf content(that would be default generated but could be manually altered). The cron script would be run every 5 minutes or so to see if any phones had altered content, and if so generate new auto-content and do a 'reload' on the asterisk server to activate the new content.
@@ -151,8 +179,6 @@ Asterisk server and the default installation of only the MySQL client libs will
- MEDIUM Add method of picking up astGUIclient parked calls by only dialing digits on a regular extension without using the astGUIclient.
- MEDIUM room manager documentation - manual. Probably a free text-only manual and a full color print book available for sale
- MEDIUM admin/installer/maintainer documentation as well as troubleshooting guide
- MEDIUM Run some lab-style load tests on VICIDIAL and show recommended configurations as well as "highest recommended" settings
@@ -167,8 +193,13 @@ Asterisk server and the default installation of only the MySQL client libs will
- LOW Revisit possibility of using something other than CallerIDname to tag a call process, probably a long shot without altering the Asterisk code, but that may be an option.
- LOW add new method for capping the maximum number of attempts to try leads of a specific status. This would be set up like HotKeys are, per campaign where you would select the status and define the maximum number of attempts for that type of status for that campaign and the VDhopper wouldn't allow that kind of lead into the hopper if it was over the max value of call_count.(having filters reduced the priority of this item since you can do a lot of custom coding to duplicate this feature in vicidial_lead_filters)
- LOW add a new script for using SoX to analyze the first 5 seconds of a ring-time recording within 15 minutes of the NA call ending and detecting if it has the SIT tone at the beginning of it so the lead can be taken out of the list as a DC. adding PRI hangup codes has reduced priority for this item.
- ON-HOLD We are also looking at creating an inbound-agent-specific GUI that would ideally work with Asterisk queues but that is not very likely given the current lack of events being given off by queues(this is much less likely to be built now that VICIDIAL takes inbound calls)
- ON-HOLD add a way of checking that there is no more than 1 channel in the meetme room with the agent before another call to that agent. Currently this happens less than 0.1% of the time(mostly on heavily loaded systems), but another check might change that to zero.
- ON-HOLD Localization of each server based on Intl. dial codes and/or country codes. very difficult to do this based on dial-codes, if used in other country, best to not use the adjust_gmt script.