vicidial.org

I run a contact center where the main activity is to receive calls for help-desk activities.

The agents have a lot of work answering also emails, faxes and performing calls to the backoffice of my clients.

Therefore they are used to answer calls only after hearing the phone ringing.

I just installed Vicidial and it looks promising in order to help them answer several lines - web forms popup - and to gather statistics.

But so far I didn't found any faciliity like phone ringing when a call is placed to an agent, with a web form for them to choose or not to answer that call.

Is it posible to have such facility (like something refered on the astguiclient has inbound_popup).

Another issue is the skill and queue mamangement.

Since I have serveral agents with diferent skill therefore with an Avaya Definity switch I was doing da following:

1. Welcome Message - ok (FreePBX)
2. Time conditions - ok (FreePBX)
3. Goto to queue os specialized agents - ok (vicidial)
4. 15 seconds - goto to queue general - ?????? - FreePBX ok but I am having troubles with agent or extensions not static or with agents or extensions static I am having a large delay due to the fact that the asterisk or freePBX is taking some time on extensions or agents not available, besides the fact that with freePBX I don't have the possibilitie to associate a web form for a specific campaign.

Regarding the skills of agents on the queue general it would be interesting to define wich agent should answer in first place (priority in each campaign - functionality available on the asterisk queues).

If a call is ringing for more than X seconds at an agent postion than the call should go again to the queue and the agent disconnected.

Finally it woul be also interesting to have the other facilities of asterisk (FreePBX?) like several announcements played during the hold music(position, hold time, voice menu and frequency).

Since I have programming experience I am willing to develop this features but I would appreciate some information if these features are already on the roadmap or if they are somehow already covered and it is only a question of how to configure the system.


Regards

Manuel Mira

Many of these are fairly advanced features that we currently do not have on the development roadmap since we have been concentrating on the outbound portions of the system.

First I must say that we strongly discourage use of the FreePBX configuration interface. It creates limited functionality dialplans that are extremely bloated and inefficient for high-capacity systems and it is very difficult to debug if something goes wrong.

Also, it is important to note that VICIDIAL has absolutely no connection to Asterisk Queues or Asterisk Agents. VICIDIAL uses it's own independant system for Queues and Agents.

Having calls ring to agents can be achieved by using "remote agents" although it will not pull the call back if it is unanswered(though this may be possible with some dialplan configurations). Also there is currently no method for them to refuse to take a call other than not answering.

As for skills-based routing. A simple method of this can be achieved by using in-groups and some dialplan modifications.

As for Queue management(welcome messages, queue position, custom queue music per in-group, etc...), we do plan on starting development on that after the 2.0.1 version is released this month.

If you are interested in developing I would strongly suggest setting VICIDIAL up and getting it functional as well as reading the SCRATCH_INSTALL docs several times. We also do have an SVN server that we do development on.

mflorell wrote:First I must say that we strongly discourage use of the FreePBX configuration interface. It creates limited functionality dialplans that are extremely bloated and inefficient for high-capacity systems and it is very difficult to debug if something goes wrong.

Also, it is important to note that VICIDIAL has absolutely no connection to Asterisk Queues or Asterisk Agents. VICIDIAL uses it's own independant system for Queues and Agents.

Well I already discovered all that, but since I had a crisis situation I decided to go with asterisk on Friday and I implemented the Contact Center in three days with FreePBX (working on Monday), now I took almost 4 days to have Vicidial working with FreePBX and the worst part was inserting the call_logging at the right places on the FreePBX dialplan and I think is still not perfect for what I need or for the Vicidial.

mflorell wrote:Having calls ring to agents can be achieved by using "remote agents" although it will not pull the call back if it is unanswered(though this may be possible with some dialplan configurations). Also there is currently no method for them to refuse to take a call other than not answering.

As for skills-based routing. A simple method of this can be achieved by using in-groups and some dialplan modifications.

Ok. I implemented two in-groups for each campaign. But I don't know how, after 15 seconds, to end the not answered call on the first in-group queue in order to go to the next in-group (two agi-VDADcloser_inboundCID.agi in sucession in the dialplan).

Regarding the inbound_popup.php that as far as I understand is used for the astguiclient.php I was wondering if it could be used on the vicidial.php for answering the calls?

mflorell wrote:As for Queue management(welcome messages, queue position, custom queue music per in-group, etc...), we do plan on starting development on that after the 2.0.1 version is released this month.

If you are interested in developing I would strongly suggest setting VICIDIAL up and getting it functional as well as reading the SCRATCH_INSTALL docs several times. We also do have an SVN server that we do development on.

Well I already have the astguiclient 2 beta 1 working, by the way I had to edit a lot of agi(s) since there were a typo error on some debug messages ("") at the beginnning, but since I never work on a development community (open-source) were can I find some info about policies and processes worklow generic or specific to the astguiclient development?

Regards

MJSMira wrote:...now I took almost 4 days to have Vicidial working with FreePBX and the worst part was inserting the call_logging at the right places on the FreePBX dialplan and I think is still not perfect for what I need or for the Vicidial.

FreePBX doesn't let you do manual entries very easily. And you didn't even have to get into any of the lovely CallerID issues that arise with outbound VICIDIAL usage in FreePBX.

Ok. I implemented two in-groups for each campaign. But I don't know how, after 15 seconds, to end the not answered call on the first in-group queue in order to go to the next in-group (two agi-VDADcloser_inboundCID.agi in sucession in the dialplan).

We have never had anyone request a feature for rolling from one in-group to another, but if you put the second agi-VDADcloser_inboundCID.agi on a different extension and put that extansion number in the "Drop Exten" field of the first in-group then the call should go from the first in-group to the second after the "Drop Call Seconds" timeout of the first in-group has been reached.

Regarding the inbound_popup.php that as far as I understand is used for the astguiclient.php I was wondering if it could be used on the vicidial.php for answering the calls?

the vicidial/vdremote.php script is what remote agents are supposed to be using to receive calls and see data on their calls. It is functional but has not been updated in quite a while since hardly anyone uses it. It does not popup, but it does refresh to display customer data. As for adapting the inbound_popup.php to work with VICIDIAL. That may be possible, but it will take quite a bit of code changes.

Well I already have the astguiclient 2 beta 1 working, by the way I had to edit a lot of agi(s) since there were a typo error on some debug messages ("") at the beginnning, but since I never work on a development community (open-source) were can I find some info about policies and processes worklow generic or specific to the astguiclient development?

Yep, those bugs have been fixed in SVN 2-X and I will be releasing beta2 tomorrow with a lot more bug fixes and a few new features. It is a beta so bugs/typos/errors are to be expected and I have received a lot of good feedback.

As for policies and processes, we do have a tracker where you can report bugs and ask for or add new features:
http://www.eflo.net/VICIDIALmantis


And the policies for the project are:

The restrictions on the web GUI are:
- must fit within a web browser application on an 800x600 VGA screen
(this leaves 770x450 actual space)
- must not have any main data fields or control functions hidden or only accessible by scrolling
- must use compatible code with Firefox 1.0 and higher and use only Javascript for client-side parsing and AJAX

The restrictions on the server-side scripts are:
- campaign features must be able to work across all campaigns, on a per campaign basis.
- all coding is done in Perl or PHP with special exception for Ploticus for graphing.
- scripts must work for systems running on multiple servers and single server setups.

And the final guideline is that I have final say on everything in the project because I wrote and maintain almost all of it.

Releases come out when coding is done, tested and ready, not on a specific schedule. SVN is publicly available for people to download from to get current development code.

Hi we've got the overflow ingroup escenario requirement.

We've configured like this:

FRONT INGROUP receiving the call, if there's no agent available after the DROP TIMEOUT it would go to another INGROUP.

The problem i'm facing is that when the call overflows to the next INGROUP it lost it's data.

When the call is dropped and goes to the ingroup, the data fields are messed, so for example i've got at the phone field this data Y03311552280 at the comments field this Y0331155228000000405.

so is there any way to preserve the data variables after the call goes to the dropped extension?

Asterisk version?

astguiclient version?

It happens with asterisk 1.2.14 and 1.2.16.2 and astguiclient 2.04

There is now a feature to do this properly in SVN trunk. It is not easily possible to do that with lead information in 2.0.4

Thks Matt!!! i'm gonna test that ASAP and give you feedback on this!!

Great!