vicidial.org

We just posted 2.7 RC 1 to sourceforge

https://sourceforge.net/projects/astguiclient/files/


Please test it out if you can and we should be able to do a full release soon.

There are dozens of new features and bug fixes in this RC, and at this point this code is installed on several dozen of our client sites and is proven to be very stable.

Here is a list of the changes since 2.4:

1. Added the ability for an agent to place manual dial calls out as if they were
in-group inbound calls. Also allows for no-dial calls that are not
actually placed out the telco, but are logged as calls. For more
information see the "In-Group Manual Dial" options in Campaign Detail

2. Added Dial In-Group CallerID override to In-Groups Modify screen, allowing
for the callerid to automatically be set per in-group on agent manual
dial in-group calls

3. Added transfer call to answering machine message with hotkey option,
currently hard-coded to these two statuses: (LTMG or XFTAMM)

4. Added User Group Login Report

5. Added Safe Harbor Audio Field to campaign options which allows for a lead
field to be used as an override safe harbor audio on a per-lead basis

6. Added Japanese agent web interface translation

7. Added ability to qualify callers before they go through a Call Menu by using
a SQL fragment(like in Filters) that you define in a Call Menu. Must be
enabled in System Settings. Calls must be passed from an outbound
campaign or an in-group.

8. Added ability to download the Max System Stats and display by date range.

9. Added 'recording' and 'webserver' functions to the agent API

10. Added add_group_alias non-agent API function

11. Added System Settings option to disable list counts in admin web screens

12. Added callback user transfer page, linked from Admin Utilities in Reports

13. Added non-loading webphone options and API agent call and webphone url
functions

14. Added outbound call logging in vicidial_dial_log table

15. Added user_group_status and in_group_status functions to the non-agent API.

16. Added update_log_entry function to the non-agent API, which allows the
updating of the status of an inbound or outbound call log entry.

17. Added ability to set an audio store file for the voicemail greeting message

18. Added ability to delete audio files from the audio store

19. Added logging of svn version info in the database at server install/update

20. Added more validation of agent sessions to prevent double-logins

21. Added campaign option to show a "Pause After Next Call" link to the agent
screen below the Pause/Resume buttons

22. Added campaign option to set owner field to user if the owner field is empty

23. Added Lead Tools web page from Admin Utilities. Allows for some basic lead
management through the web interface.

24. Added recording filename variable INGROUP for the In-Group ID of a call

25. Added QC functionality contributed by PoundTeam and Affinity On-Call.
To enable, go to Admin -> System Settings and set "QC Features Active"
to "1", then submit.

26. Added QM Socket-Send functionality for custom queue_log record sending.

27. Added inbound call time holiday functions.

28. Added list expiration date feature

29. Added Other Campaign DNC List campaign option

30. Added Enhanced Disconnect logging for outside the USA, disabled by default

31. Added a campaign feature to allow an agent to search for a lead while on an
inbound call and have their active lead change to a search result lead
that they select.

32. Added inbound email to queue to agent features. To enable, you must add an
"E" keepalive to ONLY ONE of your servers. You will also need to enable
email in System Settings.

33. Added Asterisk 1.8 support. This is in the BETA testing phase currently,
and is not recommended for use in production at this time. If you are
upgrading an existing system, please make note of the many conf file
changes(extensions.conf and manager.conf) that will need to be made for
Vicidial to function properly while interfacing with Asterisk 1.8. Also,
make sure that the Admin -> Servers -> modify server "Asterisk Version"
setting is set properly for the version you have installed on that
server.

34. Added campaign options to send answering machine detected calls to a
Call Menu or In-Group.

35. Added System Settings option to disable Add/Copy admin functions for level
8 users.

36. Added ability to use .agi scripts in Call Menu prompt. Also added
cm_date.agi script to say date with options

37. Converted ereg to preg in the agc agent interface PHP scripts

38. Added agi-phone_monitor.agi script to allow monitoring of agents using only
their user IDs or phone extensions and no web browser needed.
You can add as an AGI to a Call Menu, or just an extension transfer

39. Added user_group as a script, form, webform, start_call_url and
dispo_call_url variable.

40. Added agent_status function to the non-agent API

41. Added method for logging SIP error codes, also display of those codes in the
agent screen. The logging goes to the vicidial_carrier_log and
vicidial_dial_log tables and includes both the SIP code and the reason
(i.e. 503 "Service Unavailable"). To get this to work, you need to patch
Asterisk with the extras/sipcausecodelog_1.4.39.1.patch file for
Asterisk 1.4, and for Asterisk 1.8 the patch is pending.

42. Added admin report logging to all reports on the system, and added the
"Admin Report Log Viewer" linked from the "Admin Utilities" page which
allows you to see the reports that have been run on the system.

43. Added option to the AST_DB_dead_cb_purge.pl script to remove duplicate
callback entries for a single lead, keeping the newest one only.

44. Added NANPA areacode and prefix phone number validation and timezone setting
options to the admin web lead loader, the Non-Agent API and the CLI lead
loader. The NANPA prefix list will give you more accurate time zone
encoding of your USA and Canadian lead files within ViciDial. The NANPA
list is available for purchase at the vicidial.org web store:
http://www.vicidial.org/store.php#NANPA

45. Added a NOINT flag option to the Call Menu Prompt so the playback cannot be
interrupted with a key press by the caller, the NOINT should be put
directly in front of the prompt filename, but should not be a part of
the filename, it is a special flag for the system.

45. Added Survey No Response DROP option to send no-response calls to the
campaign drop method and added Survey Wait Seconds to allow you to set
the amount of time the person called will have to respond after the
survey message is played.

46. Added AST_timecheck.pl script to check for Asterisk update sync issues. Also
added red indicator on Reports page servers table if server falls out of
sync.

47. Updated and cleaned up other language builds for agent and admin.

Cool New Toys excited to test

ready for test,

will update on this...waiting for final version

Hi, Im upgrading my vicidial but I can't find the following path: usr/src/asterisk/asterisk-1.4.39.1-vici

Is this something I need to download from somewhere?

My version was:
VERSION: 2.6-396a
BUILD: 130402-2322


41. Added method for logging SIP error codes, also display of those codes in the
agent screen. The logging goes to the vicidial_carrier_log and
vicidial_dial_log tables and includes both the SIP code and the reason
(i.e. 503 "Service Unavailable"). To get this to work, you need to patch
Asterisk with the extras/sipcausecodelog_1.4.39.1.patch file for
Asterisk 1.4, and for Asterisk 1.8 the patch is pending.

cd /usr/src/asterisk/asterisk-1.4.39.1-vici
patch -p1 < ./sipcausecodelog_1.4.39.1.patch
(file to patch: channels/chan_sip.c)
make clean
./configure
make
make install
(then stop and restart asterisk)

That step is optional, but if you want to do it you first need to tell us how you installed your system. I would recommend starting a new thread for this.

Hello guys ! am new on this forum and it feels good to join you, i really need your help, how do you pause a call that you have already received, sometimes i need pausing when a customer needs my help and i have to talk with someone besides me to get more help..

bmwesigwa wrote:Hello guys ! am new on this forum and it feels good to join you, i really need your help, how do you pause a call that you have already received, sometimes i need pausing when a customer needs my help and i have to talk with someone besides me to get more help..

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) It is not good manners to hijack an unrelated post to ask your question. Open a new thread/create a new post which has a subject line pertinent to your question/problem/suggestion, much better idea.

4) Your best bet is the Vicidial Manager's Manual and Vicidial Agent Manual ...both of which are available in both free and paid versions on EFLO.net. You'll find the information regarding parking a call in there somewhere, but read the whole thing ... you'll enjoy the information it gives quite a bit. And if you intend to make money with this system, you should seriously consider an investment in the Manager's Manual Paid version. It'll help you make ... more money without the waste of trying to find all this stuff out on your own.

Happy Hunting!

Hello Guys,

I have some doubts and I hope to find some answers in this forum.

Since 2006 I am a system administrator of a call center and the Institute works perfectly with both primary flow in voip.

My problem is this. I wanted to upgrade the system by installing Asterisk 1.8 and vicidial 2.7rc1, but reading the scrath_install in the zip file I think anything has changed from older versions of Slackware 11, including the operating system and the version of Asterisk and astgui.

My question is have any of you tried to install Asterisk 1.8 and 2.7rc1 astguiclient on Slackware 14.1 with latest kernel?

Why scrath_install file has not been updated to new versions of Asterisk and astguiclient?

Because scratch install is no longer needed. Vicidial is now installed with an .iso image initially and maintained via SVN after the installation.

Visit Vicibox.com and use the server .iso to perform the installation. There's a PDF on the site to help you install.

After installation, download the Free version of the manual at EFLO.net to configure the system and test it out.

At this point you can either manually import your old data, or attempt to upgrade your prior DB into this system and test it again.

Can anyone tell me how this works?

17. Added ability to set an audio store file for the voicemail greeting message

You have to enable "Allow Voicemail Greeting Chooser" in System Settings before this will work.

Thanks Matt, found it.

One more question Matt, i can't find this setting:

31. Added a campaign feature to allow an agent to search for a lead while on an
inbound call and have their active lead change to a search result lead
that they select.

It's a per campaign setting. You need to set 'Agent Lead Search' to 'LIVE_CALL_INBOUND_AND_MANUAL'.

Thanks, i found the setting and modified it for the campaign. I did a manual dial call from that campaign to test and then clicked LEAD SEARCH but it said i needed to be paused in order to use lead search. Do i need to set something else?

You can only lead search BEFORE starting a manual dial call or DURING an inbound call. Those are the only two conditions that allow it.

Understand that if you did a "manual call", you KNOW who you called. If you are on an Inbound call, however, the CID was either used to identify the caller or just create a new lead. This is obviously not perfect and can often result in an agent NOT being on the right lead. So someone paid to have a system in place to allow the agent to jump to the right lead.

No one has yet paid for a functional to allow this jump on a Manually Dialled call. Likely never will. I can't think of a scenario where it would be needed ... oops: Just did! LOL. Overnight dropped inbound call placed in a "manual dial in the morning" list. Agent calls ... finds out that this is a prior customer who called on a new phone ... now we need to find the real lead but can't.

Fund it.

Hmm i didn't think of that scenario Bill, i'll ask for funding once we get to work with the current feature. I managed to simulate an inbound call and do a live search. Some feedback however:

When i did a search on postal code only i got this:

ERROR: You must enter in search terms, one of these must be populated: lead ID, vendor ID, phone number, last name

Most clients know their postal code and since you can't search with wildcards the change you don't find the right record is still very likely. Why can't i search on postal_code only?

Let me get this straight: You have a client that does not know their own phone number or last name? Really?

As for "why", that would likely be because someone decided during the creation of the feature that this artificial restriction should be in place for some security reason (to avoid an agent "surfing" leads, most likely). The results of a Zip code search could be fairly huge and would allow an agent to (one at a time) copy information from a virtually unlimited set of records without having any real information about them. With "codes" or "Names" they must have at least some basic information to find the lead.

Modifying the code to remove the restriction shouldn't be too terrible.

FYI: We have another client who modified the system to limit the results of this search to specific groups of lists, defined in the Ingroup (campaign lists, ingroup DID lists which are defined as the list_id field from any DID routed to that ingroup, and same DID), each with ALLOW and DISALLOW. Apparently they have multiple call center sponsors and have to be sure to segregate the client lists.

This is for a business to business project so yeah of course people always know their phone number but sometimes companies have different phone numbers and the number that is known by the person who is calling back is not the same as the number that has been dialed.

Good to know it can be modified. We will see how the current limitations work out.

Very well-explained. I'm not sure if extra code would be required or if it would be merely the "lifting" of the restriction, but it's not a difficult modification in either situation.

32. Added inbound email to queue to agent features. To enable, you must add an
"E" keepalive to ONLY ONE of your servers. You will also need to enable
email in System Settings.

I'm trying to setup the new email queue feature.

Do i need to edit /etc/astguiclient.conf and add an E like this?

VARactive_keepalives => 123456789E

I installed the cpan modules and added the cronjob. But i don't see my e-mails coming in. Any tips?

Btw, is it possible to assign e-mail owners or can any agent grab any e-mail?

There is a different post regarding the email features. Probably better to add to that post than to put that information into this one.