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Wednesday, October 24, 2007


PSTN or POTS = Public Switched Telephone Network 
its the phone service we use for every ordinary phone call.

vicidial dialer

part of the astGUIclient suite

This system was initially created to fill the need of a customer service group being able to call up clients efficiently and without using a predictive dialer that would potentially lose calls and alienate clients. Since then we have added the ability to do predictive dialing alongside the traditional one-call-at-a-time dialing as well as the ability to take inbound calls and set up campaigns to do blended inbound and outbound at the same time.

Web-client program uses AJAX(php/javascript/XMLHTTPRequest) and requires a web browser.

This system was initially created to fill the need of a customer service group
being able to call up clients efficiently and without using a predictive dialer
that would potentially lose calls and alienate clients. Since then we have added
the ability to do predictive dialing alongside ratio dialing and the traditional
one-call-at-a-time dialing as well as the ability to take inbound calls and set
up campaigns to do blended inbound and outbound call handling at the same time.

Major Features:
- Inbound, Outbound and Blended call handling
- Outbound agent-controlled, broadcast and predictive dialing
- Full USA FTC-compliance capability
- Web-based agent and administrative interfaces
- Ability to have agents operate remotely
- Integrated call recording
- Three-Way calling within the agent application
- Scheduled Callbacks: Agent-Only and Anyone
- Scalable to hundreds of seats
- Ability to use standard Telco lines and VOIP trunks
- Open-Source GPL licensed, with no software licensing cost

Complete Features List:
- Ability for an agent to call clients in succession from a database through a web-client
- Ability to display a script for the agent to read with fields like name, address, etc. filled-in
- Ability to set a campaign to auto-dial and send live calls to available agents
- Ability to dial predictively in a campaign with an adaptive dialing algorithm
- Ability to dial on a single campaign across multiple Asterisk servers, or multiple campaigns on a single server
- Ability to transfer calls with customer data to a closer on the local system or a remote Asterisk server
- Ability to open a custom web page with user data from the call, per campaign
- Ability to autodial campaigns to start with a simple IVR then direct to agent
- Ability to broadcast dial to customers with a pre-recorded message
- Ability to park the customer with custom music per campaign
- Ability to send a dropped call to a voicemail box per campaign if no agent is available
- Ability to set outbound CallerID per campaign
- Ability to take inbound calls grabbing CallerID
- Ability to function as an ACD for inbound and fronter/closer verification calls
- Ability to have an agent take both inbound and outbound calls in one session(blended)
- Ability to start and stop recording an agent's calls at any time
- Ability to automatically record all calls
- Ability to manually or automatically call upto two other customer numbers for the same lead
- Ability to schedule a callback with a customer as either any-agent or agent-specific
- Ability in Manual dial mode to preview leads before dialing
- Ability for agents to be logged in remotely anywhere with just a phone and a web browser
- Faster hangup and dispositioning of calls with one key press (HotKeys)
- Definable Agent Wrapup-time per campaign
- Ability to add custom call dispositions per campaign
- Ability to use custom database queries in campaign dialing
- Recycling of Busy calls at a specified interval without resetting a list
- Dialing with custom TimeZone restrictions including per state and per day-of-the-week
- Dialing with Answering Machine Detection, also playing a message for AM calls
- Multiple campaigns and lead-lists are possible
- Option of a drop timer with safe-harbor message for FTC compliance
- Variable drop call percentage when dialing predictively for FTC compliance
- Internal DNC list can optionally be activated per campaign
- All calls are logged and statuses of calls are logged as well as agent time breakdowns
- Load Balancing across multiple inbound or outbound Asterisk servers is possible
- Several real-time and summary reports available
- Real-time campaign display screens
- 3rd party conferencing(with DTMF macros and number presets)
- 3rd party blind call transfer
- 3rd party conferencing with agent drop-off
- Ability to set user levels and permissions for certain features and campaigns
- Ability for managers to listen-in on agent conversations
- Ability for managers to enter conversations with agents and customers
- Ability for agents to select a Pause Code when they are not active
- Ability for agents to control volume levels and mute themselves
- Outbound dialing is QueueMetrics-compatible
- Web-based administration
- Client web-app web pages available in English, Spanish, Greek, German, French, Italian, Polish, Portuguese and Brazillian Portuguese
- Admin web pages available in English, Spanish, Greek and German

Required components:
- Asterisk server with Zap, IAX or SIP trunks
- SIP/IAX/Zap phones(hard or soft single-line or multiple)
- 500MHz Computer with Firefox installed
- Asterisk server with proper software installed
- astGUIclient must be installed with the Asterisk Central Queue System(ACQS) running

This system is dependant on a MySQL server(version 4.0.XX or higher) acting as the storage place of all data
and settings for the system. This was done so that it would be very scalable and flexible. This application draws
on a lot of the core scripting inside of the astGUIclient suite of applications and was built with flexibility in mind.
We have used VICIDIAL for over 3 years now on up to 160 seats at once across 7 separate Asterisk servers
all using the same MySQL server and dialing on the same campaign. It is important to keep in mind that
because VICIDIAL uses Asterisk meetme rooms for it's dialing that the Asterisk server will see an increased
load as compared to a simple Zap/SIP-only calls.

Look at the INSTALL FROM SCRATCH documentation and start at phase 6

Process flow of a one-call-at-a-time call:

1. all components are installed and running(with Asterisk and MySQL servers)
2. user logs in to the client GUI (user/password/campaign)
3. GUI assigns an open meetme extension to the user, this is called the session_id
4. GUI sends manager command ORIGINATE to connect the user to the meetme ext
5. user clicks DIAL NEXT NUMBER and GUI UPDATEs first available number-to-call in DB to PENDING status
6. GUI SELECTs info on that number(name/address/...)
7. GUI sends ORIGINATE command to call that client's number from meetme ext
8. user is talking to the client
* user transfers call to 3rd party, client Zap channel is REDIRECTed to 3rd party
* User transfers client to a closer to finish the call
* user puts client on park and dials 3rd party, then transfers client back to meetme room
9. Call is Hung up
* user hangs up call and clicks HANGUP CALL on GUI

* person called hangs up so user clicks on the CALL HUNGUP then HANGUP CALL
10. call is disconnected
11. user selects disposition status of the call and clicks OK
12. goes back to step 4 and repeats

Process flow of an auto-dial call:

1. The lead hopper app(AST_VDhopper.pl) that is running every minute keeps the hopper for each campaign
up to at least the minimum hopper lead size providing there are enough leads to fill it.
2. The call center manager can either set the auto_dial_level (on the campaign screen of the admin web page)
to 0(zero) for no autodialing or any other choice for a different number of lines-per-agent-logged-in.
3. As soon as an agent logs into a campaign, the client app will look at the auto_dial_level setting of the
campaign and determine whether
* it will insert a record into the vicidial_live_agents table(and set status to READY) and update that record
every second with a random 8 digit number to show that the client is still alive and ready. Also, the client app will
show a "pause" and "resume" button to start or stop receiving calls automatically instead of the usual "dial next number" button.
* it will allow normal manual dialing operation
4. The auto_dial app(AST_VDauto_dial.pl) will:
1. count how many agents are READY,QUEUE,INCALL with updated last_update_time in the last 5 seconds
2. look at the campaign's auto_dial_level to determine how many calls should be placed each specific campaign on each server.
3. update the vicidial_auto_calls records by searching for active callerids in the vicidial_manager to see if they have hung up
4. count the number of active calls that are occuring(vicidial_auto_calls records with a status of SENT,RINGING or LIVE)
5. place the calls for each server if the active calls is less than the total that should be placed
5. The auto_dial app will then:
1. grab the next lead to be called from the vicidial_hopper for each campaign(it will update that lead record in the vicidial_hopper
table to QUEUE with VDAD as the user)
2. grab the lead information from vicidial_list and update called_since_last_reset Y, the user to VDAD and increment the called_count by 1
3. delete the record from vicidial_hopper for that lead
4. place the call first to the Zap trunk(Exten: 917275551212), then to a specified Local extension number
(Channel: Local/99999@default) where the auto_dial_transfer_agent app(agi-VDADtransfer.agi) is waiting to take over.
5. if the call is busy, disconnect or telco voicemail the call is never technically picked up so it never makes it to the auto_dial_transfer_agent app
6. check for dead calls by looking in the call_log for and end time of the call keyed by callerID(caller_code) and delete them t=from the vicidial_auto_calls table
6. The auto_dial_transfer_agent app takes the call and:
1. immediately grabs the uniqueid and callerid of the call and updates the vicidial_auto_calls record to LIVE
2. logs the call to vicidial_log using the callerid to find out the lead number(VmmddhhmmssLLLLLLLLL(where LLLLLLLLL
is the zero-padded lead_id from vicidial_list))
3. inserts the number_dialed and the caller_code in the vicidial_log table for the uniqueid record of the call
4. attempts to update(limit 1) a record in the vicidial_live_agents table to see who is the next agent to receive a
call(ordered by last_call_time)(update the status to QUEUE, the lead_id to the proper lead_id and the uniqueid and callerID to proper values for the call)
5. if there are agents available(affected rows = 1)(goto substep VII.)
6. if there are no agents available:
1. wait in 1 second increments and then checking substep IV.
2. if no agent is available for 1 minute, the call is hung up and logged as ABANDN in vicidial_log under user VDAD
7. grabs the agent info from the vicidial_live_agents(conf_exten) and does a redirect to the agent's meetme room.
8. agent's app checks the vicidial_live_agents record every second for itself to see if it is changed to QUEUE
9. agent's app changes the vicidial_live_agents status to INCALL and updates the vicidial_list record to INCALL and user
to the agent's userID and the call progresses normally
7. The agent hangs up and the vicidial_live_agents record is changed back to READY with the lead and call fields cleared out
so the agent is ready to take another call

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Tuesday, October 23, 2007

Our SPA941 IP Phones

today our supplier has just called and informed us that our Linksys SPA941 IP Phones has just arrived. we will be using this phones in our call center."bluebox global solutions"

Friday, October 19, 2007

Games Download for K-PEX100

download games for K-PEX100 games are available for download in this site

My New Kingstone K-PEX100 Portable

Product Specifications
Memory: 1GB or 2GB (built-in) with free 256 MB miniSD (expandable to 2GB)
Display: 2-inch color TFT LCD display
USB 2.0 compliant
Supported format: Music (MP3, WMA, Ogg(Q10), WAV), Video (AVI, ASF, MPEG 1&2, WMV to MPX), TXT files, JPEG
Included Accessories: USB 2.0 Cable and USB host cable, Earphones, Transcoder Software CD, Protective Cover, Line-in cable, Lanyard, Manual
Size: 94cm x 45.7cm x 14.5cm
Weight: 65g

Wednesday, October 17, 2007


SIP = Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet.

VOIP Phones for Asterisk IP PBX

Morpheus has confirm.and i already orderd 12 VOIP Phones for startup.. and i am planning to use it for Asterisk. Its a Linksys SPA921 IP Phone.